diff --git a/.coveragerc b/.coveragerc
deleted file mode 100644
index b178b094..00000000
--- a/.coveragerc
+++ /dev/null
@@ -1,19 +0,0 @@
-# Generated by synthtool. DO NOT EDIT!
-[run]
-branch = True
-
-[report]
-fail_under = 100
-show_missing = True
-exclude_lines =
-    # Re-enable the standard pragma
-    pragma: NO COVER
-    # Ignore debug-only repr
-    def __repr__
-    # Ignore abstract methods
-    raise NotImplementedError
-omit =
-  */gapic/*.py
-  */proto/*.py
-  */core/*.py
-  */site-packages/*.py
\ No newline at end of file
diff --git a/.flake8 b/.flake8
deleted file mode 100644
index 0268ecc9..00000000
--- a/.flake8
+++ /dev/null
@@ -1,14 +0,0 @@
-# Generated by synthtool. DO NOT EDIT!
-[flake8]
-ignore = E203, E266, E501, W503
-exclude =
-  # Exclude generated code.
-  **/proto/**
-  **/gapic/**
-  *_pb2.py
-
-  # Standard linting exemptions.
-  __pycache__,
-  .git,
-  *.pyc,
-  conf.py
diff --git a/.github/CONTRIBUTING.md b/.github/CONTRIBUTING.md
deleted file mode 100644
index 939e5341..00000000
--- a/.github/CONTRIBUTING.md
+++ /dev/null
@@ -1,28 +0,0 @@
-# How to Contribute
-
-We'd love to accept your patches and contributions to this project. There are
-just a few small guidelines you need to follow.
-
-## Contributor License Agreement
-
-Contributions to this project must be accompanied by a Contributor License
-Agreement. You (or your employer) retain the copyright to your contribution;
-this simply gives us permission to use and redistribute your contributions as
-part of the project. Head over to <https://cla.developers.google.com/> to see
-your current agreements on file or to sign a new one.
-
-You generally only need to submit a CLA once, so if you've already submitted one
-(even if it was for a different project), you probably don't need to do it
-again.
-
-## Code reviews
-
-All submissions, including submissions by project members, require review. We
-use GitHub pull requests for this purpose. Consult
-[GitHub Help](https://help.github.com/articles/about-pull-requests/) for more
-information on using pull requests.
-
-## Community Guidelines
-
-This project follows [Google's Open Source Community
-Guidelines](https://opensource.google.com/conduct/).
diff --git a/.github/ISSUE_TEMPLATE/bug_report.md b/.github/ISSUE_TEMPLATE/bug_report.md
deleted file mode 100644
index 2924b3fb..00000000
--- a/.github/ISSUE_TEMPLATE/bug_report.md
+++ /dev/null
@@ -1,44 +0,0 @@
----
-name: Bug report
-about: Create a report to help us improve
-
----
-
-Thanks for stopping by to let us know something could be better!
-
-**PLEASE READ**: If you have a support contract with Google, please create an issue in the [support console](https://cloud.google.com/support/) instead of filing on GitHub. This will ensure a timely response.
-
-Please run down the following list and make sure you've tried the usual "quick fixes":
-
-  - Search the issues already opened: https://github.com/googleapis/python-speech/issues
-  - Search the issues on our "catch-all" repository: https://github.com/googleapis/google-cloud-python
-  - Search StackOverflow: http://stackoverflow.com/questions/tagged/google-cloud-platform+python
-
-If you are still having issues, please be sure to include as much information as possible:
-
-#### Environment details
-
-  - OS type and version:
-  - Python version: `python --version`
-  - pip version: `pip --version`
-  - `google-cloud-speech` version: `pip show google-cloud-speech`
-
-#### Steps to reproduce
-
-  1. ?
-  2. ?
-
-#### Code example
-
-```python
-# example
-```
-
-#### Stack trace
-```
-# example
-```
-
-Making sure to follow these steps will guarantee the quickest resolution possible.
-
-Thanks!
diff --git a/.github/ISSUE_TEMPLATE/feature_request.md b/.github/ISSUE_TEMPLATE/feature_request.md
deleted file mode 100644
index 6365857f..00000000
--- a/.github/ISSUE_TEMPLATE/feature_request.md
+++ /dev/null
@@ -1,18 +0,0 @@
----
-name: Feature request
-about: Suggest an idea for this library
-
----
-
-Thanks for stopping by to let us know something could be better!
-
-**PLEASE READ**: If you have a support contract with Google, please create an issue in the [support console](https://cloud.google.com/support/) instead of filing on GitHub. This will ensure a timely response.
-
- **Is your feature request related to a problem? Please describe.**
-A clear and concise description of what the problem is. Ex. I'm always frustrated when [...]
- **Describe the solution you'd like**
-A clear and concise description of what you want to happen.
- **Describe alternatives you've considered**
-A clear and concise description of any alternative solutions or features you've considered.
- **Additional context**
-Add any other context or screenshots about the feature request here.
diff --git a/.github/ISSUE_TEMPLATE/support_request.md b/.github/ISSUE_TEMPLATE/support_request.md
deleted file mode 100644
index 99586903..00000000
--- a/.github/ISSUE_TEMPLATE/support_request.md
+++ /dev/null
@@ -1,7 +0,0 @@
----
-name: Support request
-about: If you have a support contract with Google, please create an issue in the Google Cloud Support console.
-
----
-
-**PLEASE READ**: If you have a support contract with Google, please create an issue in the [support console](https://cloud.google.com/support/) instead of filing on GitHub. This will ensure a timely response.
diff --git a/.github/PULL_REQUEST_TEMPLATE.md b/.github/PULL_REQUEST_TEMPLATE.md
deleted file mode 100644
index 74b4cadc..00000000
--- a/.github/PULL_REQUEST_TEMPLATE.md
+++ /dev/null
@@ -1,7 +0,0 @@
-Thank you for opening a Pull Request! Before submitting your PR, there are a few things you can do to make sure it goes smoothly:
-- [ ] Make sure to open an issue as a [bug/issue](https://github.com/googleapis/python-speech/issues/new/choose) before writing your code!  That way we can discuss the change, evaluate designs, and agree on the general idea
-- [ ] Ensure the tests and linter pass
-- [ ] Code coverage does not decrease (if any source code was changed)
-- [ ] Appropriate docs were updated (if necessary)
-
-Fixes #<issue_number_goes_here> 🦕
diff --git a/.github/release-please.yml b/.github/release-please.yml
deleted file mode 100644
index 4507ad05..00000000
--- a/.github/release-please.yml
+++ /dev/null
@@ -1 +0,0 @@
-releaseType: python
diff --git a/.gitignore b/.gitignore
deleted file mode 100644
index 3fb06e09..00000000
--- a/.gitignore
+++ /dev/null
@@ -1,58 +0,0 @@
-*.py[cod]
-*.sw[op]
-
-# C extensions
-*.so
-
-# Packages
-*.egg
-*.egg-info
-dist
-build
-eggs
-parts
-bin
-var
-sdist
-develop-eggs
-.installed.cfg
-lib
-lib64
-__pycache__
-
-# Installer logs
-pip-log.txt
-
-# Unit test / coverage reports
-.coverage
-.nox
-.cache
-.pytest_cache
-
-
-# Mac
-.DS_Store
-
-# JetBrains
-.idea
-
-# VS Code
-.vscode
-
-# emacs
-*~
-
-# Built documentation
-docs/_build
-bigquery/docs/generated
-
-# Virtual environment
-env/
-coverage.xml
-
-# System test environment variables.
-system_tests/local_test_setup
-
-# Make sure a generated file isn't accidentally committed.
-pylintrc
-pylintrc.test
\ No newline at end of file
diff --git a/.kokoro/build.sh b/.kokoro/build.sh
deleted file mode 100755
index aa692ac7..00000000
--- a/.kokoro/build.sh
+++ /dev/null
@@ -1,39 +0,0 @@
-#!/bin/bash
-# Copyright 2018 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-set -eo pipefail
-
-cd github/python-speech
-
-# Disable buffering, so that the logs stream through.
-export PYTHONUNBUFFERED=1
-
-# Debug: show build environment
-env | grep KOKORO
-
-# Setup service account credentials.
-export GOOGLE_APPLICATION_CREDENTIALS=${KOKORO_GFILE_DIR}/service-account.json
-
-# Setup project id.
-export PROJECT_ID=$(cat "${KOKORO_GFILE_DIR}/project-id.json")
-
-# Remove old nox
-python3.6 -m pip uninstall --yes --quiet nox-automation
-
-# Install nox
-python3.6 -m pip install --upgrade --quiet nox
-python3.6 -m nox --version
-
-python3.6 -m nox
diff --git a/.kokoro/continuous/common.cfg b/.kokoro/continuous/common.cfg
deleted file mode 100644
index f75e6ab4..00000000
--- a/.kokoro/continuous/common.cfg
+++ /dev/null
@@ -1,27 +0,0 @@
-# Format: //devtools/kokoro/config/proto/build.proto
-
-# Build logs will be here
-action {
-  define_artifacts {
-    regex: "**/*sponge_log.xml"
-  }
-}
-
-# Download trampoline resources.
-gfile_resources: "/bigstore/cloud-devrel-kokoro-resources/trampoline"
-
-# Download resources for system tests (service account key, etc.)
-gfile_resources: "/bigstore/cloud-devrel-kokoro-resources/google-cloud-python"
-
-# Use the trampoline script to run in docker.
-build_file: "python-speech/.kokoro/trampoline.sh"
-
-# Configure the docker image for kokoro-trampoline.
-env_vars: {
-    key: "TRAMPOLINE_IMAGE"
-    value: "gcr.io/cloud-devrel-kokoro-resources/python-multi"
-}
-env_vars: {
-    key: "TRAMPOLINE_BUILD_FILE"
-    value: "github/python-speech/.kokoro/build.sh"
-}
diff --git a/.kokoro/continuous/continuous.cfg b/.kokoro/continuous/continuous.cfg
deleted file mode 100644
index 8f43917d..00000000
--- a/.kokoro/continuous/continuous.cfg
+++ /dev/null
@@ -1 +0,0 @@
-# Format: //devtools/kokoro/config/proto/build.proto
\ No newline at end of file
diff --git a/.kokoro/docs/common.cfg b/.kokoro/docs/common.cfg
deleted file mode 100644
index 2d3db445..00000000
--- a/.kokoro/docs/common.cfg
+++ /dev/null
@@ -1,48 +0,0 @@
-# Format: //devtools/kokoro/config/proto/build.proto
-
-# Build logs will be here
-action {
-  define_artifacts {
-    regex: "**/*sponge_log.xml"
-  }
-}
-
-# Download trampoline resources.
-gfile_resources: "/bigstore/cloud-devrel-kokoro-resources/trampoline"
-
-# Use the trampoline script to run in docker.
-build_file: "python-speech/.kokoro/trampoline.sh"
-
-# Configure the docker image for kokoro-trampoline.
-env_vars: {
-    key: "TRAMPOLINE_IMAGE"
-    value: "gcr.io/cloud-devrel-kokoro-resources/python-multi"
-}
-env_vars: {
-    key: "TRAMPOLINE_BUILD_FILE"
-    value: "github/python-speech/.kokoro/publish-docs.sh"
-}
-
-env_vars: {
-    key: "STAGING_BUCKET"
-    value: "docs-staging"
-}
-
-# Fetch the token needed for reporting release status to GitHub
-before_action {
-  fetch_keystore {
-    keystore_resource {
-      keystore_config_id: 73713
-      keyname: "yoshi-automation-github-key"
-    }
-  }
-}
-
-before_action {
-  fetch_keystore {
-    keystore_resource {
-      keystore_config_id: 73713
-      keyname: "docuploader_service_account"
-    }
-  }
-}
\ No newline at end of file
diff --git a/.kokoro/docs/docs.cfg b/.kokoro/docs/docs.cfg
deleted file mode 100644
index 8f43917d..00000000
--- a/.kokoro/docs/docs.cfg
+++ /dev/null
@@ -1 +0,0 @@
-# Format: //devtools/kokoro/config/proto/build.proto
\ No newline at end of file
diff --git a/.kokoro/presubmit/common.cfg b/.kokoro/presubmit/common.cfg
deleted file mode 100644
index f75e6ab4..00000000
--- a/.kokoro/presubmit/common.cfg
+++ /dev/null
@@ -1,27 +0,0 @@
-# Format: //devtools/kokoro/config/proto/build.proto
-
-# Build logs will be here
-action {
-  define_artifacts {
-    regex: "**/*sponge_log.xml"
-  }
-}
-
-# Download trampoline resources.
-gfile_resources: "/bigstore/cloud-devrel-kokoro-resources/trampoline"
-
-# Download resources for system tests (service account key, etc.)
-gfile_resources: "/bigstore/cloud-devrel-kokoro-resources/google-cloud-python"
-
-# Use the trampoline script to run in docker.
-build_file: "python-speech/.kokoro/trampoline.sh"
-
-# Configure the docker image for kokoro-trampoline.
-env_vars: {
-    key: "TRAMPOLINE_IMAGE"
-    value: "gcr.io/cloud-devrel-kokoro-resources/python-multi"
-}
-env_vars: {
-    key: "TRAMPOLINE_BUILD_FILE"
-    value: "github/python-speech/.kokoro/build.sh"
-}
diff --git a/.kokoro/presubmit/presubmit.cfg b/.kokoro/presubmit/presubmit.cfg
deleted file mode 100644
index 8f43917d..00000000
--- a/.kokoro/presubmit/presubmit.cfg
+++ /dev/null
@@ -1 +0,0 @@
-# Format: //devtools/kokoro/config/proto/build.proto
\ No newline at end of file
diff --git a/.kokoro/publish-docs.sh b/.kokoro/publish-docs.sh
deleted file mode 100755
index 53c8bb08..00000000
--- a/.kokoro/publish-docs.sh
+++ /dev/null
@@ -1,57 +0,0 @@
-#!/bin/bash
-# Copyright 2020 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-#!/bin/bash
-
-set -eo pipefail
-
-# Disable buffering, so that the logs stream through.
-export PYTHONUNBUFFERED=1
-
-cd github/python-speech
-
-# Remove old nox
-python3.6 -m pip uninstall --yes --quiet nox-automation
-
-# Install nox
-python3.6 -m pip install --upgrade --quiet nox
-python3.6 -m nox --version
-
-# build docs
-nox -s docs
-
-python3 -m pip install gcp-docuploader
-
-# install a json parser
-sudo apt-get update
-sudo apt-get -y install software-properties-common
-sudo add-apt-repository universe
-sudo apt-get update
-sudo apt-get -y install jq
-
-# create metadata
-python3 -m docuploader create-metadata \
-  --name=$(jq --raw-output '.name // empty' .repo-metadata.json) \
-  --version=$(python3 setup.py --version) \
-  --language=$(jq --raw-output '.language // empty' .repo-metadata.json) \
-  --distribution-name=$(python3 setup.py --name) \
-  --product-page=$(jq --raw-output '.product_documentation // empty' .repo-metadata.json) \
-  --github-repository=$(jq --raw-output '.repo // empty' .repo-metadata.json) \
-  --issue-tracker=$(jq --raw-output '.issue_tracker // empty' .repo-metadata.json)
-
-cat docs.metadata
-
-# upload docs
-python3 -m docuploader upload docs/_build/html --metadata-file docs.metadata --staging-bucket docs-staging
diff --git a/.kokoro/release.sh b/.kokoro/release.sh
deleted file mode 100755
index 357a9b99..00000000
--- a/.kokoro/release.sh
+++ /dev/null
@@ -1,34 +0,0 @@
-#!/bin/bash
-# Copyright 2020 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-#!/bin/bash
-
-set -eo pipefail
-
-# Start the releasetool reporter
-python3 -m pip install gcp-releasetool
-python3 -m releasetool publish-reporter-script > /tmp/publisher-script; source /tmp/publisher-script
-
-# Ensure that we have the latest versions of Twine, Wheel, and Setuptools.
-python3 -m pip install --upgrade twine wheel setuptools
-
-# Disable buffering, so that the logs stream through.
-export PYTHONUNBUFFERED=1
-
-# Move into the package, build the distribution and upload.
-TWINE_PASSWORD=$(cat "${KOKORO_KEYSTORE_DIR}/73713_google_cloud_pypi_password")
-cd github/python-speech
-python3 setup.py sdist bdist_wheel
-twine upload --username gcloudpypi --password "${TWINE_PASSWORD}" dist/*
diff --git a/.kokoro/release/common.cfg b/.kokoro/release/common.cfg
deleted file mode 100644
index 22280827..00000000
--- a/.kokoro/release/common.cfg
+++ /dev/null
@@ -1,64 +0,0 @@
-# Format: //devtools/kokoro/config/proto/build.proto
-
-# Build logs will be here
-action {
-  define_artifacts {
-    regex: "**/*sponge_log.xml"
-  }
-}
-
-# Download trampoline resources.
-gfile_resources: "/bigstore/cloud-devrel-kokoro-resources/trampoline"
-
-# Use the trampoline script to run in docker.
-build_file: "python-speech/.kokoro/trampoline.sh"
-
-# Configure the docker image for kokoro-trampoline.
-env_vars: {
-    key: "TRAMPOLINE_IMAGE"
-    value: "gcr.io/cloud-devrel-kokoro-resources/python-multi"
-}
-env_vars: {
-    key: "TRAMPOLINE_BUILD_FILE"
-    value: "github/python-speech/.kokoro/release.sh"
-}
-
-# Fetch the token needed for reporting release status to GitHub
-before_action {
-  fetch_keystore {
-    keystore_resource {
-      keystore_config_id: 73713
-      keyname: "yoshi-automation-github-key"
-    }
-  }
-}
-
-# Fetch PyPI password
-before_action {
-  fetch_keystore {
-    keystore_resource {
-      keystore_config_id: 73713
-      keyname: "google_cloud_pypi_password"
-    }
-  }
-}
-
-# Fetch magictoken to use with Magic Github Proxy 
-before_action {
-  fetch_keystore {
-    keystore_resource {
-      keystore_config_id: 73713
-      keyname: "releasetool-magictoken"
-    }
-  }
-}
-
-# Fetch api key to use with Magic Github Proxy 
-before_action {
-  fetch_keystore {
-    keystore_resource {
-      keystore_config_id: 73713
-      keyname: "magic-github-proxy-api-key"
-    }
-  }
-}
diff --git a/.kokoro/release/release.cfg b/.kokoro/release/release.cfg
deleted file mode 100644
index 8f43917d..00000000
--- a/.kokoro/release/release.cfg
+++ /dev/null
@@ -1 +0,0 @@
-# Format: //devtools/kokoro/config/proto/build.proto
\ No newline at end of file
diff --git a/.kokoro/trampoline.sh b/.kokoro/trampoline.sh
deleted file mode 100755
index e8c4251f..00000000
--- a/.kokoro/trampoline.sh
+++ /dev/null
@@ -1,23 +0,0 @@
-#!/bin/bash
-# Copyright 2017 Google Inc.
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#      http://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-set -eo pipefail
-
-python3 "${KOKORO_GFILE_DIR}/trampoline_v1.py"  || ret_code=$?
-
-chmod +x ${KOKORO_GFILE_DIR}/trampoline_cleanup.sh
-${KOKORO_GFILE_DIR}/trampoline_cleanup.sh || true
-
-exit ${ret_code}
diff --git a/.repo-metadata.json b/.repo-metadata.json
index 15006765..47b2eb69 100644
--- a/.repo-metadata.json
+++ b/.repo-metadata.json
@@ -1,13 +1,18 @@
 {
-  "name": "speech",
-  "name_pretty": "Cloud Speech",
-  "product_documentation": "https://cloud.google.com/speech-to-text/docs/",
-  "client_documentation": "https://googleapis.dev/python/speech/latest",
-  "issue_tracker": "https://issuetracker.google.com/savedsearches/559758",
-  "release_level": "ga",
-  "language": "python",
-  "repo": "googleapis/python-speech",
-  "distribution_name": "google-cloud-speech",
-  "api_id": "speech.googleapis.com",
-  "requires_billing": false
-}
\ No newline at end of file
+    "name": "speech",
+    "name_pretty": "Cloud Speech",
+    "product_documentation": "https://cloud.google.com/speech-to-text/docs/",
+    "client_documentation": "https://cloud.google.com/python/docs/reference/speech/latest",
+    "issue_tracker": "https://issuetracker.google.com/savedsearches/559758",
+    "release_level": "stable",
+    "language": "python",
+    "library_type": "GAPIC_COMBO",
+    "repo": "googleapis/python-speech",
+    "distribution_name": "google-cloud-speech",
+    "api_id": "speech.googleapis.com",
+    "requires_billing": false,
+    "default_version": "v1",
+    "codeowner_team": "@googleapis/cdpe-cloudai",
+    "api_shortname": "speech",
+    "api_description": "enables easy integration of Google speech recognition technologies into developer applications. Send audio and receive a text transcription from the Speech-to-Text API service."
+}
diff --git a/CHANGELOG.md b/CHANGELOG.md
deleted file mode 100644
index 5c61fa3a..00000000
--- a/CHANGELOG.md
+++ /dev/null
@@ -1,262 +0,0 @@
-# Changelog
-
-[PyPI History][1]
-
-[1]: https://pypi.org/project/google-cloud-speech/#history
-
-### [1.3.2](https://www.github.com/googleapis/python-speech/compare/v1.3.1...v1.3.2) (2020-02-03)
-
-
-### Bug Fixes
-
-* **speech:** increase default timeout for v1p1beta1 (via synth) ([#9999](https://www.github.com/googleapis/python-speech/issues/9999)) ([e9b4919](https://www.github.com/googleapis/python-speech/commit/e9b4919f493a9206406944093fc1c3408b5f0265))
-* **speech:** increase timeout values in client config (via synth) ([#9922](https://www.github.com/googleapis/python-speech/issues/9922)) ([8d34bea](https://www.github.com/googleapis/python-speech/commit/8d34beab73273e1b0965fe622af5434fdbac01ca))
-* **speech:** mark `Recognize` idempotent (via synth) ([#9874](https://www.github.com/googleapis/python-speech/issues/9874)) ([a69e08c](https://www.github.com/googleapis/python-speech/commit/a69e08ce0bafcd9c4f1bafe51dce6e6b4716994f))
-* **speech:** move 'speaker_tag' field from 'SpeakerDiarizationConfig' to 'WordInfo' (via synth) ([#9576](https://www.github.com/googleapis/python-speech/issues/9576)) ([2ba5ab9](https://www.github.com/googleapis/python-speech/commit/2ba5ab969e09df5cafcf67861c9d7d6cc91bd0ca))
-* **speech:** re-add unused speaker_tag; update spacing in docs templates (via synth) ([#9765](https://www.github.com/googleapis/python-speech/issues/9765)) ([e1c5a54](https://www.github.com/googleapis/python-speech/commit/e1c5a54a321f5e76f074f8be1f5b6d5aedd612c7))
-
-## 1.3.1
-
-12-06-2019 13:05 PST
-
-### Implementation Changes
-- Increase timeout values in client config (via synth). ([#9922](https://github.com/googleapis/google-cloud-python/pull/9922))
-
-## 1.3.0
-
-11-21-2019 14:03 PST
-
-
-### Implementation Changes
-- Mark `Recognize` as idempotent (via synth). ([#9874](https://github.com/googleapis/google-cloud-python/pull/9874))
-- Re-add unused `speaker_tag`; update spacing in docs templates (via synth). ([#9765](https://github.com/googleapis/google-cloud-python/pull/9765))
-- Move `speaker_tag` field from `SpeakerDiarizationConfig` to `WordInfo` (via synth). ([#9576](https://github.com/googleapis/google-cloud-python/pull/9576))
-- Remove send/recv msg size limit (via synth). ([#8969](https://github.com/googleapis/google-cloud-python/pull/8969))
-
-### New Features
-- Add speaker diarization configuration support (via synth). ([#9202](https://github.com/googleapis/google-cloud-python/pull/9202))
-- Add `SpeakerDiarizationConfig`, deprecate enable_speaker_diarization and diarization_speaker_count (via synth). ([#8795](https://github.com/googleapis/google-cloud-python/pull/8795))
-
-### Documentation
-- Add python 2 sunset banner to documentation. ([#9036](https://github.com/googleapis/google-cloud-python/pull/9036))
-- Reorder samples. ([#9313](https://github.com/googleapis/google-cloud-python/pull/9313))
-- Update docstrings (via synth). ([#9292](https://github.com/googleapis/google-cloud-python/pull/9292))
-- Fix intersphinx reference to requests. ([#9294](https://github.com/googleapis/google-cloud-python/pull/9294))
-- Add generated code samples. ([#9153](https://github.com/googleapis/google-cloud-python/pull/9153))
-- Remove CI for gh-pages, use googleapis.dev for `api_core` refs. ([#9085](https://github.com/googleapis/google-cloud-python/pull/9085))
-- Remove compatibility badges from READMEs. ([#9035](https://github.com/googleapis/google-cloud-python/pull/9035))
-- Update intersphinx mapping for requests. ([#8805](https://github.com/googleapis/google-cloud-python/pull/8805))
-
-### Internal / Testing Changes
-- Add v1p1beta1 systests for longrunning / streaming recognize. ([#9287](https://github.com/googleapis/google-cloud-python/pull/9287))
-- Add v1 systests for longrunning / streaming recognize. ([#9285](https://github.com/googleapis/google-cloud-python/pull/9285))
-- Update samples manifest (via synth). ([#9211](https://github.com/googleapis/google-cloud-python/pull/9211))
-
-## 1.2.0
-
-07-24-2019 17:35 PDT
-
-### New Features
-- Add 'client_options' support (via synth). ([#8534](https://github.com/googleapis/google-cloud-python/pull/8534))
-
-### Dependencies
-- Bump minimum version for google-api-core to 1.14.0. ([#8709](https://github.com/googleapis/google-cloud-python/pull/8709))
-- Pin black version (via synth). ([#8596](https://github.com/googleapis/google-cloud-python/pull/8596))
-
-### Documentation
-- Link to googleapis.dev documentation in READMEs. ([#8705](https://github.com/googleapis/google-cloud-python/pull/8705))
-- Add compatibility check badges to READMEs. ([#8288](https://github.com/googleapis/google-cloud-python/pull/8288))
-
-## 1.1.0
-
-06-27-2019 16:55 PDT
-
-### Implementation Changes
-- Allow kwargs to be passed to 'create_channel' (via synth). ([#8428](https://github.com/googleapis/google-cloud-python/pull/8428))
-- Remove classifier for Python 3.4 for end-of-life. ([#7535](https://github.com/googleapis/google-cloud-python/pull/7535))
-
-### New Features
-- Increase speech max received msg size to 256 MiB ([#8338](https://github.com/googleapis/google-cloud-python/pull/8338))
-- Add MP3 to Audio Encoding and add boost to Speech Context (via synth). ([#8109](https://github.com/googleapis/google-cloud-python/pull/8109))
-- Add Recognition Metadata (via synth). ([#7961](https://github.com/googleapis/google-cloud-python/pull/7961))
-
-### Documentation
-- Update to show 'google-cloud-speech' is GA. ([#8453](https://github.com/googleapis/google-cloud-python/pull/8453))
-- Tweak 'SpeechContext' docstring (via synth). ([#8223](https://github.com/googleapis/google-cloud-python/pull/8223))
-
-### Internal / Testing Changes
-- All: Add docs job to publish to googleapis.dev. ([#8464](https://github.com/googleapis/google-cloud-python/pull/8464))
-- (Re)-blacken (via synth). ([#8446](https://github.com/googleapis/google-cloud-python/pull/8446))
-- Add disclaimer to auto-generated template files (via synth).  ([#8328](https://github.com/googleapis/google-cloud-python/pull/8328))
-- Suppress checking 'cov-fail-under' in nox default session (via synth).  ([#8252](https://github.com/googleapis/google-cloud-python/pull/8252))
-- Fix coverage in 'types.py' (via synth). ([#8164](https://github.com/googleapis/google-cloud-python/pull/8164))
-- Blacken noxfile.py, setup.py (via synth). ([#8132](https://github.com/googleapis/google-cloud-python/pull/8132))
-- Add empty lines. ([#8072](https://github.com/googleapis/google-cloud-python/pull/8072))
-- Update noxfile (via synth). ([#7836](https://github.com/googleapis/google-cloud-python/pull/7836))
-- Add nox session `docs` (via synth). ([#7782](https://github.com/googleapis/google-cloud-python/pull/7782))
-
-## 1.0.0
-
-03-18-2019 08:05 PDT
-
-
-### Implementation Changes
-- Remove unused message exports. ([#7275](https://github.com/googleapis/google-cloud-python/pull/7275))
-
-### New Features
-- Promote google-cloud-speech to GA ([#7525](https://github.com/googleapis/google-cloud-python/pull/7525))
-
-### Documentation
-- Updated client library documentation URLs. ([#7307](https://github.com/googleapis/google-cloud-python/pull/7307))
-
-### Internal / Testing Changes
-- Speech: copy lintified proto files (via synth).
-- Add clarifying comment to blacken nox target. ([#7404](https://github.com/googleapis/google-cloud-python/pull/7404))
-- Copy proto files alongside protoc versions. Remove unneeded utf-8 header.
-
-## 0.36.3
-
-01-31-2019 09:57 PST
-
-
-### New Features
-- Add 'RecognitionConfig.audio_channel_count' field via synth. ([#7240](https://github.com/googleapis/google-cloud-python/pull/7240))
-
-### Documentation
-- Modify file headers. ([#7158](https://github.com/googleapis/google-cloud-python/pull/7158))
-
-### Internal / Testing Changes
-- Add protos as an artifact to library ([#7205](https://github.com/googleapis/google-cloud-python/pull/7205))
-
-## 0.36.2
-
-01-10-2019 15:36 PST
-
-### Implementation Changes
-- Protoc-generated serialization update. ([#7106](https://github.com/googleapis/google-cloud-python/pull/7106))
-
-### Documentation
-- Regenerate speech to change quote chars in docstr.
-- Pick up stub docstring fix in GAPIC generator. ([#6982](https://github.com/googleapis/google-cloud-python/pull/6982))
-
-## 0.36.1
-
-12-18-2018 09:46 PST
-
-
-### Implementation Changes
-- Import `iam.policy` from `google.api_core`. ([#6741](https://github.com/googleapis/google-cloud-python/pull/6741))
-- Pick up fixes to GAIPC generator. ([#6508](https://github.com/googleapis/google-cloud-python/pull/6508))
-- Add `result_end_time`, docstring changes via synth. ([#6462](https://github.com/googleapis/google-cloud-python/pull/6462))
-- Assorted synth fixups / cleanups ([#6400](https://github.com/googleapis/google-cloud-python/pull/6400))
-- Fix `client_info` bug, update docstrings and timeouts. ([#6421](https://github.com/googleapis/google-cloud-python/pull/6421))
-- Re-generate library using speech/synth.py ([#5979](https://github.com/googleapis/google-cloud-python/pull/5979))
-
-### Dependencies
-- Bump minimum `api_core` version for all GAPIC libs to 1.4.1. ([#6391](https://github.com/googleapis/google-cloud-python/pull/6391))
-
-### Documentation
-- Document Python 2 deprecation ([#6910](https://github.com/googleapis/google-cloud-python/pull/6910))
-- Clarify passed arguments in speech examples. ([#6857](https://github.com/googleapis/google-cloud-python/pull/6857))
-- Docs: normalize use of support level badges ([#6159](https://github.com/googleapis/google-cloud-python/pull/6159))
-- Fix client library URL. ([#6052](https://github.com/googleapis/google-cloud-python/pull/6052))
-- Prep docs for repo split. ([#6017](https://github.com/googleapis/google-cloud-python/pull/6017))
-
-### Internal / Testing Changes
-- Synth.metadata. ([#6868](https://github.com/googleapis/google-cloud-python/pull/6868))
-- Update noxfile.
-- Blacken all gen'd libs ([#6792](https://github.com/googleapis/google-cloud-python/pull/6792))
-- Omit local deps ([#6701](https://github.com/googleapis/google-cloud-python/pull/6701))
-- Run black at end of synth.py ([#6698](https://github.com/googleapis/google-cloud-python/pull/6698))
-- Run Black on Generated libraries ([#6666](https://github.com/googleapis/google-cloud-python/pull/6666))
-- Add templates for flake8, coveragerc, noxfile, and black. ([#6642](https://github.com/googleapis/google-cloud-python/pull/6642))
-- Add / fix badges for PyPI / versions. ([#6158](https://github.com/googleapis/google-cloud-python/pull/6158))
-- Use new Nox ([#6175](https://github.com/googleapis/google-cloud-python/pull/6175))
-
-## 0.36.0
-
-### New Features
-
-- Re-generate the library to pick up changes and new features in the underlying API. ([#5915](https://github.com/GoogleCloudPlatform/google-cloud-python/pull/5915))
-
-### Documentation
-
-- Fix broken links to description of 'Beta' ([#5917](https://github.com/GoogleCloudPlatform/google-cloud-python/pull/5917))
-- Replace links to '/stable/' with '/latest/'. ([#5901](https://github.com/GoogleCloudPlatform/google-cloud-python/pull/5901))
-
-## 0.35.0
-
-### Implementation Changes
-
-- Re-generated the library to pick up new API features. (#5577)
-
-### Internal / Testing Changes
-
-- Add Test runs for Python 3.7 and remove 3.4 (#5295)
-- Avoid overwriting '__module__' of messages from shared modules. (#5364)
-- Modify system tests to use prerelease versions of grpcio (#5304)
-
-## 0.34.0
-
-### Implementation Changes
-- Regenerate GAPIC to account for the removal of GoogleDataCollectionConfig and google_data_collection_opt_in  (#5235)
-
-## 0.33.0
-
-### New Features
-
-- Add Audio Logging and Recognition Metadata. (#5123)
-
-### Internal / Testing Changes
-
-- Fix bad trove classifier
-
-## 0.32.1
-
-### Dependencies
-
-- Update dependency range for api-core to include v1.0.0 releases (#4944)
-
-### Testing and internal changes
-
-- Install local dependencies when running lint (#4936)
-- Re-enable lint for tests, remove usage of pylint (#4921)
-- Normalize all setup.py files (#4909)
-
-## 0.31.1
-
-### Bugfixes
-
-- Fix speech helpers to properly pass retry and timeout args. (#4828, #4830)
-
-## 0.31.0
-
-This is the (hopefully) final release candidate before 1.0.
-
-### Breaking Changes
-
-- The deprecated Speech layer (deprecated since 0.27.0) has been removed. If you are still using  it, the [migration guide](https://cloud.google.com/speech/docs/python-client-migration) is still available.
-- The following changes are _technically_ breaking but very unlikely to affect you directly:
-  * `google.cloud.gapic.speech.v1` moved to `google.cloud.speech_v1.gapic`, in accordance with more recent clients.
-  * `google.cloud.proto.speech.v1` moved to `google.cloud.speech_v1.proto`, in accordance with more recent clients.
-
-### Dependencies
-
-  * Removed dependency on `google-gax`.
-  * Added dependency on `google-api-core`, its replacement.
-
-## 0.30.0
-
-### Documentation
-
-- Added link to "Python Development Environment Setup Guide" in
-  project README (#4187, h/t to @michaelawyu)
-
-### Dependencies
-
-- Upgrading to `google-cloud-core >= 0.28.0` and adding dependency
-  on `google-api-core` (#4221, #4280)
-- Deferring to `google-api-core` for `grpcio` and
-  `googleapis-common-protos`dependencies (#4096, #4098)
-
-PyPI: https://pypi.org/project/google-cloud-speech/0.30.0/
diff --git a/CODE_OF_CONDUCT.md b/CODE_OF_CONDUCT.md
deleted file mode 100644
index b3d1f602..00000000
--- a/CODE_OF_CONDUCT.md
+++ /dev/null
@@ -1,44 +0,0 @@
-<!-- # Generated by synthtool. DO NOT EDIT! !-->
-# Contributor Code of Conduct
-
-As contributors and maintainers of this project,
-and in the interest of fostering an open and welcoming community,
-we pledge to respect all people who contribute through reporting issues,
-posting feature requests, updating documentation,
-submitting pull requests or patches, and other activities.
-
-We are committed to making participation in this project
-a harassment-free experience for everyone,
-regardless of level of experience, gender, gender identity and expression,
-sexual orientation, disability, personal appearance,
-body size, race, ethnicity, age, religion, or nationality.
-
-Examples of unacceptable behavior by participants include:
-
-* The use of sexualized language or imagery
-* Personal attacks
-* Trolling or insulting/derogatory comments
-* Public or private harassment
-* Publishing other's private information,
-such as physical or electronic
-addresses, without explicit permission
-* Other unethical or unprofessional conduct.
-
-Project maintainers have the right and responsibility to remove, edit, or reject
-comments, commits, code, wiki edits, issues, and other contributions
-that are not aligned to this Code of Conduct.
-By adopting this Code of Conduct,
-project maintainers commit themselves to fairly and consistently
-applying these principles to every aspect of managing this project.
-Project maintainers who do not follow or enforce the Code of Conduct
-may be permanently removed from the project team.
-
-This code of conduct applies both within project spaces and in public spaces
-when an individual is representing the project or its community.
-
-Instances of abusive, harassing, or otherwise unacceptable behavior
-may be reported by opening an issue
-or contacting one or more of the project maintainers.
-
-This Code of Conduct is adapted from the [Contributor Covenant](http://contributor-covenant.org), version 1.2.0,
-available at [http://contributor-covenant.org/version/1/2/0/](http://contributor-covenant.org/version/1/2/0/)
diff --git a/CONTRIBUTING.rst b/CONTRIBUTING.rst
deleted file mode 100644
index c37e4a6c..00000000
--- a/CONTRIBUTING.rst
+++ /dev/null
@@ -1,279 +0,0 @@
-.. Generated by synthtool. DO NOT EDIT!
-############
-Contributing
-############
-
-#. **Please sign one of the contributor license agreements below.**
-#. Fork the repo, develop and test your code changes, add docs.
-#. Make sure that your commit messages clearly describe the changes.
-#. Send a pull request. (Please Read: `Faster Pull Request Reviews`_)
-
-.. _Faster Pull Request Reviews: https://github.com/kubernetes/community/blob/master/contributors/guide/pull-requests.md#best-practices-for-faster-reviews
-
-.. contents:: Here are some guidelines for hacking on the Google Cloud Client libraries.
-
-***************
-Adding Features
-***************
-
-In order to add a feature:
-
-- The feature must be documented in both the API and narrative
-  documentation.
-
-- The feature must work fully on the following CPython versions:  2.7,
-  3.5, 3.6, and 3.7 on both UNIX and Windows.
-
-- The feature must not add unnecessary dependencies (where
-  "unnecessary" is of course subjective, but new dependencies should
-  be discussed).
-
-****************************
-Using a Development Checkout
-****************************
-
-You'll have to create a development environment using a Git checkout:
-
-- While logged into your GitHub account, navigate to the
-  ``python-speech`` `repo`_ on GitHub.
-
-- Fork and clone the ``python-speech`` repository to your GitHub account by
-  clicking the "Fork" button.
-
-- Clone your fork of ``python-speech`` from your GitHub account to your local
-  computer, substituting your account username and specifying the destination
-  as ``hack-on-python-speech``.  E.g.::
-
-   $ cd ${HOME}
-   $ git clone git@github.com:USERNAME/python-speech.git hack-on-python-speech
-   $ cd hack-on-python-speech
-   # Configure remotes such that you can pull changes from the googleapis/python-speech
-   # repository into your local repository.
-   $ git remote add upstream git@github.com:googleapis/python-speech.git
-   # fetch and merge changes from upstream into master
-   $ git fetch upstream
-   $ git merge upstream/master
-
-Now your local repo is set up such that you will push changes to your GitHub
-repo, from which you can submit a pull request.
-
-To work on the codebase and run the tests, we recommend using ``nox``,
-but you can also use a ``virtualenv`` of your own creation.
-
-.. _repo: https://github.com/googleapis/python-speech
-
-Using ``nox``
-=============
-
-We use `nox <https://nox.readthedocs.io/en/latest/>`__ to instrument our tests.
-
-- To test your changes, run unit tests with ``nox``::
-
-    $ nox -s unit-2.7
-    $ nox -s unit-3.7
-    $ ...
-
-  .. note::
-
-    The unit tests and system tests are described in the
-    ``noxfile.py`` files in each directory.
-
-.. nox: https://pypi.org/project/nox/
-
-Note on Editable Installs / Develop Mode
-========================================
-
-- As mentioned previously, using ``setuptools`` in `develop mode`_
-  or a ``pip`` `editable install`_ is not possible with this
-  library. This is because this library uses `namespace packages`_.
-  For context see `Issue #2316`_ and the relevant `PyPA issue`_.
-
-  Since ``editable`` / ``develop`` mode can't be used, packages
-  need to be installed directly. Hence your changes to the source
-  tree don't get incorporated into the **already installed**
-  package.
-
-.. _namespace packages: https://www.python.org/dev/peps/pep-0420/
-.. _Issue #2316: https://github.com/GoogleCloudPlatform/google-cloud-python/issues/2316
-.. _PyPA issue: https://github.com/pypa/packaging-problems/issues/12
-.. _develop mode: https://setuptools.readthedocs.io/en/latest/setuptools.html#development-mode
-.. _editable install: https://pip.pypa.io/en/stable/reference/pip_install/#editable-installs
-
-*****************************************
-I'm getting weird errors... Can you help?
-*****************************************
-
-If the error mentions ``Python.h`` not being found,
-install ``python-dev`` and try again.
-On Debian/Ubuntu::
-
-  $ sudo apt-get install python-dev
-
-************
-Coding Style
-************
-
-- PEP8 compliance, with exceptions defined in the linter configuration.
-  If you have ``nox`` installed, you can test that you have not introduced
-  any non-compliant code via::
-
-   $ nox -s lint
-
-- In order to make ``nox -s lint`` run faster, you can set some environment
-  variables::
-
-   export GOOGLE_CLOUD_TESTING_REMOTE="upstream"
-   export GOOGLE_CLOUD_TESTING_BRANCH="master"
-
-  By doing this, you are specifying the location of the most up-to-date
-  version of ``python-speech``. The the suggested remote name ``upstream``
-  should point to the official ``googleapis`` checkout and the
-  the branch should be the main branch on that remote (``master``).
-
-Exceptions to PEP8:
-
-- Many unit tests use a helper method, ``_call_fut`` ("FUT" is short for
-  "Function-Under-Test"), which is PEP8-incompliant, but more readable.
-  Some also use a local variable, ``MUT`` (short for "Module-Under-Test").
-
-********************
-Running System Tests
-********************
-
-- To run system tests, you can execute::
-
-   $ nox -s system-3.7
-   $ nox -s system-2.7
-
-  .. note::
-
-      System tests are only configured to run under Python 2.7 and
-      Python 3.7. For expediency, we do not run them in older versions
-      of Python 3.
-
-  This alone will not run the tests. You'll need to change some local
-  auth settings and change some configuration in your project to
-  run all the tests.
-
-- System tests will be run against an actual project and
-  so you'll need to provide some environment variables to facilitate
-  authentication to your project:
-
-  - ``GOOGLE_APPLICATION_CREDENTIALS``: The path to a JSON key file;
-    Such a file can be downloaded directly from the developer's console by clicking
-    "Generate new JSON key". See private key
-    `docs <https://cloud.google.com/storage/docs/authentication#generating-a-private-key>`__
-    for more details.
-
-- Once you have downloaded your json keys, set the environment variable 
-  ``GOOGLE_APPLICATION_CREDENTIALS`` to the absolute path of the json file::
-
-   $ export GOOGLE_APPLICATION_CREDENTIALS="/Users/<your_username>/path/to/app_credentials.json"
-
-
-*************
-Test Coverage
-*************
-
-- The codebase *must* have 100% test statement coverage after each commit.
-  You can test coverage via ``nox -s cover``.
-
-******************************************************
-Documentation Coverage and Building HTML Documentation
-******************************************************
-
-If you fix a bug, and the bug requires an API or behavior modification, all
-documentation in this package which references that API or behavior must be
-changed to reflect the bug fix, ideally in the same commit that fixes the bug
-or adds the feature.
-
-Build the docs via:
-
-   $ nox -s docs
-
-********************************************
-Note About ``README`` as it pertains to PyPI
-********************************************
-
-The `description on PyPI`_ for the project comes directly from the
-``README``. Due to the reStructuredText (``rst``) parser used by
-PyPI, relative links which will work on GitHub (e.g. ``CONTRIBUTING.rst``
-instead of
-``https://github.com/googleapis/python-speech/blob/master/CONTRIBUTING.rst``)
-may cause problems creating links or rendering the description.
-
-.. _description on PyPI: https://pypi.org/project/google-cloud-speech
-
-
-*************************
-Supported Python Versions
-*************************
-
-We support:
-
--  `Python 3.5`_
--  `Python 3.6`_
--  `Python 3.7`_
-
-.. _Python 3.5: https://docs.python.org/3.5/
-.. _Python 3.6: https://docs.python.org/3.6/
-.. _Python 3.7: https://docs.python.org/3.7/
-
-
-Supported versions can be found in our ``noxfile.py`` `config`_.
-
-.. _config: https://github.com/googleapis/python-speech/blob/master/noxfile.py
-
-We explicitly decided not to support `Python 2.5`_ due to `decreased usage`_
-and lack of continuous integration `support`_.
-
-.. _Python 2.5: https://docs.python.org/2.5/
-.. _decreased usage: https://caremad.io/2013/10/a-look-at-pypi-downloads/
-.. _support: https://blog.travis-ci.com/2013-11-18-upcoming-build-environment-updates/
-
-We have `dropped 2.6`_ as a supported version as well since Python 2.6 is no
-longer supported by the core development team.
-
-Python 2.7 support is deprecated. All code changes should maintain Python 2.7 compatibility until January 1, 2020.
-
-We also explicitly decided to support Python 3 beginning with version
-3.5. Reasons for this include:
-
--  Encouraging use of newest versions of Python 3
--  Taking the lead of `prominent`_ open-source `projects`_
--  `Unicode literal support`_ which allows for a cleaner codebase that
-   works in both Python 2 and Python 3
-
-.. _prominent: https://docs.djangoproject.com/en/1.9/faq/install/#what-python-version-can-i-use-with-django
-.. _projects: http://flask.pocoo.org/docs/0.10/python3/
-.. _Unicode literal support: https://www.python.org/dev/peps/pep-0414/
-.. _dropped 2.6: https://github.com/googleapis/google-cloud-python/issues/995
-
-**********
-Versioning
-**********
-
-This library follows `Semantic Versioning`_.
-
-.. _Semantic Versioning: http://semver.org/
-
-Some packages are currently in major version zero (``0.y.z``), which means that
-anything may change at any time and the public API should not be considered
-stable.
-
-******************************
-Contributor License Agreements
-******************************
-
-Before we can accept your pull requests you'll need to sign a Contributor
-License Agreement (CLA):
-
-- **If you are an individual writing original source code** and **you own the
-  intellectual property**, then you'll need to sign an
-  `individual CLA <https://developers.google.com/open-source/cla/individual>`__.
-- **If you work for a company that wants to allow you to contribute your work**,
-  then you'll need to sign a
-  `corporate CLA <https://developers.google.com/open-source/cla/corporate>`__.
-
-You can sign these electronically (just scroll to the bottom). After that,
-we'll be able to accept your pull requests.
diff --git a/LICENSE b/LICENSE
index a8ee855d..d6456956 100644
--- a/LICENSE
+++ b/LICENSE
@@ -1,6 +1,7 @@
-                            Apache License
+
+                                 Apache License
                            Version 2.0, January 2004
-                        https://www.apache.org/licenses/
+                        http://www.apache.org/licenses/
 
    TERMS AND CONDITIONS FOR USE, REPRODUCTION, AND DISTRIBUTION
 
@@ -192,7 +193,7 @@
    you may not use this file except in compliance with the License.
    You may obtain a copy of the License at
 
-       https://www.apache.org/licenses/LICENSE-2.0
+       http://www.apache.org/licenses/LICENSE-2.0
 
    Unless required by applicable law or agreed to in writing, software
    distributed under the License is distributed on an "AS IS" BASIS,
diff --git a/MANIFEST.in b/MANIFEST.in
deleted file mode 100644
index cd011be2..00000000
--- a/MANIFEST.in
+++ /dev/null
@@ -1,6 +0,0 @@
-# Generated by synthtool. DO NOT EDIT!
-include README.rst LICENSE
-recursive-include google *.json *.proto
-recursive-include tests *
-global-exclude *.py[co]
-global-exclude __pycache__
diff --git a/README.rst b/README.rst
index 3cdf9fda..f92b8bbb 100644
--- a/README.rst
+++ b/README.rst
@@ -1,25 +1,27 @@
+:**NOTE**: **This github repository is archived. The repository contents and history have moved to** `google-cloud-python`_.
+
+.. _google-cloud-python: https://github.com/googleapis/google-cloud-python/tree/main/packages/google-cloud-speech
+
+
 Python Client for Cloud Speech API
 ==================================
 
+|stable| |pypi| |versions|
 
-|GA| |pypi| |versions| 
-
-The `Cloud Speech API`_ enables developers to convert audio to text by applying
-powerful neural network models.  The API recognizes over 80 languages and
-variants, to support your global user base.
+`Cloud Speech API`_: enables easy integration of Google speech recognition technologies into developer applications. Send audio and receive a text transcription from the Speech-to-Text API service.
 
 - `Client Library Documentation`_
 - `Product Documentation`_
 
-.. |GA| image:: https://img.shields.io/badge/support-GA-gold.svg
-   :target: https://github.com/googleapis/google-cloud-python/blob/master/README.rst#general-availability
+.. |stable| image:: https://img.shields.io/badge/support-stable-gold.svg
+   :target: https://github.com/googleapis/google-cloud-python/blob/main/README.rst#stability-levels
 .. |pypi| image:: https://img.shields.io/pypi/v/google-cloud-speech.svg
    :target: https://pypi.org/project/google-cloud-speech/
 .. |versions| image:: https://img.shields.io/pypi/pyversions/google-cloud-speech.svg
    :target: https://pypi.org/project/google-cloud-speech/
-.. _Cloud Speech API: https://cloud.google.com/speech
-.. _Client Library Documentation: https://googleapis.dev/python/speech/latest
-.. _Product Documentation:  https://cloud.google.com/speech
+.. _Cloud Speech API: https://cloud.google.com/speech-to-text/docs/
+.. _Client Library Documentation: https://cloud.google.com/python/docs/reference/speech/latest
+.. _Product Documentation:  https://cloud.google.com/speech-to-text/docs/
 
 Quick Start
 -----------
@@ -33,41 +35,59 @@ In order to use this library, you first need to go through the following steps:
 
 .. _Select or create a Cloud Platform project.: https://console.cloud.google.com/project
 .. _Enable billing for your project.: https://cloud.google.com/billing/docs/how-to/modify-project#enable_billing_for_a_project
-.. _Enable the Cloud Speech API.:  https://cloud.google.com/speech
+.. _Enable the Cloud Speech API.:  https://cloud.google.com/speech-to-text/docs/
 .. _Setup Authentication.: https://googleapis.dev/python/google-api-core/latest/auth.html
 
 Installation
 ~~~~~~~~~~~~
 
-Install this library in a `virtualenv`_ using pip. `virtualenv`_ is a tool to
-create isolated Python environments. The basic problem it addresses is one of
-dependencies and versions, and indirectly permissions.
+Install this library in a virtual environment using `venv`_. `venv`_ is a tool that
+creates isolated Python environments. These isolated environments can have separate
+versions of Python packages, which allows you to isolate one project's dependencies
+from the dependencies of other projects.
 
-With `virtualenv`_, it's possible to install this library without needing system
+With `venv`_, it's possible to install this library without needing system
 install permissions, and without clashing with the installed system
 dependencies.
 
-.. _`virtualenv`: https://virtualenv.pypa.io/en/latest/
+.. _`venv`: https://docs.python.org/3/library/venv.html
+
+
+Code samples and snippets
+~~~~~~~~~~~~~~~~~~~~~~~~~
+
+Code samples and snippets live in the `samples/`_ folder.
+
+.. _samples/: https://github.com/googleapis/python-speech/tree/main/samples
 
 
 Supported Python Versions
 ^^^^^^^^^^^^^^^^^^^^^^^^^
-Python >= 3.5
+Our client libraries are compatible with all current `active`_ and `maintenance`_ versions of
+Python.
+
+Python >= 3.7
 
-Deprecated Python Versions
-^^^^^^^^^^^^^^^^^^^^^^^^^^
-Python == 2.7. Python 2.7 support will be removed on January 1, 2020.
+.. _active: https://devguide.python.org/devcycle/#in-development-main-branch
+.. _maintenance: https://devguide.python.org/devcycle/#maintenance-branches
 
+Unsupported Python Versions
+^^^^^^^^^^^^^^^^^^^^^^^^^^^
+Python <= 3.6
+
+If you are using an `end-of-life`_
+version of Python, we recommend that you update as soon as possible to an actively supported version.
+
+.. _end-of-life: https://devguide.python.org/devcycle/#end-of-life-branches
 
 Mac/Linux
 ^^^^^^^^^
 
 .. code-block:: console
 
-    pip install virtualenv
-    virtualenv <your-env>
+    python3 -m venv <your-env>
     source <your-env>/bin/activate
-    <your-env>/bin/pip install google-cloud-speech
+    pip install google-cloud-speech
 
 
 Windows
@@ -75,35 +95,19 @@ Windows
 
 .. code-block:: console
 
-    pip install virtualenv
-    virtualenv <your-env>
-    <your-env>\Scripts\activate
-    <your-env>\Scripts\pip.exe install google-cloud-speech
-
-Example Usage
-~~~~~~~~~~~~~
-
-.. code:: py
-
-    from google.cloud import speech_v1
-    from google.cloud.speech_v1 import enums
-
-    client = speech_v1.SpeechClient()
-
-    encoding = enums.RecognitionConfig.AudioEncoding.FLAC
-    sample_rate_hertz = 44100
-    language_code = 'en-US'
-    config = {'encoding': encoding, 'sample_rate_hertz': sample_rate_hertz, 'language_code': language_code}
-    uri = 'gs://bucket_name/file_name.flac'
-    audio = {'uri': uri}
-
-    response = client.recognize(config, audio)
+    py -m venv <your-env>
+    .\<your-env>\Scripts\activate
+    pip install google-cloud-speech
 
 Next Steps
 ~~~~~~~~~~
 
 -  Read the `Client Library Documentation`_ for Cloud Speech API
-   API to see other available methods on the client.
--  Read the `Product documentation`_ to learn
+   to see other available methods on the client.
+-  Read the `Cloud Speech API Product documentation`_ to learn
    more about the product and see How-to Guides.
+-  View this `README`_ to see the full list of Cloud
    APIs that we cover.
+
+.. _Cloud Speech API Product documentation:  https://cloud.google.com/speech-to-text/docs/
+.. _README: https://github.com/googleapis/google-cloud-python/blob/main/README.rst
diff --git a/SECURITY.md b/SECURITY.md
new file mode 100644
index 00000000..8b58ae9c
--- /dev/null
+++ b/SECURITY.md
@@ -0,0 +1,7 @@
+# Security Policy
+
+To report a security issue, please use [g.co/vulnz](https://g.co/vulnz).
+
+The Google Security Team will respond within 5 working days of your report on g.co/vulnz.
+
+We use g.co/vulnz for our intake, and do coordination and disclosure here using GitHub Security Advisory to privately discuss and fix the issue.
diff --git a/docs/README.rst b/docs/README.rst
deleted file mode 120000
index 89a01069..00000000
--- a/docs/README.rst
+++ /dev/null
@@ -1 +0,0 @@
-../README.rst
\ No newline at end of file
diff --git a/docs/_static/custom.css b/docs/_static/custom.css
deleted file mode 100644
index 0abaf229..00000000
--- a/docs/_static/custom.css
+++ /dev/null
@@ -1,4 +0,0 @@
-div#python2-eol {
-	border-color: red;
-	border-width: medium;
-} 
\ No newline at end of file
diff --git a/docs/_templates/layout.html b/docs/_templates/layout.html
deleted file mode 100644
index 228529ef..00000000
--- a/docs/_templates/layout.html
+++ /dev/null
@@ -1,50 +0,0 @@
-
-{% extends "!layout.html" %}
-{%- block content %}
-{%- if theme_fixed_sidebar|lower == 'true' %}
-  <div class="document">
-    {{ sidebar() }}
-    {%- block document %}
-      <div class="documentwrapper">
-      {%- if render_sidebar %}
-        <div class="bodywrapper">
-      {%- endif %}
-
-          {%- block relbar_top %}
-            {%- if theme_show_relbar_top|tobool %}
-              <div class="related top">
-                &nbsp;
-                {{- rellink_markup () }}
-              </div>
-            {%- endif %}
-          {% endblock %}
-
-          <div class="body" role="main">
-          	<div class="admonition" id="python2-eol"> 
-          	 On January 1, 2020 this library will no longer support Python 2 on the latest released version. 
-          	 Previously released library versions will continue to be available. For more information please
-          	 visit <a href="https://cloud.google.com/python/docs/python2-sunset/">Python 2 support on Google Cloud</a>.
-          	</div>
-            {% block body %} {% endblock %}
-          </div>
-
-          {%- block relbar_bottom %}
-            {%- if theme_show_relbar_bottom|tobool %}
-              <div class="related bottom">
-                &nbsp;
-                {{- rellink_markup () }}
-              </div>
-            {%- endif %}
-          {% endblock %}
-
-      {%- if render_sidebar %}
-        </div>
-      {%- endif %}
-      </div>
-    {%- endblock %}
-    <div class="clearer"></div>
-  </div>
-{%- else %}
-{{ super() }}
-{%- endif %}
-{%- endblock %}
diff --git a/docs/changelog.md b/docs/changelog.md
deleted file mode 120000
index 04c99a55..00000000
--- a/docs/changelog.md
+++ /dev/null
@@ -1 +0,0 @@
-../CHANGELOG.md
\ No newline at end of file
diff --git a/docs/conf.py b/docs/conf.py
deleted file mode 100644
index be3e6aa6..00000000
--- a/docs/conf.py
+++ /dev/null
@@ -1,358 +0,0 @@
-# -*- coding: utf-8 -*-
-#
-# google-cloud-speech documentation build configuration file
-#
-# This file is execfile()d with the current directory set to its
-# containing dir.
-#
-# Note that not all possible configuration values are present in this
-# autogenerated file.
-#
-# All configuration values have a default; values that are commented out
-# serve to show the default.
-
-import sys
-import os
-import shlex
-
-# If extensions (or modules to document with autodoc) are in another directory,
-# add these directories to sys.path here. If the directory is relative to the
-# documentation root, use os.path.abspath to make it absolute, like shown here.
-sys.path.insert(0, os.path.abspath(".."))
-
-__version__ = ""
-
-# -- General configuration ------------------------------------------------
-
-# If your documentation needs a minimal Sphinx version, state it here.
-needs_sphinx = "1.6.3"
-
-# Add any Sphinx extension module names here, as strings. They can be
-# extensions coming with Sphinx (named 'sphinx.ext.*') or your custom
-# ones.
-extensions = [
-    "sphinx.ext.autodoc",
-    "sphinx.ext.autosummary",
-    "sphinx.ext.intersphinx",
-    "sphinx.ext.coverage",
-    "sphinx.ext.napoleon",
-    "sphinx.ext.todo",
-    "sphinx.ext.viewcode",
-]
-
-# autodoc/autosummary flags
-autoclass_content = "both"
-autodoc_default_flags = ["members"]
-autosummary_generate = True
-
-
-# Add any paths that contain templates here, relative to this directory.
-templates_path = ["_templates"]
-
-# Allow markdown includes (so releases.md can include CHANGLEOG.md)
-# http://www.sphinx-doc.org/en/master/markdown.html
-source_parsers = {".md": "recommonmark.parser.CommonMarkParser"}
-
-# The suffix(es) of source filenames.
-# You can specify multiple suffix as a list of string:
-# source_suffix = ['.rst', '.md']
-source_suffix = [".rst", ".md"]
-
-# The encoding of source files.
-# source_encoding = 'utf-8-sig'
-
-# The master toctree document.
-master_doc = "index"
-
-# General information about the project.
-project = u"google-cloud-speech"
-copyright = u"2019, Google"
-author = u"Google APIs"
-
-# The version info for the project you're documenting, acts as replacement for
-# |version| and |release|, also used in various other places throughout the
-# built documents.
-#
-# The full version, including alpha/beta/rc tags.
-release = __version__
-# The short X.Y version.
-version = ".".join(release.split(".")[0:2])
-
-# The language for content autogenerated by Sphinx. Refer to documentation
-# for a list of supported languages.
-#
-# This is also used if you do content translation via gettext catalogs.
-# Usually you set "language" from the command line for these cases.
-language = None
-
-# There are two options for replacing |today|: either, you set today to some
-# non-false value, then it is used:
-# today = ''
-# Else, today_fmt is used as the format for a strftime call.
-# today_fmt = '%B %d, %Y'
-
-# List of patterns, relative to source directory, that match files and
-# directories to ignore when looking for source files.
-exclude_patterns = ["_build"]
-
-# The reST default role (used for this markup: `text`) to use for all
-# documents.
-# default_role = None
-
-# If true, '()' will be appended to :func: etc. cross-reference text.
-# add_function_parentheses = True
-
-# If true, the current module name will be prepended to all description
-# unit titles (such as .. function::).
-# add_module_names = True
-
-# If true, sectionauthor and moduleauthor directives will be shown in the
-# output. They are ignored by default.
-# show_authors = False
-
-# The name of the Pygments (syntax highlighting) style to use.
-pygments_style = "sphinx"
-
-# A list of ignored prefixes for module index sorting.
-# modindex_common_prefix = []
-
-# If true, keep warnings as "system message" paragraphs in the built documents.
-# keep_warnings = False
-
-# If true, `todo` and `todoList` produce output, else they produce nothing.
-todo_include_todos = True
-
-
-# -- Options for HTML output ----------------------------------------------
-
-# The theme to use for HTML and HTML Help pages.  See the documentation for
-# a list of builtin themes.
-html_theme = "alabaster"
-
-# Theme options are theme-specific and customize the look and feel of a theme
-# further.  For a list of options available for each theme, see the
-# documentation.
-html_theme_options = {
-    "description": "Google Cloud Client Libraries for google-cloud-speech",
-    "github_user": "googleapis",
-    "github_repo": "python-speech",
-    "github_banner": True,
-    "font_family": "'Roboto', Georgia, sans",
-    "head_font_family": "'Roboto', Georgia, serif",
-    "code_font_family": "'Roboto Mono', 'Consolas', monospace",
-}
-
-# Add any paths that contain custom themes here, relative to this directory.
-# html_theme_path = []
-
-# The name for this set of Sphinx documents.  If None, it defaults to
-# "<project> v<release> documentation".
-# html_title = None
-
-# A shorter title for the navigation bar.  Default is the same as html_title.
-# html_short_title = None
-
-# The name of an image file (relative to this directory) to place at the top
-# of the sidebar.
-# html_logo = None
-
-# The name of an image file (within the static path) to use as favicon of the
-# docs.  This file should be a Windows icon file (.ico) being 16x16 or 32x32
-# pixels large.
-# html_favicon = None
-
-# Add any paths that contain custom static files (such as style sheets) here,
-# relative to this directory. They are copied after the builtin static files,
-# so a file named "default.css" will overwrite the builtin "default.css".
-html_static_path = ["_static"]
-
-# Add any extra paths that contain custom files (such as robots.txt or
-# .htaccess) here, relative to this directory. These files are copied
-# directly to the root of the documentation.
-# html_extra_path = []
-
-# If not '', a 'Last updated on:' timestamp is inserted at every page bottom,
-# using the given strftime format.
-# html_last_updated_fmt = '%b %d, %Y'
-
-# If true, SmartyPants will be used to convert quotes and dashes to
-# typographically correct entities.
-# html_use_smartypants = True
-
-# Custom sidebar templates, maps document names to template names.
-# html_sidebars = {}
-
-# Additional templates that should be rendered to pages, maps page names to
-# template names.
-# html_additional_pages = {}
-
-# If false, no module index is generated.
-# html_domain_indices = True
-
-# If false, no index is generated.
-# html_use_index = True
-
-# If true, the index is split into individual pages for each letter.
-# html_split_index = False
-
-# If true, links to the reST sources are added to the pages.
-# html_show_sourcelink = True
-
-# If true, "Created using Sphinx" is shown in the HTML footer. Default is True.
-# html_show_sphinx = True
-
-# If true, "(C) Copyright ..." is shown in the HTML footer. Default is True.
-# html_show_copyright = True
-
-# If true, an OpenSearch description file will be output, and all pages will
-# contain a <link> tag referring to it.  The value of this option must be the
-# base URL from which the finished HTML is served.
-# html_use_opensearch = ''
-
-# This is the file name suffix for HTML files (e.g. ".xhtml").
-# html_file_suffix = None
-
-# Language to be used for generating the HTML full-text search index.
-# Sphinx supports the following languages:
-#   'da', 'de', 'en', 'es', 'fi', 'fr', 'hu', 'it', 'ja'
-#   'nl', 'no', 'pt', 'ro', 'ru', 'sv', 'tr'
-# html_search_language = 'en'
-
-# A dictionary with options for the search language support, empty by default.
-# Now only 'ja' uses this config value
-# html_search_options = {'type': 'default'}
-
-# The name of a javascript file (relative to the configuration directory) that
-# implements a search results scorer. If empty, the default will be used.
-# html_search_scorer = 'scorer.js'
-
-# Output file base name for HTML help builder.
-htmlhelp_basename = "google-cloud-speech-doc"
-
-# -- Options for warnings ------------------------------------------------------
-
-
-suppress_warnings = [
-    # Temporarily suppress this to avoid "more than one target found for
-    # cross-reference" warning, which are intractable for us to avoid while in
-    # a mono-repo.
-    # See https://github.com/sphinx-doc/sphinx/blob
-    # /2a65ffeef5c107c19084fabdd706cdff3f52d93c/sphinx/domains/python.py#L843
-    "ref.python"
-]
-
-# -- Options for LaTeX output ---------------------------------------------
-
-latex_elements = {
-    # The paper size ('letterpaper' or 'a4paper').
-    #'papersize': 'letterpaper',
-    # The font size ('10pt', '11pt' or '12pt').
-    #'pointsize': '10pt',
-    # Additional stuff for the LaTeX preamble.
-    #'preamble': '',
-    # Latex figure (float) alignment
-    #'figure_align': 'htbp',
-}
-
-# Grouping the document tree into LaTeX files. List of tuples
-# (source start file, target name, title,
-#  author, documentclass [howto, manual, or own class]).
-latex_documents = [
-    (
-        master_doc,
-        "google-cloud-speech.tex",
-        u"google-cloud-speech Documentation",
-        author,
-        "manual",
-    )
-]
-
-# The name of an image file (relative to this directory) to place at the top of
-# the title page.
-# latex_logo = None
-
-# For "manual" documents, if this is true, then toplevel headings are parts,
-# not chapters.
-# latex_use_parts = False
-
-# If true, show page references after internal links.
-# latex_show_pagerefs = False
-
-# If true, show URL addresses after external links.
-# latex_show_urls = False
-
-# Documents to append as an appendix to all manuals.
-# latex_appendices = []
-
-# If false, no module index is generated.
-# latex_domain_indices = True
-
-
-# -- Options for manual page output ---------------------------------------
-
-# One entry per manual page. List of tuples
-# (source start file, name, description, authors, manual section).
-man_pages = [
-    (
-        master_doc,
-        "google-cloud-speech",
-        u"google-cloud-speech Documentation",
-        [author],
-        1,
-    )
-]
-
-# If true, show URL addresses after external links.
-# man_show_urls = False
-
-
-# -- Options for Texinfo output -------------------------------------------
-
-# Grouping the document tree into Texinfo files. List of tuples
-# (source start file, target name, title, author,
-#  dir menu entry, description, category)
-texinfo_documents = [
-    (
-        master_doc,
-        "google-cloud-speech",
-        u"google-cloud-speech Documentation",
-        author,
-        "google-cloud-speech",
-        "google-cloud-speech Library",
-        "APIs",
-    )
-]
-
-# Documents to append as an appendix to all manuals.
-# texinfo_appendices = []
-
-# If false, no module index is generated.
-# texinfo_domain_indices = True
-
-# How to display URL addresses: 'footnote', 'no', or 'inline'.
-# texinfo_show_urls = 'footnote'
-
-# If true, do not generate a @detailmenu in the "Top" node's menu.
-# texinfo_no_detailmenu = False
-
-
-# Example configuration for intersphinx: refer to the Python standard library.
-intersphinx_mapping = {
-    "python": ("http://python.readthedocs.org/en/latest/", None),
-    "google-auth": ("https://google-auth.readthedocs.io/en/stable", None),
-    "google.api_core": ("https://googleapis.dev/python/google-api-core/latest/", None),
-    "grpc": ("https://grpc.io/grpc/python/", None),
-}
-
-
-# Napoleon settings
-napoleon_google_docstring = True
-napoleon_numpy_docstring = True
-napoleon_include_private_with_doc = False
-napoleon_include_special_with_doc = True
-napoleon_use_admonition_for_examples = False
-napoleon_use_admonition_for_notes = False
-napoleon_use_admonition_for_references = False
-napoleon_use_ivar = False
-napoleon_use_param = True
-napoleon_use_rtype = True
diff --git a/docs/gapic/v1/api.rst b/docs/gapic/v1/api.rst
deleted file mode 100644
index a04f18ad..00000000
--- a/docs/gapic/v1/api.rst
+++ /dev/null
@@ -1,6 +0,0 @@
-Client for Cloud Speech-to-Text API
-===================================
-
-.. automodule:: google.cloud.speech_v1
-    :members:
-    :inherited-members:
\ No newline at end of file
diff --git a/docs/gapic/v1/types.rst b/docs/gapic/v1/types.rst
deleted file mode 100644
index 71a61184..00000000
--- a/docs/gapic/v1/types.rst
+++ /dev/null
@@ -1,5 +0,0 @@
-Types for Cloud Speech-to-Text API Client
-=========================================
-
-.. automodule:: google.cloud.speech_v1.types
-    :members:
\ No newline at end of file
diff --git a/docs/gapic/v1p1beta1/api.rst b/docs/gapic/v1p1beta1/api.rst
deleted file mode 100644
index 9493c970..00000000
--- a/docs/gapic/v1p1beta1/api.rst
+++ /dev/null
@@ -1,6 +0,0 @@
-Client for Cloud Speech-to-Text API
-===================================
-
-.. automodule:: google.cloud.speech_v1p1beta1
-    :members:
-    :inherited-members:
\ No newline at end of file
diff --git a/docs/gapic/v1p1beta1/types.rst b/docs/gapic/v1p1beta1/types.rst
deleted file mode 100644
index 6ce93d56..00000000
--- a/docs/gapic/v1p1beta1/types.rst
+++ /dev/null
@@ -1,5 +0,0 @@
-Types for Cloud Speech-to-Text API Client
-=========================================
-
-.. automodule:: google.cloud.speech_v1p1beta1.types
-    :members:
\ No newline at end of file
diff --git a/docs/index.rst b/docs/index.rst
deleted file mode 100644
index 02f947a7..00000000
--- a/docs/index.rst
+++ /dev/null
@@ -1,282 +0,0 @@
-.. include:: README.rst
-
-Using the Library
------------------
-
-Asynchronous Recognition
-~~~~~~~~~~~~~~~~~~~~~~~~
-
-The :meth:`~.speech_v1.SpeechClient.long_running_recognize` method
-sends audio data to the Speech API and initiates a Long Running Operation.
-
-Using this operation, you can periodically poll for recognition results.
-Use asynchronous requests for audio data of any duration up to 80 minutes.
-
-See: `Speech Asynchronous Recognize`_
-
-
-.. code-block:: python
-
-    >>> from google.cloud import speech
-    >>> client = speech.SpeechClient()
-    >>> audio = speech.types.RecognitionAudio(
-    ...     uri='gs://my-bucket/recording.flac')
-    >>> config = speech.types.RecognitionConfig(
-    ...     encoding=speech.enums.RecognitionConfig.AudioEncoding.LINEAR16,
-    ...     language_code='en-US',
-    ...     sample_rate_hertz=44100)
-    >>> operation = client.long_running_recognize(config=config, audio=audio)
-    >>> op_result = operation.result()
-    >>> for result in op_result.results:
-    ...     for alternative in result.alternatives:
-    ...         print('=' * 20)
-    ...         print(alternative.transcript)
-    ...         print(alternative.confidence)
-    ====================
-    'how old is the Brooklyn Bridge'
-    0.98267895
-
-
-Synchronous Recognition
-~~~~~~~~~~~~~~~~~~~~~~~
-
-The :meth:`~.speech_v1.SpeechClient.recognize` method converts speech
-data to text and returns alternative text transcriptions.
-
-This example uses ``language_code='en-GB'`` to better recognize a dialect from
-Great Britain.
-
-.. code-block:: python
-
-    >>> from google.cloud import speech
-    >>> client = speech.SpeechClient()
-    >>> audio = speech.types.RecognitionAudio(
-    ...     uri='gs://my-bucket/recording.flac')
-    >>> config = speech.types.RecognitionConfig(
-    ...     encoding=speech.enums.RecognitionConfig.AudioEncoding.LINEAR16,
-    ...     language_code='en-US',
-    ...     sample_rate_hertz=44100)
-    >>> results = client.recognize(config=config, audio=audio)
-    >>> for result in results:
-    ...     for alternative in result.alternatives:
-    ...         print('=' * 20)
-    ...         print('transcript: ' + alternative.transcript)
-    ...         print('confidence: ' + str(alternative.confidence))
-    ====================
-    transcript: Hello, this is a test
-    confidence: 0.81
-    ====================
-    transcript: Hello, this is one test
-    confidence: 0
-
-Example of using the profanity filter.
-
-.. code-block:: python
-
-    >>> from google.cloud import speech
-    >>> client = speech.SpeechClient()
-    >>> audio = speech.types.RecognitionAudio(
-    ...     uri='gs://my-bucket/recording.flac')
-    >>> config = speech.types.RecognitionConfig(
-    ...     encoding=speech.enums.RecognitionConfig.AudioEncoding.LINEAR16,
-    ...     language_code='en-US',
-    ...     sample_rate_hertz=44100,
-    ...     profanity_filter=True)
-    >>> results = client.recognize(config=config, audio=audio)
-    >>> for result in results:
-    ...     for alternative in result.alternatives:
-    ...         print('=' * 20)
-    ...         print('transcript: ' + alternative.transcript)
-    ...         print('confidence: ' + str(alternative.confidence))
-    ====================
-    transcript: Hello, this is a f****** test
-    confidence: 0.81
-
-Using speech context hints to get better results. This can be used to improve
-the accuracy for specific words and phrases. This can also be used to add new
-words to the vocabulary of the recognizer.
-
-.. code-block:: python
-
-    >>> from google.cloud import speech
-    >>> from google.cloud import speech
-    >>> client = speech.SpeechClient()
-    >>> audio = speech.types.RecognitionAudio(
-    ...     uri='gs://my-bucket/recording.flac')
-    >>> config = speech.types.RecognitionConfig(
-    ...     encoding=speech.enums.RecognitionConfig.AudioEncoding.LINEAR16,
-    ...     language_code='en-US',
-    ...     sample_rate_hertz=44100,
-    ...     speech_contexts=[speech.types.SpeechContext(
-    ...         phrases=['hi', 'good afternoon'],
-    ...     )])
-    >>> results = client.recognize(config=config, audio=audio)
-    >>> for result in results:
-    ...     for alternative in result.alternatives:
-    ...         print('=' * 20)
-    ...         print('transcript: ' + alternative.transcript)
-    ...         print('confidence: ' + str(alternative.confidence))
-    ====================
-    transcript: Hello, this is a test
-    confidence: 0.81
-
-
-Streaming Recognition
-~~~~~~~~~~~~~~~~~~~~~
-
-The :meth:`~speech_v1.SpeechClient.streaming_recognize` method converts
-speech data to possible text alternatives on the fly.
-
-.. note::
-    Streaming recognition requests are limited to 1 minute of audio.
-
-    See: https://cloud.google.com/speech/limits#content
-
-.. code-block:: python
-
-    >>> import io
-    >>> from google.cloud import speech
-    >>> client = speech.SpeechClient()
-    >>> config = speech.types.RecognitionConfig(
-    ...     encoding=speech.enums.RecognitionConfig.AudioEncoding.LINEAR16,
-    ...     language_code='en-US',
-    ...     sample_rate_hertz=44100,
-    ... )
-    >>> with io.open('./hello.wav', 'rb') as stream:
-    ...     requests = [speech.types.StreamingRecognizeRequest(
-    ...         audio_content=stream.read(),
-    ...     )]
-    >>> results = sample.streaming_recognize(
-    ...     config=speech.types.StreamingRecognitionConfig(config=config),
-    ...     requests,
-    ... )
-    >>> for result in results:
-    ...     for alternative in result.alternatives:
-    ...         print('=' * 20)
-    ...         print('transcript: ' + alternative.transcript)
-    ...         print('confidence: ' + str(alternative.confidence))
-    ====================
-    transcript: hello thank you for using Google Cloud platform
-    confidence: 0.927983105183
-
-
-By default the API will perform continuous recognition
-(continuing to process audio even if the speaker in the audio pauses speaking)
-until the client closes the output stream or until the maximum time limit has
-been reached.
-
-If you only want to recognize a single utterance you can set
-``single_utterance`` to :data:`True` and only one result will be returned.
-
-See: `Single Utterance`_
-
-.. code-block:: python
-
-    >>> import io
-    >>> from google.cloud import speech
-    >>> client = speech.SpeechClient()
-    >>> config = speech.types.RecognitionConfig(
-    ...     encoding=speech.enums.RecognitionConfig.AudioEncoding.LINEAR16,
-    ...     language_code='en-US',
-    ...     sample_rate_hertz=44100,
-    ... )
-    >>> with io.open('./hello-pause-goodbye.wav', 'rb') as stream:
-    ...     requests = [speech.types.StreamingRecognizeRequest(
-    ...         audio_content=stream.read(),
-    ...     )]
-    >>> results = sample.streaming_recognize(
-    ...     config=speech.types.StreamingRecognitionConfig(
-    ...         config=config,
-    ...         single_utterance=False,
-    ...     ),
-    ...     requests,
-    ... )
-    >>> for result in results:
-    ...     for alternative in result.alternatives:
-    ...         print('=' * 20)
-    ...         print('transcript: ' + alternative.transcript)
-    ...         print('confidence: ' + str(alternative.confidence))
-    ...     for result in results:
-    ...         for alternative in result.alternatives:
-    ...             print('=' * 20)
-    ...             print('transcript: ' + alternative.transcript)
-    ...             print('confidence: ' + str(alternative.confidence))
-    ====================
-    transcript: testing a pause
-    confidence: 0.933770477772
-
-If ``interim_results`` is set to :data:`True`, interim results
-(tentative hypotheses) may be returned as they become available.
-
-.. code-block:: python
-
-    >>> import io
-    >>> from google.cloud import speech
-    >>> client = speech.SpeechClient()
-    >>> config = speech.types.RecognitionConfig(
-    ...     encoding=speech.enums.RecognitionConfig.AudioEncoding.LINEAR16,
-    ...     language_code='en-US',
-    ...     sample_rate_hertz=44100,
-    ... )
-    >>> with io.open('./hello.wav', 'rb') as stream:
-    ...     requests = [speech.types.StreamingRecognizeRequest(
-    ...         audio_content=stream.read(),
-    ...     )]
-    >>> config = speech.types.StreamingRecognitionConfig(config=config)
-    >>> responses = client.streaming_recognize(config,requests)
-    >>> for response in responses:
-    ...     for result in response:
-    ...         for alternative in result.alternatives:
-    ...             print('=' * 20)
-    ...             print('transcript: ' + alternative.transcript)
-    ...             print('confidence: ' + str(alternative.confidence))
-    ...             print('is_final:' + str(result.is_final))
-    ====================
-    'he'
-    None
-    False
-    ====================
-    'hell'
-    None
-    False
-    ====================
-    'hello'
-    0.973458576
-    True
-
-
-.. _Single Utterance: https://cloud.google.com/speech/reference/rpc/google.cloud.speech.v1beta1#streamingrecognitionconfig
-.. _sync_recognize: https://cloud.google.com/speech/reference/rest/v1beta1/speech/syncrecognize
-.. _Speech Asynchronous Recognize: https://cloud.google.com/speech/reference/rest/v1beta1/speech/asyncrecognize
-
-
-API Reference
--------------
-
-.. toctree::
-  :maxdepth: 2
-
-  gapic/v1/api
-  gapic/v1/types
-
-A new beta release, spelled ``v1p1beta1``, is provided to provide for preview
-of upcoming features. In order to use this, you will want to import from
-``google.cloud.speech_v1p1beta1`` in lieu of ``google.cloud.speech``.
-
-An API and type reference is provided the first beta also:
-
-.. toctree::
-  :maxdepth: 2
-
-  gapic/v1p1beta1/api
-  gapic/v1p1beta1/types
-
-Changelog
----------
-
-For a list of all ``google-cloud-speech`` releases:
-
-.. toctree::
-  :maxdepth: 2
-
-  changelog
diff --git a/google/__init__.py b/google/__init__.py
deleted file mode 100644
index dd3a9f48..00000000
--- a/google/__init__.py
+++ /dev/null
@@ -1,22 +0,0 @@
-# Copyright 2018 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-try:
-    import pkg_resources
-
-    pkg_resources.declare_namespace(__name__)
-except ImportError:
-    import pkgutil
-
-    __path__ = pkgutil.extend_path(__path__, __name__)
diff --git a/google/cloud/__init__.py b/google/cloud/__init__.py
deleted file mode 100644
index dd3a9f48..00000000
--- a/google/cloud/__init__.py
+++ /dev/null
@@ -1,22 +0,0 @@
-# Copyright 2018 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-try:
-    import pkg_resources
-
-    pkg_resources.declare_namespace(__name__)
-except ImportError:
-    import pkgutil
-
-    __path__ = pkgutil.extend_path(__path__, __name__)
diff --git a/google/cloud/speech.py b/google/cloud/speech.py
deleted file mode 100644
index 792e3f67..00000000
--- a/google/cloud/speech.py
+++ /dev/null
@@ -1,25 +0,0 @@
-# -*- coding: utf-8 -*-
-#
-# Copyright 2020 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-
-from __future__ import absolute_import
-
-from google.cloud.speech_v1 import SpeechClient
-from google.cloud.speech_v1 import enums
-from google.cloud.speech_v1 import types
-
-
-__all__ = ("enums", "types", "SpeechClient")
diff --git a/google/cloud/speech_v1/__init__.py b/google/cloud/speech_v1/__init__.py
deleted file mode 100644
index 81314685..00000000
--- a/google/cloud/speech_v1/__init__.py
+++ /dev/null
@@ -1,30 +0,0 @@
-# Copyright 2017 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     http://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-from __future__ import absolute_import
-
-from google.cloud.speech_v1.gapic import speech_client
-from google.cloud.speech_v1.gapic import enums
-
-from google.cloud.speech_v1.helpers import SpeechHelpers
-from google.cloud.speech_v1 import types
-
-
-class SpeechClient(SpeechHelpers, speech_client.SpeechClient):
-    __doc__ = speech_client.SpeechClient.__doc__
-    enums = enums
-    types = types
-
-
-__all__ = ("enums", "SpeechClient", "types")
diff --git a/google/cloud/speech_v1/gapic/__init__.py b/google/cloud/speech_v1/gapic/__init__.py
deleted file mode 100644
index e69de29b..00000000
diff --git a/google/cloud/speech_v1/gapic/enums.py b/google/cloud/speech_v1/gapic/enums.py
deleted file mode 100644
index fa6218ab..00000000
--- a/google/cloud/speech_v1/gapic/enums.py
+++ /dev/null
@@ -1,196 +0,0 @@
-# -*- coding: utf-8 -*-
-#
-# Copyright 2020 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-"""Wrappers for protocol buffer enum types."""
-
-import enum
-
-
-class RecognitionConfig(object):
-    class AudioEncoding(enum.IntEnum):
-        """
-        The encoding of the audio data sent in the request.
-
-        All encodings support only 1 channel (mono) audio, unless the
-        ``audio_channel_count`` and ``enable_separate_recognition_per_channel``
-        fields are set.
-
-        For best results, the audio source should be captured and transmitted
-        using a lossless encoding (``FLAC`` or ``LINEAR16``). The accuracy of
-        the speech recognition can be reduced if lossy codecs are used to
-        capture or transmit audio, particularly if background noise is present.
-        Lossy codecs include ``MULAW``, ``AMR``, ``AMR_WB``, ``OGG_OPUS``,
-        ``SPEEX_WITH_HEADER_BYTE``, and ``MP3``.
-
-        The ``FLAC`` and ``WAV`` audio file formats include a header that
-        describes the included audio content. You can request recognition for
-        ``WAV`` files that contain either ``LINEAR16`` or ``MULAW`` encoded
-        audio. If you send ``FLAC`` or ``WAV`` audio file format in your
-        request, you do not need to specify an ``AudioEncoding``; the audio
-        encoding format is determined from the file header. If you specify an
-        ``AudioEncoding`` when you send send ``FLAC`` or ``WAV`` audio, the
-        encoding configuration must match the encoding described in the audio
-        header; otherwise the request returns an
-        ``google.rpc.Code.INVALID_ARGUMENT`` error code.
-
-        Attributes:
-          ENCODING_UNSPECIFIED (int): Not specified.
-          LINEAR16 (int): Uncompressed 16-bit signed little-endian samples (Linear PCM).
-          FLAC (int): ``FLAC`` (Free Lossless Audio Codec) is the recommended encoding because
-          it is lossless--therefore recognition is not compromised--and requires
-          only about half the bandwidth of ``LINEAR16``. ``FLAC`` stream encoding
-          supports 16-bit and 24-bit samples, however, not all fields in
-          ``STREAMINFO`` are supported.
-          MULAW (int): 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
-          AMR (int): Adaptive Multi-Rate Narrowband codec. ``sample_rate_hertz`` must be
-          8000.
-          AMR_WB (int): Adaptive Multi-Rate Wideband codec. ``sample_rate_hertz`` must be 16000.
-          OGG_OPUS (int): Opus encoded audio frames in Ogg container
-          (`OggOpus <https://wiki.xiph.org/OggOpus>`__). ``sample_rate_hertz``
-          must be one of 8000, 12000, 16000, 24000, or 48000.
-          SPEEX_WITH_HEADER_BYTE (int): Although the use of lossy encodings is not recommended, if a very low
-          bitrate encoding is required, ``OGG_OPUS`` is highly preferred over
-          Speex encoding. The `Speex <https://speex.org/>`__ encoding supported by
-          Cloud Speech API has a header byte in each block, as in MIME type
-          ``audio/x-speex-with-header-byte``. It is a variant of the RTP Speex
-          encoding defined in `RFC 5574 <https://tools.ietf.org/html/rfc5574>`__.
-          The stream is a sequence of blocks, one block per RTP packet. Each block
-          starts with a byte containing the length of the block, in bytes,
-          followed by one or more frames of Speex data, padded to an integral
-          number of bytes (octets) as specified in RFC 5574. In other words, each
-          RTP header is replaced with a single byte containing the block length.
-          Only Speex wideband is supported. ``sample_rate_hertz`` must be 16000.
-        """
-
-        ENCODING_UNSPECIFIED = 0
-        LINEAR16 = 1
-        FLAC = 2
-        MULAW = 3
-        AMR = 4
-        AMR_WB = 5
-        OGG_OPUS = 6
-        SPEEX_WITH_HEADER_BYTE = 7
-
-
-class RecognitionMetadata(object):
-    class InteractionType(enum.IntEnum):
-        """
-        Use case categories that the audio recognition request can be described
-        by.
-
-        Attributes:
-          INTERACTION_TYPE_UNSPECIFIED (int): Use case is either unknown or is something other than one of the other
-          values below.
-          DISCUSSION (int): Multiple people in a conversation or discussion. For example in a
-          meeting with two or more people actively participating. Typically all
-          the primary people speaking would be in the same room (if not, see
-          PHONE\_CALL)
-          PRESENTATION (int): One or more persons lecturing or presenting to others, mostly
-          uninterrupted.
-          PHONE_CALL (int): A phone-call or video-conference in which two or more people, who are
-          not in the same room, are actively participating.
-          VOICEMAIL (int): A recorded message intended for another person to listen to.
-          PROFESSIONALLY_PRODUCED (int): Professionally produced audio (eg. TV Show, Podcast).
-          VOICE_SEARCH (int): Transcribe spoken questions and queries into text.
-          VOICE_COMMAND (int): Transcribe voice commands, such as for controlling a device.
-          DICTATION (int): Transcribe speech to text to create a written document, such as a
-          text-message, email or report.
-        """
-
-        INTERACTION_TYPE_UNSPECIFIED = 0
-        DISCUSSION = 1
-        PRESENTATION = 2
-        PHONE_CALL = 3
-        VOICEMAIL = 4
-        PROFESSIONALLY_PRODUCED = 5
-        VOICE_SEARCH = 6
-        VOICE_COMMAND = 7
-        DICTATION = 8
-
-    class MicrophoneDistance(enum.IntEnum):
-        """
-        Enumerates the types of capture settings describing an audio file.
-
-        Attributes:
-          MICROPHONE_DISTANCE_UNSPECIFIED (int): Audio type is not known.
-          NEARFIELD (int): The audio was captured from a closely placed microphone. Eg. phone,
-          dictaphone, or handheld microphone. Generally if there speaker is within
-          1 meter of the microphone.
-          MIDFIELD (int): The speaker if within 3 meters of the microphone.
-          FARFIELD (int): The speaker is more than 3 meters away from the microphone.
-        """
-
-        MICROPHONE_DISTANCE_UNSPECIFIED = 0
-        NEARFIELD = 1
-        MIDFIELD = 2
-        FARFIELD = 3
-
-    class OriginalMediaType(enum.IntEnum):
-        """
-        The original media the speech was recorded on.
-
-        Attributes:
-          ORIGINAL_MEDIA_TYPE_UNSPECIFIED (int): Unknown original media type.
-          AUDIO (int): The speech data is an audio recording.
-          VIDEO (int): The speech data originally recorded on a video.
-        """
-
-        ORIGINAL_MEDIA_TYPE_UNSPECIFIED = 0
-        AUDIO = 1
-        VIDEO = 2
-
-    class RecordingDeviceType(enum.IntEnum):
-        """
-        The type of device the speech was recorded with.
-
-        Attributes:
-          RECORDING_DEVICE_TYPE_UNSPECIFIED (int): The recording device is unknown.
-          SMARTPHONE (int): Speech was recorded on a smartphone.
-          PC (int): Speech was recorded using a personal computer or tablet.
-          PHONE_LINE (int): Speech was recorded over a phone line.
-          VEHICLE (int): Speech was recorded in a vehicle.
-          OTHER_OUTDOOR_DEVICE (int): Speech was recorded outdoors.
-          OTHER_INDOOR_DEVICE (int): Speech was recorded indoors.
-        """
-
-        RECORDING_DEVICE_TYPE_UNSPECIFIED = 0
-        SMARTPHONE = 1
-        PC = 2
-        PHONE_LINE = 3
-        VEHICLE = 4
-        OTHER_OUTDOOR_DEVICE = 5
-        OTHER_INDOOR_DEVICE = 6
-
-
-class StreamingRecognizeResponse(object):
-    class SpeechEventType(enum.IntEnum):
-        """
-        Indicates the type of speech event.
-
-        Attributes:
-          SPEECH_EVENT_UNSPECIFIED (int): No speech event specified.
-          END_OF_SINGLE_UTTERANCE (int): This event indicates that the server has detected the end of the user's
-          speech utterance and expects no additional speech. Therefore, the server
-          will not process additional audio (although it may subsequently return
-          additional results). The client should stop sending additional audio
-          data, half-close the gRPC connection, and wait for any additional
-          results until the server closes the gRPC connection. This event is only
-          sent if ``single_utterance`` was set to ``true``, and is not used
-          otherwise.
-        """
-
-        SPEECH_EVENT_UNSPECIFIED = 0
-        END_OF_SINGLE_UTTERANCE = 1
diff --git a/google/cloud/speech_v1/gapic/speech_client.py b/google/cloud/speech_v1/gapic/speech_client.py
deleted file mode 100644
index 83cb7f44..00000000
--- a/google/cloud/speech_v1/gapic/speech_client.py
+++ /dev/null
@@ -1,412 +0,0 @@
-# -*- coding: utf-8 -*-
-#
-# Copyright 2020 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-"""Accesses the google.cloud.speech.v1 Speech API."""
-
-import pkg_resources
-import warnings
-
-from google.oauth2 import service_account
-import google.api_core.client_options
-import google.api_core.gapic_v1.client_info
-import google.api_core.gapic_v1.config
-import google.api_core.gapic_v1.method
-import google.api_core.grpc_helpers
-import google.api_core.operation
-import google.api_core.operations_v1
-import google.api_core.protobuf_helpers
-import grpc
-
-from google.cloud.speech_v1.gapic import enums
-from google.cloud.speech_v1.gapic import speech_client_config
-from google.cloud.speech_v1.gapic.transports import speech_grpc_transport
-from google.cloud.speech_v1.proto import cloud_speech_pb2
-from google.cloud.speech_v1.proto import cloud_speech_pb2_grpc
-from google.longrunning import operations_pb2
-
-
-_GAPIC_LIBRARY_VERSION = pkg_resources.get_distribution("google-cloud-speech").version
-
-
-class SpeechClient(object):
-    """Service that implements Google Cloud Speech API."""
-
-    SERVICE_ADDRESS = "speech.googleapis.com:443"
-    """The default address of the service."""
-
-    # The name of the interface for this client. This is the key used to
-    # find the method configuration in the client_config dictionary.
-    _INTERFACE_NAME = "google.cloud.speech.v1.Speech"
-
-    @classmethod
-    def from_service_account_file(cls, filename, *args, **kwargs):
-        """Creates an instance of this client using the provided credentials
-        file.
-
-        Args:
-            filename (str): The path to the service account private key json
-                file.
-            args: Additional arguments to pass to the constructor.
-            kwargs: Additional arguments to pass to the constructor.
-
-        Returns:
-            SpeechClient: The constructed client.
-        """
-        credentials = service_account.Credentials.from_service_account_file(filename)
-        kwargs["credentials"] = credentials
-        return cls(*args, **kwargs)
-
-    from_service_account_json = from_service_account_file
-
-    def __init__(
-        self,
-        transport=None,
-        channel=None,
-        credentials=None,
-        client_config=None,
-        client_info=None,
-        client_options=None,
-    ):
-        """Constructor.
-
-        Args:
-            transport (Union[~.SpeechGrpcTransport,
-                    Callable[[~.Credentials, type], ~.SpeechGrpcTransport]): A transport
-                instance, responsible for actually making the API calls.
-                The default transport uses the gRPC protocol.
-                This argument may also be a callable which returns a
-                transport instance. Callables will be sent the credentials
-                as the first argument and the default transport class as
-                the second argument.
-            channel (grpc.Channel): DEPRECATED. A ``Channel`` instance
-                through which to make calls. This argument is mutually exclusive
-                with ``credentials``; providing both will raise an exception.
-            credentials (google.auth.credentials.Credentials): The
-                authorization credentials to attach to requests. These
-                credentials identify this application to the service. If none
-                are specified, the client will attempt to ascertain the
-                credentials from the environment.
-                This argument is mutually exclusive with providing a
-                transport instance to ``transport``; doing so will raise
-                an exception.
-            client_config (dict): DEPRECATED. A dictionary of call options for
-                each method. If not specified, the default configuration is used.
-            client_info (google.api_core.gapic_v1.client_info.ClientInfo):
-                The client info used to send a user-agent string along with
-                API requests. If ``None``, then default info will be used.
-                Generally, you only need to set this if you're developing
-                your own client library.
-            client_options (Union[dict, google.api_core.client_options.ClientOptions]):
-                Client options used to set user options on the client. API Endpoint
-                should be set through client_options.
-        """
-        # Raise deprecation warnings for things we want to go away.
-        if client_config is not None:
-            warnings.warn(
-                "The `client_config` argument is deprecated.",
-                PendingDeprecationWarning,
-                stacklevel=2,
-            )
-        else:
-            client_config = speech_client_config.config
-
-        if channel:
-            warnings.warn(
-                "The `channel` argument is deprecated; use " "`transport` instead.",
-                PendingDeprecationWarning,
-                stacklevel=2,
-            )
-
-        api_endpoint = self.SERVICE_ADDRESS
-        if client_options:
-            if type(client_options) == dict:
-                client_options = google.api_core.client_options.from_dict(
-                    client_options
-                )
-            if client_options.api_endpoint:
-                api_endpoint = client_options.api_endpoint
-
-        # Instantiate the transport.
-        # The transport is responsible for handling serialization and
-        # deserialization and actually sending data to the service.
-        if transport:
-            if callable(transport):
-                self.transport = transport(
-                    credentials=credentials,
-                    default_class=speech_grpc_transport.SpeechGrpcTransport,
-                    address=api_endpoint,
-                )
-            else:
-                if credentials:
-                    raise ValueError(
-                        "Received both a transport instance and "
-                        "credentials; these are mutually exclusive."
-                    )
-                self.transport = transport
-        else:
-            self.transport = speech_grpc_transport.SpeechGrpcTransport(
-                address=api_endpoint, channel=channel, credentials=credentials
-            )
-
-        if client_info is None:
-            client_info = google.api_core.gapic_v1.client_info.ClientInfo(
-                gapic_version=_GAPIC_LIBRARY_VERSION
-            )
-        else:
-            client_info.gapic_version = _GAPIC_LIBRARY_VERSION
-        self._client_info = client_info
-
-        # Parse out the default settings for retry and timeout for each RPC
-        # from the client configuration.
-        # (Ordinarily, these are the defaults specified in the `*_config.py`
-        # file next to this one.)
-        self._method_configs = google.api_core.gapic_v1.config.parse_method_configs(
-            client_config["interfaces"][self._INTERFACE_NAME]
-        )
-
-        # Save a dictionary of cached API call functions.
-        # These are the actual callables which invoke the proper
-        # transport methods, wrapped with `wrap_method` to add retry,
-        # timeout, and the like.
-        self._inner_api_calls = {}
-
-    # Service calls
-    def recognize(
-        self,
-        config,
-        audio,
-        retry=google.api_core.gapic_v1.method.DEFAULT,
-        timeout=google.api_core.gapic_v1.method.DEFAULT,
-        metadata=None,
-    ):
-        """
-        Performs synchronous speech recognition: receive results after all audio
-        has been sent and processed.
-
-        Example:
-            >>> from google.cloud import speech_v1
-            >>> from google.cloud.speech_v1 import enums
-            >>>
-            >>> client = speech_v1.SpeechClient()
-            >>>
-            >>> encoding = enums.RecognitionConfig.AudioEncoding.FLAC
-            >>> sample_rate_hertz = 44100
-            >>> language_code = 'en-US'
-            >>> config = {'encoding': encoding, 'sample_rate_hertz': sample_rate_hertz, 'language_code': language_code}
-            >>> uri = 'gs://bucket_name/file_name.flac'
-            >>> audio = {'uri': uri}
-            >>>
-            >>> response = client.recognize(config, audio)
-
-        Args:
-            config (Union[dict, ~google.cloud.speech_v1.types.RecognitionConfig]): Required. Provides information to the recognizer that specifies how to
-                process the request.
-
-                If a dict is provided, it must be of the same form as the protobuf
-                message :class:`~google.cloud.speech_v1.types.RecognitionConfig`
-            audio (Union[dict, ~google.cloud.speech_v1.types.RecognitionAudio]): Required. The audio data to be recognized.
-
-                If a dict is provided, it must be of the same form as the protobuf
-                message :class:`~google.cloud.speech_v1.types.RecognitionAudio`
-            retry (Optional[google.api_core.retry.Retry]):  A retry object used
-                to retry requests. If ``None`` is specified, requests will
-                be retried using a default configuration.
-            timeout (Optional[float]): The amount of time, in seconds, to wait
-                for the request to complete. Note that if ``retry`` is
-                specified, the timeout applies to each individual attempt.
-            metadata (Optional[Sequence[Tuple[str, str]]]): Additional metadata
-                that is provided to the method.
-
-        Returns:
-            A :class:`~google.cloud.speech_v1.types.RecognizeResponse` instance.
-
-        Raises:
-            google.api_core.exceptions.GoogleAPICallError: If the request
-                    failed for any reason.
-            google.api_core.exceptions.RetryError: If the request failed due
-                    to a retryable error and retry attempts failed.
-            ValueError: If the parameters are invalid.
-        """
-        # Wrap the transport method to add retry and timeout logic.
-        if "recognize" not in self._inner_api_calls:
-            self._inner_api_calls[
-                "recognize"
-            ] = google.api_core.gapic_v1.method.wrap_method(
-                self.transport.recognize,
-                default_retry=self._method_configs["Recognize"].retry,
-                default_timeout=self._method_configs["Recognize"].timeout,
-                client_info=self._client_info,
-            )
-
-        request = cloud_speech_pb2.RecognizeRequest(config=config, audio=audio)
-        return self._inner_api_calls["recognize"](
-            request, retry=retry, timeout=timeout, metadata=metadata
-        )
-
-    def long_running_recognize(
-        self,
-        config,
-        audio,
-        retry=google.api_core.gapic_v1.method.DEFAULT,
-        timeout=google.api_core.gapic_v1.method.DEFAULT,
-        metadata=None,
-    ):
-        """
-        Performs asynchronous speech recognition: receive results via the
-        google.longrunning.Operations interface. Returns either an
-        ``Operation.error`` or an ``Operation.response`` which contains a
-        ``LongRunningRecognizeResponse`` message. For more information on
-        asynchronous speech recognition, see the
-        `how-to <https://cloud.google.com/speech-to-text/docs/async-recognize>`__.
-
-        Example:
-            >>> from google.cloud import speech_v1
-            >>> from google.cloud.speech_v1 import enums
-            >>>
-            >>> client = speech_v1.SpeechClient()
-            >>>
-            >>> encoding = enums.RecognitionConfig.AudioEncoding.FLAC
-            >>> sample_rate_hertz = 44100
-            >>> language_code = 'en-US'
-            >>> config = {'encoding': encoding, 'sample_rate_hertz': sample_rate_hertz, 'language_code': language_code}
-            >>> uri = 'gs://bucket_name/file_name.flac'
-            >>> audio = {'uri': uri}
-            >>>
-            >>> response = client.long_running_recognize(config, audio)
-            >>>
-            >>> def callback(operation_future):
-            ...     # Handle result.
-            ...     result = operation_future.result()
-            >>>
-            >>> response.add_done_callback(callback)
-            >>>
-            >>> # Handle metadata.
-            >>> metadata = response.metadata()
-
-        Args:
-            config (Union[dict, ~google.cloud.speech_v1.types.RecognitionConfig]): Required. Provides information to the recognizer that specifies how to
-                process the request.
-
-                If a dict is provided, it must be of the same form as the protobuf
-                message :class:`~google.cloud.speech_v1.types.RecognitionConfig`
-            audio (Union[dict, ~google.cloud.speech_v1.types.RecognitionAudio]): Required. The audio data to be recognized.
-
-                If a dict is provided, it must be of the same form as the protobuf
-                message :class:`~google.cloud.speech_v1.types.RecognitionAudio`
-            retry (Optional[google.api_core.retry.Retry]):  A retry object used
-                to retry requests. If ``None`` is specified, requests will
-                be retried using a default configuration.
-            timeout (Optional[float]): The amount of time, in seconds, to wait
-                for the request to complete. Note that if ``retry`` is
-                specified, the timeout applies to each individual attempt.
-            metadata (Optional[Sequence[Tuple[str, str]]]): Additional metadata
-                that is provided to the method.
-
-        Returns:
-            A :class:`~google.cloud.speech_v1.types._OperationFuture` instance.
-
-        Raises:
-            google.api_core.exceptions.GoogleAPICallError: If the request
-                    failed for any reason.
-            google.api_core.exceptions.RetryError: If the request failed due
-                    to a retryable error and retry attempts failed.
-            ValueError: If the parameters are invalid.
-        """
-        # Wrap the transport method to add retry and timeout logic.
-        if "long_running_recognize" not in self._inner_api_calls:
-            self._inner_api_calls[
-                "long_running_recognize"
-            ] = google.api_core.gapic_v1.method.wrap_method(
-                self.transport.long_running_recognize,
-                default_retry=self._method_configs["LongRunningRecognize"].retry,
-                default_timeout=self._method_configs["LongRunningRecognize"].timeout,
-                client_info=self._client_info,
-            )
-
-        request = cloud_speech_pb2.LongRunningRecognizeRequest(
-            config=config, audio=audio
-        )
-        operation = self._inner_api_calls["long_running_recognize"](
-            request, retry=retry, timeout=timeout, metadata=metadata
-        )
-        return google.api_core.operation.from_gapic(
-            operation,
-            self.transport._operations_client,
-            cloud_speech_pb2.LongRunningRecognizeResponse,
-            metadata_type=cloud_speech_pb2.LongRunningRecognizeMetadata,
-        )
-
-    def streaming_recognize(
-        self,
-        requests,
-        retry=google.api_core.gapic_v1.method.DEFAULT,
-        timeout=google.api_core.gapic_v1.method.DEFAULT,
-        metadata=None,
-    ):
-        """
-        Performs bidirectional streaming speech recognition: receive results while
-        sending audio. This method is only available via the gRPC API (not REST).
-
-        EXPERIMENTAL: This method interface might change in the future.
-
-        Example:
-            >>> from google.cloud import speech_v1
-            >>>
-            >>> client = speech_v1.SpeechClient()
-            >>>
-            >>> request = {}
-            >>>
-            >>> requests = [request]
-            >>> for element in client.streaming_recognize(requests):
-            ...     # process element
-            ...     pass
-
-        Args:
-            requests (iterator[dict|google.cloud.speech_v1.proto.cloud_speech_pb2.StreamingRecognizeRequest]): The input objects. If a dict is provided, it must be of the
-                same form as the protobuf message :class:`~google.cloud.speech_v1.types.StreamingRecognizeRequest`
-            retry (Optional[google.api_core.retry.Retry]):  A retry object used
-                to retry requests. If ``None`` is specified, requests will
-                be retried using a default configuration.
-            timeout (Optional[float]): The amount of time, in seconds, to wait
-                for the request to complete. Note that if ``retry`` is
-                specified, the timeout applies to each individual attempt.
-            metadata (Optional[Sequence[Tuple[str, str]]]): Additional metadata
-                that is provided to the method.
-
-        Returns:
-            Iterable[~google.cloud.speech_v1.types.StreamingRecognizeResponse].
-
-        Raises:
-            google.api_core.exceptions.GoogleAPICallError: If the request
-                    failed for any reason.
-            google.api_core.exceptions.RetryError: If the request failed due
-                    to a retryable error and retry attempts failed.
-            ValueError: If the parameters are invalid.
-        """
-        # Wrap the transport method to add retry and timeout logic.
-        if "streaming_recognize" not in self._inner_api_calls:
-            self._inner_api_calls[
-                "streaming_recognize"
-            ] = google.api_core.gapic_v1.method.wrap_method(
-                self.transport.streaming_recognize,
-                default_retry=self._method_configs["StreamingRecognize"].retry,
-                default_timeout=self._method_configs["StreamingRecognize"].timeout,
-                client_info=self._client_info,
-            )
-
-        return self._inner_api_calls["streaming_recognize"](
-            requests, retry=retry, timeout=timeout, metadata=metadata
-        )
diff --git a/google/cloud/speech_v1/gapic/speech_client_config.py b/google/cloud/speech_v1/gapic/speech_client_config.py
deleted file mode 100644
index 555d3f2a..00000000
--- a/google/cloud/speech_v1/gapic/speech_client_config.py
+++ /dev/null
@@ -1,38 +0,0 @@
-config = {
-    "interfaces": {
-        "google.cloud.speech.v1.Speech": {
-            "retry_codes": {
-                "idempotent": ["DEADLINE_EXCEEDED", "UNAVAILABLE"],
-                "non_idempotent": [],
-            },
-            "retry_params": {
-                "default": {
-                    "initial_retry_delay_millis": 100,
-                    "retry_delay_multiplier": 1.3,
-                    "max_retry_delay_millis": 60000,
-                    "initial_rpc_timeout_millis": 1000000,
-                    "rpc_timeout_multiplier": 1.0,
-                    "max_rpc_timeout_millis": 1000000,
-                    "total_timeout_millis": 5000000,
-                }
-            },
-            "methods": {
-                "Recognize": {
-                    "timeout_millis": 60000,
-                    "retry_codes_name": "idempotent",
-                    "retry_params_name": "default",
-                },
-                "LongRunningRecognize": {
-                    "timeout_millis": 60000,
-                    "retry_codes_name": "non_idempotent",
-                    "retry_params_name": "default",
-                },
-                "StreamingRecognize": {
-                    "timeout_millis": 60000,
-                    "retry_codes_name": "non_idempotent",
-                    "retry_params_name": "default",
-                },
-            },
-        }
-    }
-}
diff --git a/google/cloud/speech_v1/gapic/transports/__init__.py b/google/cloud/speech_v1/gapic/transports/__init__.py
deleted file mode 100644
index e69de29b..00000000
diff --git a/google/cloud/speech_v1/gapic/transports/speech_grpc_transport.py b/google/cloud/speech_v1/gapic/transports/speech_grpc_transport.py
deleted file mode 100644
index c0d9c39d..00000000
--- a/google/cloud/speech_v1/gapic/transports/speech_grpc_transport.py
+++ /dev/null
@@ -1,160 +0,0 @@
-# -*- coding: utf-8 -*-
-#
-# Copyright 2020 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-
-import google.api_core.grpc_helpers
-import google.api_core.operations_v1
-
-from google.cloud.speech_v1.proto import cloud_speech_pb2_grpc
-
-
-class SpeechGrpcTransport(object):
-    """gRPC transport class providing stubs for
-    google.cloud.speech.v1 Speech API.
-
-    The transport provides access to the raw gRPC stubs,
-    which can be used to take advantage of advanced
-    features of gRPC.
-    """
-
-    # The scopes needed to make gRPC calls to all of the methods defined
-    # in this service.
-    _OAUTH_SCOPES = ("https://www.googleapis.com/auth/cloud-platform",)
-
-    def __init__(
-        self, channel=None, credentials=None, address="speech.googleapis.com:443"
-    ):
-        """Instantiate the transport class.
-
-        Args:
-            channel (grpc.Channel): A ``Channel`` instance through
-                which to make calls. This argument is mutually exclusive
-                with ``credentials``; providing both will raise an exception.
-            credentials (google.auth.credentials.Credentials): The
-                authorization credentials to attach to requests. These
-                credentials identify this application to the service. If none
-                are specified, the client will attempt to ascertain the
-                credentials from the environment.
-            address (str): The address where the service is hosted.
-        """
-        # If both `channel` and `credentials` are specified, raise an
-        # exception (channels come with credentials baked in already).
-        if channel is not None and credentials is not None:
-            raise ValueError(
-                "The `channel` and `credentials` arguments are mutually " "exclusive."
-            )
-
-        # Create the channel.
-        if channel is None:
-            channel = self.create_channel(
-                address=address,
-                credentials=credentials,
-                options={
-                    "grpc.max_send_message_length": -1,
-                    "grpc.max_receive_message_length": -1,
-                }.items(),
-            )
-
-        self._channel = channel
-
-        # gRPC uses objects called "stubs" that are bound to the
-        # channel and provide a basic method for each RPC.
-        self._stubs = {"speech_stub": cloud_speech_pb2_grpc.SpeechStub(channel)}
-
-        # Because this API includes a method that returns a
-        # long-running operation (proto: google.longrunning.Operation),
-        # instantiate an LRO client.
-        self._operations_client = google.api_core.operations_v1.OperationsClient(
-            channel
-        )
-
-    @classmethod
-    def create_channel(
-        cls, address="speech.googleapis.com:443", credentials=None, **kwargs
-    ):
-        """Create and return a gRPC channel object.
-
-        Args:
-            address (str): The host for the channel to use.
-            credentials (~.Credentials): The
-                authorization credentials to attach to requests. These
-                credentials identify this application to the service. If
-                none are specified, the client will attempt to ascertain
-                the credentials from the environment.
-            kwargs (dict): Keyword arguments, which are passed to the
-                channel creation.
-
-        Returns:
-            grpc.Channel: A gRPC channel object.
-        """
-        return google.api_core.grpc_helpers.create_channel(
-            address, credentials=credentials, scopes=cls._OAUTH_SCOPES, **kwargs
-        )
-
-    @property
-    def channel(self):
-        """The gRPC channel used by the transport.
-
-        Returns:
-            grpc.Channel: A gRPC channel object.
-        """
-        return self._channel
-
-    @property
-    def recognize(self):
-        """Return the gRPC stub for :meth:`SpeechClient.recognize`.
-
-        Performs synchronous speech recognition: receive results after all audio
-        has been sent and processed.
-
-        Returns:
-            Callable: A callable which accepts the appropriate
-                deserialized request object and returns a
-                deserialized response object.
-        """
-        return self._stubs["speech_stub"].Recognize
-
-    @property
-    def long_running_recognize(self):
-        """Return the gRPC stub for :meth:`SpeechClient.long_running_recognize`.
-
-        Performs asynchronous speech recognition: receive results via the
-        google.longrunning.Operations interface. Returns either an
-        ``Operation.error`` or an ``Operation.response`` which contains a
-        ``LongRunningRecognizeResponse`` message. For more information on
-        asynchronous speech recognition, see the
-        `how-to <https://cloud.google.com/speech-to-text/docs/async-recognize>`__.
-
-        Returns:
-            Callable: A callable which accepts the appropriate
-                deserialized request object and returns a
-                deserialized response object.
-        """
-        return self._stubs["speech_stub"].LongRunningRecognize
-
-    @property
-    def streaming_recognize(self):
-        """Return the gRPC stub for :meth:`SpeechClient.streaming_recognize`.
-
-        Performs bidirectional streaming speech recognition: receive results while
-        sending audio. This method is only available via the gRPC API (not REST).
-
-        Returns:
-            Callable: A callable which accepts the appropriate
-                deserialized request object and returns a
-                deserialized response object.
-        """
-        return self._stubs["speech_stub"].StreamingRecognize
diff --git a/google/cloud/speech_v1/helpers.py b/google/cloud/speech_v1/helpers.py
deleted file mode 100644
index 26f17dbe..00000000
--- a/google/cloud/speech_v1/helpers.py
+++ /dev/null
@@ -1,102 +0,0 @@
-# Copyright 2017, Google LLC All rights reserved.
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     http://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-from __future__ import absolute_import
-
-import google.api_core.gapic_v1.method
-
-
-class SpeechHelpers(object):
-    """A set of convenience methods to make the Speech client easier to use.
-
-    This class should be considered abstract; it is used as a superclass
-    in a multiple-inheritance construction alongside the applicable GAPIC.
-    See the :class:`~google.cloud.speech_v1.SpeechClient`.
-    """
-
-    def streaming_recognize(
-        self,
-        config,
-        requests,
-        retry=google.api_core.gapic_v1.method.DEFAULT,
-        timeout=google.api_core.gapic_v1.method.DEFAULT,
-    ):
-        """Perform bi-directional speech recognition.
-
-        This method allows you to receive results while sending audio;
-        it is only available via. gRPC (not REST).
-
-        .. warning::
-
-            This method is EXPERIMENTAL. Its interface might change in the
-            future.
-
-        Example:
-          >>> from google.cloud.speech_v1 import enums
-          >>> from google.cloud.speech_v1 import SpeechClient
-          >>> from google.cloud.speech_v1 import types
-          >>> client = SpeechClient()
-          >>> config = types.StreamingRecognitionConfig(
-          ...     config=types.RecognitionConfig(
-          ...         encoding=enums.RecognitionConfig.AudioEncoding.FLAC,
-          ...     ),
-          ... )
-          >>> request = types.StreamingRecognizeRequest(audio_content=b'...')
-          >>> requests = [request]
-          >>> for element in client.streaming_recognize(config, requests):
-          ...     # process element
-          ...     pass
-
-        Args:
-            config (:class:`~.types.StreamingRecognitionConfig`): The
-                configuration to use for the stream.
-            requests (Iterable[:class:`~.types.StreamingRecognizeRequest`]):
-                The input objects.
-            retry (Optional[google.api_core.retry.Retry]):  A retry object used
-                to retry requests. If ``None`` is specified, requests will not
-                be retried.
-            timeout (Optional[float]): The amount of time, in seconds, to wait
-                for the request to complete. Note that if ``retry`` is
-                specified, the timeout applies to each individual attempt.
-
-        Returns:
-          Iterable[:class:`~.types.StreamingRecognizeResponse`]
-
-        Raises:
-          :exc:`google.gax.errors.GaxError` if the RPC is aborted.
-          :exc:`ValueError` if the parameters are invalid.
-        """
-        return super(SpeechHelpers, self).streaming_recognize(
-            self._streaming_request_iterable(config, requests),
-            retry=retry,
-            timeout=timeout,
-        )
-
-    def _streaming_request_iterable(self, config, requests):
-        """A generator that yields the config followed by the requests.
-
-        Args:
-            config (~.speech_v1.types.StreamingRecognitionConfig): The
-                configuration to use for the stream.
-            requests (Iterable[~.speech_v1.types.StreamingRecognizeRequest]):
-                The input objects.
-
-        Returns:
-            Iterable[~.speech_v1.types.StreamingRecognizeRequest]): The
-                correctly formatted input for
-                :meth:`~.speech_v1.SpeechClient.streaming_recognize`.
-        """
-        yield self.types.StreamingRecognizeRequest(streaming_config=config)
-        for request in requests:
-            yield request
diff --git a/google/cloud/speech_v1/proto/__init__.py b/google/cloud/speech_v1/proto/__init__.py
deleted file mode 100644
index e69de29b..00000000
diff --git a/google/cloud/speech_v1/proto/cloud_speech.proto b/google/cloud/speech_v1/proto/cloud_speech.proto
deleted file mode 100644
index f343fa21..00000000
--- a/google/cloud/speech_v1/proto/cloud_speech.proto
+++ /dev/null
@@ -1,759 +0,0 @@
-// Copyright 2019 Google LLC.
-//
-// Licensed under the Apache License, Version 2.0 (the "License");
-// you may not use this file except in compliance with the License.
-// You may obtain a copy of the License at
-//
-//     http://www.apache.org/licenses/LICENSE-2.0
-//
-// Unless required by applicable law or agreed to in writing, software
-// distributed under the License is distributed on an "AS IS" BASIS,
-// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-// See the License for the specific language governing permissions and
-// limitations under the License.
-//
-
-syntax = "proto3";
-
-package google.cloud.speech.v1;
-
-import "google/api/annotations.proto";
-import "google/api/client.proto";
-import "google/api/field_behavior.proto";
-import "google/longrunning/operations.proto";
-import "google/protobuf/any.proto";
-import "google/protobuf/duration.proto";
-import "google/protobuf/timestamp.proto";
-import "google/rpc/status.proto";
-
-option cc_enable_arenas = true;
-option go_package = "google.golang.org/genproto/googleapis/cloud/speech/v1;speech";
-option java_multiple_files = true;
-option java_outer_classname = "SpeechProto";
-option java_package = "com.google.cloud.speech.v1";
-option objc_class_prefix = "GCS";
-
-// Service that implements Google Cloud Speech API.
-service Speech {
-  option (google.api.default_host) = "speech.googleapis.com";
-  option (google.api.oauth_scopes) = "https://www.googleapis.com/auth/cloud-platform";
-
-  // Performs synchronous speech recognition: receive results after all audio
-  // has been sent and processed.
-  rpc Recognize(RecognizeRequest) returns (RecognizeResponse) {
-    option (google.api.http) = {
-      post: "/v1/speech:recognize"
-      body: "*"
-    };
-    option (google.api.method_signature) = "config,audio";
-  }
-
-  // Performs asynchronous speech recognition: receive results via the
-  // google.longrunning.Operations interface. Returns either an
-  // `Operation.error` or an `Operation.response` which contains
-  // a `LongRunningRecognizeResponse` message.
-  // For more information on asynchronous speech recognition, see the
-  // [how-to](https://cloud.google.com/speech-to-text/docs/async-recognize).
-  rpc LongRunningRecognize(LongRunningRecognizeRequest) returns (google.longrunning.Operation) {
-    option (google.api.http) = {
-      post: "/v1/speech:longrunningrecognize"
-      body: "*"
-    };
-    option (google.api.method_signature) = "config,audio";
-    option (google.longrunning.operation_info) = {
-      response_type: "LongRunningRecognizeResponse"
-      metadata_type: "LongRunningRecognizeMetadata"
-    };
-  }
-
-  // Performs bidirectional streaming speech recognition: receive results while
-  // sending audio. This method is only available via the gRPC API (not REST).
-  rpc StreamingRecognize(stream StreamingRecognizeRequest) returns (stream StreamingRecognizeResponse) {
-  }
-}
-
-// The top-level message sent by the client for the `Recognize` method.
-message RecognizeRequest {
-  // Required. Provides information to the recognizer that specifies how to
-  // process the request.
-  RecognitionConfig config = 1 [(google.api.field_behavior) = REQUIRED];
-
-  // Required. The audio data to be recognized.
-  RecognitionAudio audio = 2 [(google.api.field_behavior) = REQUIRED];
-}
-
-// The top-level message sent by the client for the `LongRunningRecognize`
-// method.
-message LongRunningRecognizeRequest {
-  // Required. Provides information to the recognizer that specifies how to
-  // process the request.
-  RecognitionConfig config = 1 [(google.api.field_behavior) = REQUIRED];
-
-  // Required. The audio data to be recognized.
-  RecognitionAudio audio = 2 [(google.api.field_behavior) = REQUIRED];
-}
-
-// The top-level message sent by the client for the `StreamingRecognize` method.
-// Multiple `StreamingRecognizeRequest` messages are sent. The first message
-// must contain a `streaming_config` message and must not contain
-// `audio_content`. All subsequent messages must contain `audio_content` and
-// must not contain a `streaming_config` message.
-message StreamingRecognizeRequest {
-  // The streaming request, which is either a streaming config or audio content.
-  oneof streaming_request {
-    // Provides information to the recognizer that specifies how to process the
-    // request. The first `StreamingRecognizeRequest` message must contain a
-    // `streaming_config`  message.
-    StreamingRecognitionConfig streaming_config = 1;
-
-    // The audio data to be recognized. Sequential chunks of audio data are sent
-    // in sequential `StreamingRecognizeRequest` messages. The first
-    // `StreamingRecognizeRequest` message must not contain `audio_content` data
-    // and all subsequent `StreamingRecognizeRequest` messages must contain
-    // `audio_content` data. The audio bytes must be encoded as specified in
-    // `RecognitionConfig`. Note: as with all bytes fields, proto buffers use a
-    // pure binary representation (not base64). See
-    // [content limits](https://cloud.google.com/speech-to-text/quotas#content).
-    bytes audio_content = 2;
-  }
-}
-
-// Provides information to the recognizer that specifies how to process the
-// request.
-message StreamingRecognitionConfig {
-  // Required. Provides information to the recognizer that specifies how to
-  // process the request.
-  RecognitionConfig config = 1 [(google.api.field_behavior) = REQUIRED];
-
-  // If `false` or omitted, the recognizer will perform continuous
-  // recognition (continuing to wait for and process audio even if the user
-  // pauses speaking) until the client closes the input stream (gRPC API) or
-  // until the maximum time limit has been reached. May return multiple
-  // `StreamingRecognitionResult`s with the `is_final` flag set to `true`.
-  //
-  // If `true`, the recognizer will detect a single spoken utterance. When it
-  // detects that the user has paused or stopped speaking, it will return an
-  // `END_OF_SINGLE_UTTERANCE` event and cease recognition. It will return no
-  // more than one `StreamingRecognitionResult` with the `is_final` flag set to
-  // `true`.
-  bool single_utterance = 2;
-
-  // If `true`, interim results (tentative hypotheses) may be
-  // returned as they become available (these interim results are indicated with
-  // the `is_final=false` flag).
-  // If `false` or omitted, only `is_final=true` result(s) are returned.
-  bool interim_results = 3;
-}
-
-// Provides information to the recognizer that specifies how to process the
-// request.
-message RecognitionConfig {
-  // The encoding of the audio data sent in the request.
-  //
-  // All encodings support only 1 channel (mono) audio, unless the
-  // `audio_channel_count` and `enable_separate_recognition_per_channel` fields
-  // are set.
-  //
-  // For best results, the audio source should be captured and transmitted using
-  // a lossless encoding (`FLAC` or `LINEAR16`). The accuracy of the speech
-  // recognition can be reduced if lossy codecs are used to capture or transmit
-  // audio, particularly if background noise is present. Lossy codecs include
-  // `MULAW`, `AMR`, `AMR_WB`, `OGG_OPUS`, `SPEEX_WITH_HEADER_BYTE`, and `MP3`.
-  //
-  // The `FLAC` and `WAV` audio file formats include a header that describes the
-  // included audio content. You can request recognition for `WAV` files that
-  // contain either `LINEAR16` or `MULAW` encoded audio.
-  // If you send `FLAC` or `WAV` audio file format in
-  // your request, you do not need to specify an `AudioEncoding`; the audio
-  // encoding format is determined from the file header. If you specify
-  // an `AudioEncoding` when you send  send `FLAC` or `WAV` audio, the
-  // encoding configuration must match the encoding described in the audio
-  // header; otherwise the request returns an
-  // [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT] error code.
-  enum AudioEncoding {
-    // Not specified.
-    ENCODING_UNSPECIFIED = 0;
-
-    // Uncompressed 16-bit signed little-endian samples (Linear PCM).
-    LINEAR16 = 1;
-
-    // `FLAC` (Free Lossless Audio
-    // Codec) is the recommended encoding because it is
-    // lossless--therefore recognition is not compromised--and
-    // requires only about half the bandwidth of `LINEAR16`. `FLAC` stream
-    // encoding supports 16-bit and 24-bit samples, however, not all fields in
-    // `STREAMINFO` are supported.
-    FLAC = 2;
-
-    // 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
-    MULAW = 3;
-
-    // Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
-    AMR = 4;
-
-    // Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000.
-    AMR_WB = 5;
-
-    // Opus encoded audio frames in Ogg container
-    // ([OggOpus](https://wiki.xiph.org/OggOpus)).
-    // `sample_rate_hertz` must be one of 8000, 12000, 16000, 24000, or 48000.
-    OGG_OPUS = 6;
-
-    // Although the use of lossy encodings is not recommended, if a very low
-    // bitrate encoding is required, `OGG_OPUS` is highly preferred over
-    // Speex encoding. The [Speex](https://speex.org/)  encoding supported by
-    // Cloud Speech API has a header byte in each block, as in MIME type
-    // `audio/x-speex-with-header-byte`.
-    // It is a variant of the RTP Speex encoding defined in
-    // [RFC 5574](https://tools.ietf.org/html/rfc5574).
-    // The stream is a sequence of blocks, one block per RTP packet. Each block
-    // starts with a byte containing the length of the block, in bytes, followed
-    // by one or more frames of Speex data, padded to an integral number of
-    // bytes (octets) as specified in RFC 5574. In other words, each RTP header
-    // is replaced with a single byte containing the block length. Only Speex
-    // wideband is supported. `sample_rate_hertz` must be 16000.
-    SPEEX_WITH_HEADER_BYTE = 7;
-  }
-
-  // Encoding of audio data sent in all `RecognitionAudio` messages.
-  // This field is optional for `FLAC` and `WAV` audio files and required
-  // for all other audio formats. For details, see [AudioEncoding][google.cloud.speech.v1.RecognitionConfig.AudioEncoding].
-  AudioEncoding encoding = 1;
-
-  // Sample rate in Hertz of the audio data sent in all
-  // `RecognitionAudio` messages. Valid values are: 8000-48000.
-  // 16000 is optimal. For best results, set the sampling rate of the audio
-  // source to 16000 Hz. If that's not possible, use the native sample rate of
-  // the audio source (instead of re-sampling).
-  // This field is optional for FLAC and WAV audio files, but is
-  // required for all other audio formats. For details, see [AudioEncoding][google.cloud.speech.v1.RecognitionConfig.AudioEncoding].
-  int32 sample_rate_hertz = 2;
-
-  // The number of channels in the input audio data.
-  // ONLY set this for MULTI-CHANNEL recognition.
-  // Valid values for LINEAR16 and FLAC are `1`-`8`.
-  // Valid values for OGG_OPUS are '1'-'254'.
-  // Valid value for MULAW, AMR, AMR_WB and SPEEX_WITH_HEADER_BYTE is only `1`.
-  // If `0` or omitted, defaults to one channel (mono).
-  // Note: We only recognize the first channel by default.
-  // To perform independent recognition on each channel set
-  // `enable_separate_recognition_per_channel` to 'true'.
-  int32 audio_channel_count = 7;
-
-  // This needs to be set to `true` explicitly and `audio_channel_count` > 1
-  // to get each channel recognized separately. The recognition result will
-  // contain a `channel_tag` field to state which channel that result belongs
-  // to. If this is not true, we will only recognize the first channel. The
-  // request is billed cumulatively for all channels recognized:
-  // `audio_channel_count` multiplied by the length of the audio.
-  bool enable_separate_recognition_per_channel = 12;
-
-  // Required. The language of the supplied audio as a
-  // [BCP-47](https://www.rfc-editor.org/rfc/bcp/bcp47.txt) language tag.
-  // Example: "en-US".
-  // See [Language
-  // Support](https://cloud.google.com/speech-to-text/docs/languages) for a list
-  // of the currently supported language codes.
-  string language_code = 3 [(google.api.field_behavior) = REQUIRED];
-
-  // Maximum number of recognition hypotheses to be returned.
-  // Specifically, the maximum number of `SpeechRecognitionAlternative` messages
-  // within each `SpeechRecognitionResult`.
-  // The server may return fewer than `max_alternatives`.
-  // Valid values are `0`-`30`. A value of `0` or `1` will return a maximum of
-  // one. If omitted, will return a maximum of one.
-  int32 max_alternatives = 4;
-
-  // If set to `true`, the server will attempt to filter out
-  // profanities, replacing all but the initial character in each filtered word
-  // with asterisks, e.g. "f***". If set to `false` or omitted, profanities
-  // won't be filtered out.
-  bool profanity_filter = 5;
-
-  // Array of [SpeechContext][google.cloud.speech.v1.SpeechContext].
-  // A means to provide context to assist the speech recognition. For more
-  // information, see
-  // [speech
-  // adaptation](https://cloud.google.com/speech-to-text/docs/context-strength).
-  repeated SpeechContext speech_contexts = 6;
-
-  // If `true`, the top result includes a list of words and
-  // the start and end time offsets (timestamps) for those words. If
-  // `false`, no word-level time offset information is returned. The default is
-  // `false`.
-  bool enable_word_time_offsets = 8;
-
-  // If 'true', adds punctuation to recognition result hypotheses.
-  // This feature is only available in select languages. Setting this for
-  // requests in other languages has no effect at all.
-  // The default 'false' value does not add punctuation to result hypotheses.
-  // Note: This is currently offered as an experimental service, complimentary
-  // to all users. In the future this may be exclusively available as a
-  // premium feature.
-  bool enable_automatic_punctuation = 11;
-
-  // Config to enable speaker diarization and set additional
-  // parameters to make diarization better suited for your application.
-  // Note: When this is enabled, we send all the words from the beginning of the
-  // audio for the top alternative in every consecutive STREAMING responses.
-  // This is done in order to improve our speaker tags as our models learn to
-  // identify the speakers in the conversation over time.
-  // For non-streaming requests, the diarization results will be provided only
-  // in the top alternative of the FINAL SpeechRecognitionResult.
-  SpeakerDiarizationConfig diarization_config = 19;
-
-  // Metadata regarding this request.
-  RecognitionMetadata metadata = 9;
-
-  // Which model to select for the given request. Select the model
-  // best suited to your domain to get best results. If a model is not
-  // explicitly specified, then we auto-select a model based on the parameters
-  // in the RecognitionConfig.
-  // <table>
-  //   <tr>
-  //     <td><b>Model</b></td>
-  //     <td><b>Description</b></td>
-  //   </tr>
-  //   <tr>
-  //     <td><code>command_and_search</code></td>
-  //     <td>Best for short queries such as voice commands or voice search.</td>
-  //   </tr>
-  //   <tr>
-  //     <td><code>phone_call</code></td>
-  //     <td>Best for audio that originated from a phone call (typically
-  //     recorded at an 8khz sampling rate).</td>
-  //   </tr>
-  //   <tr>
-  //     <td><code>video</code></td>
-  //     <td>Best for audio that originated from from video or includes multiple
-  //         speakers. Ideally the audio is recorded at a 16khz or greater
-  //         sampling rate. This is a premium model that costs more than the
-  //         standard rate.</td>
-  //   </tr>
-  //   <tr>
-  //     <td><code>default</code></td>
-  //     <td>Best for audio that is not one of the specific audio models.
-  //         For example, long-form audio. Ideally the audio is high-fidelity,
-  //         recorded at a 16khz or greater sampling rate.</td>
-  //   </tr>
-  // </table>
-  string model = 13;
-
-  // Set to true to use an enhanced model for speech recognition.
-  // If `use_enhanced` is set to true and the `model` field is not set, then
-  // an appropriate enhanced model is chosen if an enhanced model exists for
-  // the audio.
-  //
-  // If `use_enhanced` is true and an enhanced version of the specified model
-  // does not exist, then the speech is recognized using the standard version
-  // of the specified model.
-  bool use_enhanced = 14;
-}
-
-// Config to enable speaker diarization.
-message SpeakerDiarizationConfig {
-  // If 'true', enables speaker detection for each recognized word in
-  // the top alternative of the recognition result using a speaker_tag provided
-  // in the WordInfo.
-  bool enable_speaker_diarization = 1;
-
-  // Minimum number of speakers in the conversation. This range gives you more
-  // flexibility by allowing the system to automatically determine the correct
-  // number of speakers. If not set, the default value is 2.
-  int32 min_speaker_count = 2;
-
-  // Maximum number of speakers in the conversation. This range gives you more
-  // flexibility by allowing the system to automatically determine the correct
-  // number of speakers. If not set, the default value is 6.
-  int32 max_speaker_count = 3;
-
-  // Unused.
-  int32 speaker_tag = 5
-      [(google.api.field_behavior) = OUTPUT_ONLY, deprecated = true];
-}
-
-// Description of audio data to be recognized.
-message RecognitionMetadata {
-  // Use case categories that the audio recognition request can be described
-  // by.
-  enum InteractionType {
-    // Use case is either unknown or is something other than one of the other
-    // values below.
-    INTERACTION_TYPE_UNSPECIFIED = 0;
-
-    // Multiple people in a conversation or discussion. For example in a
-    // meeting with two or more people actively participating. Typically
-    // all the primary people speaking would be in the same room (if not,
-    // see PHONE_CALL)
-    DISCUSSION = 1;
-
-    // One or more persons lecturing or presenting to others, mostly
-    // uninterrupted.
-    PRESENTATION = 2;
-
-    // A phone-call or video-conference in which two or more people, who are
-    // not in the same room, are actively participating.
-    PHONE_CALL = 3;
-
-    // A recorded message intended for another person to listen to.
-    VOICEMAIL = 4;
-
-    // Professionally produced audio (eg. TV Show, Podcast).
-    PROFESSIONALLY_PRODUCED = 5;
-
-    // Transcribe spoken questions and queries into text.
-    VOICE_SEARCH = 6;
-
-    // Transcribe voice commands, such as for controlling a device.
-    VOICE_COMMAND = 7;
-
-    // Transcribe speech to text to create a written document, such as a
-    // text-message, email or report.
-    DICTATION = 8;
-  }
-
-  // Enumerates the types of capture settings describing an audio file.
-  enum MicrophoneDistance {
-    // Audio type is not known.
-    MICROPHONE_DISTANCE_UNSPECIFIED = 0;
-
-    // The audio was captured from a closely placed microphone. Eg. phone,
-    // dictaphone, or handheld microphone. Generally if there speaker is within
-    // 1 meter of the microphone.
-    NEARFIELD = 1;
-
-    // The speaker if within 3 meters of the microphone.
-    MIDFIELD = 2;
-
-    // The speaker is more than 3 meters away from the microphone.
-    FARFIELD = 3;
-  }
-
-  // The original media the speech was recorded on.
-  enum OriginalMediaType {
-    // Unknown original media type.
-    ORIGINAL_MEDIA_TYPE_UNSPECIFIED = 0;
-
-    // The speech data is an audio recording.
-    AUDIO = 1;
-
-    // The speech data originally recorded on a video.
-    VIDEO = 2;
-  }
-
-  // The type of device the speech was recorded with.
-  enum RecordingDeviceType {
-    // The recording device is unknown.
-    RECORDING_DEVICE_TYPE_UNSPECIFIED = 0;
-
-    // Speech was recorded on a smartphone.
-    SMARTPHONE = 1;
-
-    // Speech was recorded using a personal computer or tablet.
-    PC = 2;
-
-    // Speech was recorded over a phone line.
-    PHONE_LINE = 3;
-
-    // Speech was recorded in a vehicle.
-    VEHICLE = 4;
-
-    // Speech was recorded outdoors.
-    OTHER_OUTDOOR_DEVICE = 5;
-
-    // Speech was recorded indoors.
-    OTHER_INDOOR_DEVICE = 6;
-  }
-
-  // The use case most closely describing the audio content to be recognized.
-  InteractionType interaction_type = 1;
-
-  // The industry vertical to which this speech recognition request most
-  // closely applies. This is most indicative of the topics contained
-  // in the audio.  Use the 6-digit NAICS code to identify the industry
-  // vertical - see https://www.naics.com/search/.
-  uint32 industry_naics_code_of_audio = 3;
-
-  // The audio type that most closely describes the audio being recognized.
-  MicrophoneDistance microphone_distance = 4;
-
-  // The original media the speech was recorded on.
-  OriginalMediaType original_media_type = 5;
-
-  // The type of device the speech was recorded with.
-  RecordingDeviceType recording_device_type = 6;
-
-  // The device used to make the recording.  Examples 'Nexus 5X' or
-  // 'Polycom SoundStation IP 6000' or 'POTS' or 'VoIP' or
-  // 'Cardioid Microphone'.
-  string recording_device_name = 7;
-
-  // Mime type of the original audio file.  For example `audio/m4a`,
-  // `audio/x-alaw-basic`, `audio/mp3`, `audio/3gpp`.
-  // A list of possible audio mime types is maintained at
-  // http://www.iana.org/assignments/media-types/media-types.xhtml#audio
-  string original_mime_type = 8;
-
-  // Description of the content. Eg. "Recordings of federal supreme court
-  // hearings from 2012".
-  string audio_topic = 10;
-}
-
-// Provides "hints" to the speech recognizer to favor specific words and phrases
-// in the results.
-message SpeechContext {
-  // A list of strings containing words and phrases "hints" so that
-  // the speech recognition is more likely to recognize them. This can be used
-  // to improve the accuracy for specific words and phrases, for example, if
-  // specific commands are typically spoken by the user. This can also be used
-  // to add additional words to the vocabulary of the recognizer. See
-  // [usage limits](https://cloud.google.com/speech-to-text/quotas#content).
-  //
-  // List items can also be set to classes for groups of words that represent
-  // common concepts that occur in natural language. For example, rather than
-  // providing phrase hints for every month of the year, using the $MONTH class
-  // improves the likelihood of correctly transcribing audio that includes
-  // months.
-  repeated string phrases = 1;
-}
-
-// Contains audio data in the encoding specified in the `RecognitionConfig`.
-// Either `content` or `uri` must be supplied. Supplying both or neither
-// returns [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT]. See
-// [content limits](https://cloud.google.com/speech-to-text/quotas#content).
-message RecognitionAudio {
-  // The audio source, which is either inline content or a Google Cloud
-  // Storage uri.
-  oneof audio_source {
-    // The audio data bytes encoded as specified in
-    // `RecognitionConfig`. Note: as with all bytes fields, proto buffers use a
-    // pure binary representation, whereas JSON representations use base64.
-    bytes content = 1;
-
-    // URI that points to a file that contains audio data bytes as specified in
-    // `RecognitionConfig`. The file must not be compressed (for example, gzip).
-    // Currently, only Google Cloud Storage URIs are
-    // supported, which must be specified in the following format:
-    // `gs://bucket_name/object_name` (other URI formats return
-    // [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT]). For more information, see
-    // [Request URIs](https://cloud.google.com/storage/docs/reference-uris).
-    string uri = 2;
-  }
-}
-
-// The only message returned to the client by the `Recognize` method. It
-// contains the result as zero or more sequential `SpeechRecognitionResult`
-// messages.
-message RecognizeResponse {
-  // Sequential list of transcription results corresponding to
-  // sequential portions of audio.
-  repeated SpeechRecognitionResult results = 2;
-}
-
-// The only message returned to the client by the `LongRunningRecognize` method.
-// It contains the result as zero or more sequential `SpeechRecognitionResult`
-// messages. It is included in the `result.response` field of the `Operation`
-// returned by the `GetOperation` call of the `google::longrunning::Operations`
-// service.
-message LongRunningRecognizeResponse {
-  // Sequential list of transcription results corresponding to
-  // sequential portions of audio.
-  repeated SpeechRecognitionResult results = 2;
-}
-
-// Describes the progress of a long-running `LongRunningRecognize` call. It is
-// included in the `metadata` field of the `Operation` returned by the
-// `GetOperation` call of the `google::longrunning::Operations` service.
-message LongRunningRecognizeMetadata {
-  // Approximate percentage of audio processed thus far. Guaranteed to be 100
-  // when the audio is fully processed and the results are available.
-  int32 progress_percent = 1;
-
-  // Time when the request was received.
-  google.protobuf.Timestamp start_time = 2;
-
-  // Time of the most recent processing update.
-  google.protobuf.Timestamp last_update_time = 3;
-}
-
-// `StreamingRecognizeResponse` is the only message returned to the client by
-// `StreamingRecognize`. A series of zero or more `StreamingRecognizeResponse`
-// messages are streamed back to the client. If there is no recognizable
-// audio, and `single_utterance` is set to false, then no messages are streamed
-// back to the client.
-//
-// Here's an example of a series of ten `StreamingRecognizeResponse`s that might
-// be returned while processing audio:
-//
-// 1. results { alternatives { transcript: "tube" } stability: 0.01 }
-//
-// 2. results { alternatives { transcript: "to be a" } stability: 0.01 }
-//
-// 3. results { alternatives { transcript: "to be" } stability: 0.9 }
-//    results { alternatives { transcript: " or not to be" } stability: 0.01 }
-//
-// 4. results { alternatives { transcript: "to be or not to be"
-//                             confidence: 0.92 }
-//              alternatives { transcript: "to bee or not to bee" }
-//              is_final: true }
-//
-// 5. results { alternatives { transcript: " that's" } stability: 0.01 }
-//
-// 6. results { alternatives { transcript: " that is" } stability: 0.9 }
-//    results { alternatives { transcript: " the question" } stability: 0.01 }
-//
-// 7. results { alternatives { transcript: " that is the question"
-//                             confidence: 0.98 }
-//              alternatives { transcript: " that was the question" }
-//              is_final: true }
-//
-// Notes:
-//
-// - Only two of the above responses #4 and #7 contain final results; they are
-//   indicated by `is_final: true`. Concatenating these together generates the
-//   full transcript: "to be or not to be that is the question".
-//
-// - The others contain interim `results`. #3 and #6 contain two interim
-//   `results`: the first portion has a high stability and is less likely to
-//   change; the second portion has a low stability and is very likely to
-//   change. A UI designer might choose to show only high stability `results`.
-//
-// - The specific `stability` and `confidence` values shown above are only for
-//   illustrative purposes. Actual values may vary.
-//
-// - In each response, only one of these fields will be set:
-//     `error`,
-//     `speech_event_type`, or
-//     one or more (repeated) `results`.
-message StreamingRecognizeResponse {
-  // Indicates the type of speech event.
-  enum SpeechEventType {
-    // No speech event specified.
-    SPEECH_EVENT_UNSPECIFIED = 0;
-
-    // This event indicates that the server has detected the end of the user's
-    // speech utterance and expects no additional speech. Therefore, the server
-    // will not process additional audio (although it may subsequently return
-    // additional results). The client should stop sending additional audio
-    // data, half-close the gRPC connection, and wait for any additional results
-    // until the server closes the gRPC connection. This event is only sent if
-    // `single_utterance` was set to `true`, and is not used otherwise.
-    END_OF_SINGLE_UTTERANCE = 1;
-  }
-
-  // If set, returns a [google.rpc.Status][google.rpc.Status] message that
-  // specifies the error for the operation.
-  google.rpc.Status error = 1;
-
-  // This repeated list contains zero or more results that
-  // correspond to consecutive portions of the audio currently being processed.
-  // It contains zero or one `is_final=true` result (the newly settled portion),
-  // followed by zero or more `is_final=false` results (the interim results).
-  repeated StreamingRecognitionResult results = 2;
-
-  // Indicates the type of speech event.
-  SpeechEventType speech_event_type = 4;
-}
-
-// A streaming speech recognition result corresponding to a portion of the audio
-// that is currently being processed.
-message StreamingRecognitionResult {
-  // May contain one or more recognition hypotheses (up to the
-  // maximum specified in `max_alternatives`).
-  // These alternatives are ordered in terms of accuracy, with the top (first)
-  // alternative being the most probable, as ranked by the recognizer.
-  repeated SpeechRecognitionAlternative alternatives = 1;
-
-  // If `false`, this `StreamingRecognitionResult` represents an
-  // interim result that may change. If `true`, this is the final time the
-  // speech service will return this particular `StreamingRecognitionResult`,
-  // the recognizer will not return any further hypotheses for this portion of
-  // the transcript and corresponding audio.
-  bool is_final = 2;
-
-  // An estimate of the likelihood that the recognizer will not
-  // change its guess about this interim result. Values range from 0.0
-  // (completely unstable) to 1.0 (completely stable).
-  // This field is only provided for interim results (`is_final=false`).
-  // The default of 0.0 is a sentinel value indicating `stability` was not set.
-  float stability = 3;
-
-  // Time offset of the end of this result relative to the
-  // beginning of the audio.
-  google.protobuf.Duration result_end_time = 4;
-
-  // For multi-channel audio, this is the channel number corresponding to the
-  // recognized result for the audio from that channel.
-  // For audio_channel_count = N, its output values can range from '1' to 'N'.
-  int32 channel_tag = 5;
-
-  // The [BCP-47](https://www.rfc-editor.org/rfc/bcp/bcp47.txt) language tag of
-  // the language in this result. This language code was detected to have the
-  // most likelihood of being spoken in the audio.
-  string language_code = 6
-    [(google.api.field_behavior) = OUTPUT_ONLY];
-}
-
-// A speech recognition result corresponding to a portion of the audio.
-message SpeechRecognitionResult {
-  // May contain one or more recognition hypotheses (up to the
-  // maximum specified in `max_alternatives`).
-  // These alternatives are ordered in terms of accuracy, with the top (first)
-  // alternative being the most probable, as ranked by the recognizer.
-  repeated SpeechRecognitionAlternative alternatives = 1;
-
-  // For multi-channel audio, this is the channel number corresponding to the
-  // recognized result for the audio from that channel.
-  // For audio_channel_count = N, its output values can range from '1' to 'N'.
-  int32 channel_tag = 2;
-}
-
-// Alternative hypotheses (a.k.a. n-best list).
-message SpeechRecognitionAlternative {
-  // Transcript text representing the words that the user spoke.
-  string transcript = 1;
-
-  // The confidence estimate between 0.0 and 1.0. A higher number
-  // indicates an estimated greater likelihood that the recognized words are
-  // correct. This field is set only for the top alternative of a non-streaming
-  // result or, of a streaming result where `is_final=true`.
-  // This field is not guaranteed to be accurate and users should not rely on it
-  // to be always provided.
-  // The default of 0.0 is a sentinel value indicating `confidence` was not set.
-  float confidence = 2;
-
-  // A list of word-specific information for each recognized word.
-  // Note: When `enable_speaker_diarization` is true, you will see all the words
-  // from the beginning of the audio.
-  repeated WordInfo words = 3;
-}
-
-// Word-specific information for recognized words.
-message WordInfo {
-  // Time offset relative to the beginning of the audio,
-  // and corresponding to the start of the spoken word.
-  // This field is only set if `enable_word_time_offsets=true` and only
-  // in the top hypothesis.
-  // This is an experimental feature and the accuracy of the time offset can
-  // vary.
-  google.protobuf.Duration start_time = 1;
-
-  // Time offset relative to the beginning of the audio,
-  // and corresponding to the end of the spoken word.
-  // This field is only set if `enable_word_time_offsets=true` and only
-  // in the top hypothesis.
-  // This is an experimental feature and the accuracy of the time offset can
-  // vary.
-  google.protobuf.Duration end_time = 2;
-
-  // The word corresponding to this set of information.
-  string word = 3;
-
-  // A distinct integer value is assigned for every speaker within
-  // the audio. This field specifies which one of those speakers was detected to
-  // have spoken this word. Value ranges from '1' to diarization_speaker_count.
-  // speaker_tag is set if enable_speaker_diarization = 'true' and only in the
-  // top alternative.
-  int32 speaker_tag = 5
-    [(google.api.field_behavior) = OUTPUT_ONLY];
-}
diff --git a/google/cloud/speech_v1/proto/cloud_speech_pb2.py b/google/cloud/speech_v1/proto/cloud_speech_pb2.py
deleted file mode 100644
index d73b7c0e..00000000
--- a/google/cloud/speech_v1/proto/cloud_speech_pb2.py
+++ /dev/null
@@ -1,2640 +0,0 @@
-# -*- coding: utf-8 -*-
-# Generated by the protocol buffer compiler.  DO NOT EDIT!
-# source: google/cloud/speech_v1/proto/cloud_speech.proto
-
-import sys
-
-_b = sys.version_info[0] < 3 and (lambda x: x) or (lambda x: x.encode("latin1"))
-from google.protobuf import descriptor as _descriptor
-from google.protobuf import message as _message
-from google.protobuf import reflection as _reflection
-from google.protobuf import symbol_database as _symbol_database
-
-# @@protoc_insertion_point(imports)
-
-_sym_db = _symbol_database.Default()
-
-
-from google.api import annotations_pb2 as google_dot_api_dot_annotations__pb2
-from google.api import client_pb2 as google_dot_api_dot_client__pb2
-from google.api import field_behavior_pb2 as google_dot_api_dot_field__behavior__pb2
-from google.longrunning import (
-    operations_pb2 as google_dot_longrunning_dot_operations__pb2,
-)
-from google.protobuf import any_pb2 as google_dot_protobuf_dot_any__pb2
-from google.protobuf import duration_pb2 as google_dot_protobuf_dot_duration__pb2
-from google.protobuf import timestamp_pb2 as google_dot_protobuf_dot_timestamp__pb2
-from google.rpc import status_pb2 as google_dot_rpc_dot_status__pb2
-
-
-DESCRIPTOR = _descriptor.FileDescriptor(
-    name="google/cloud/speech_v1/proto/cloud_speech.proto",
-    package="google.cloud.speech.v1",
-    syntax="proto3",
-    serialized_options=_b(
-        "\n\032com.google.cloud.speech.v1B\013SpeechProtoP\001Z<google.golang.org/genproto/googleapis/cloud/speech/v1;speech\370\001\001\242\002\003GCS"
-    ),
-    serialized_pb=_b(
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-    ),
-    dependencies=[
-        google_dot_api_dot_annotations__pb2.DESCRIPTOR,
-        google_dot_api_dot_client__pb2.DESCRIPTOR,
-        google_dot_api_dot_field__behavior__pb2.DESCRIPTOR,
-        google_dot_longrunning_dot_operations__pb2.DESCRIPTOR,
-        google_dot_protobuf_dot_any__pb2.DESCRIPTOR,
-        google_dot_protobuf_dot_duration__pb2.DESCRIPTOR,
-        google_dot_protobuf_dot_timestamp__pb2.DESCRIPTOR,
-        google_dot_rpc_dot_status__pb2.DESCRIPTOR,
-    ],
-)
-
-
-_RECOGNITIONCONFIG_AUDIOENCODING = _descriptor.EnumDescriptor(
-    name="AudioEncoding",
-    full_name="google.cloud.speech.v1.RecognitionConfig.AudioEncoding",
-    filename=None,
-    file=DESCRIPTOR,
-    values=[
-        _descriptor.EnumValueDescriptor(
-            name="ENCODING_UNSPECIFIED",
-            index=0,
-            number=0,
-            serialized_options=None,
-            type=None,
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="LINEAR16", index=1, number=1, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="FLAC", index=2, number=2, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="MULAW", index=3, number=3, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="AMR", index=4, number=4, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="AMR_WB", index=5, number=5, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="OGG_OPUS", index=6, number=6, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="SPEEX_WITH_HEADER_BYTE",
-            index=7,
-            number=7,
-            serialized_options=None,
-            type=None,
-        ),
-    ],
-    containing_type=None,
-    serialized_options=None,
-    serialized_start=1521,
-    serialized_end=1660,
-)
-_sym_db.RegisterEnumDescriptor(_RECOGNITIONCONFIG_AUDIOENCODING)
-
-_RECOGNITIONMETADATA_INTERACTIONTYPE = _descriptor.EnumDescriptor(
-    name="InteractionType",
-    full_name="google.cloud.speech.v1.RecognitionMetadata.InteractionType",
-    filename=None,
-    file=DESCRIPTOR,
-    values=[
-        _descriptor.EnumValueDescriptor(
-            name="INTERACTION_TYPE_UNSPECIFIED",
-            index=0,
-            number=0,
-            serialized_options=None,
-            type=None,
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="DISCUSSION", index=1, number=1, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="PRESENTATION", index=2, number=2, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="PHONE_CALL", index=3, number=3, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="VOICEMAIL", index=4, number=4, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="PROFESSIONALLY_PRODUCED",
-            index=5,
-            number=5,
-            serialized_options=None,
-            type=None,
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="VOICE_SEARCH", index=6, number=6, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="VOICE_COMMAND", index=7, number=7, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="DICTATION", index=8, number=8, serialized_options=None, type=None
-        ),
-    ],
-    containing_type=None,
-    serialized_options=None,
-    serialized_start=2320,
-    serialized_end=2517,
-)
-_sym_db.RegisterEnumDescriptor(_RECOGNITIONMETADATA_INTERACTIONTYPE)
-
-_RECOGNITIONMETADATA_MICROPHONEDISTANCE = _descriptor.EnumDescriptor(
-    name="MicrophoneDistance",
-    full_name="google.cloud.speech.v1.RecognitionMetadata.MicrophoneDistance",
-    filename=None,
-    file=DESCRIPTOR,
-    values=[
-        _descriptor.EnumValueDescriptor(
-            name="MICROPHONE_DISTANCE_UNSPECIFIED",
-            index=0,
-            number=0,
-            serialized_options=None,
-            type=None,
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="NEARFIELD", index=1, number=1, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="MIDFIELD", index=2, number=2, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="FARFIELD", index=3, number=3, serialized_options=None, type=None
-        ),
-    ],
-    containing_type=None,
-    serialized_options=None,
-    serialized_start=2519,
-    serialized_end=2619,
-)
-_sym_db.RegisterEnumDescriptor(_RECOGNITIONMETADATA_MICROPHONEDISTANCE)
-
-_RECOGNITIONMETADATA_ORIGINALMEDIATYPE = _descriptor.EnumDescriptor(
-    name="OriginalMediaType",
-    full_name="google.cloud.speech.v1.RecognitionMetadata.OriginalMediaType",
-    filename=None,
-    file=DESCRIPTOR,
-    values=[
-        _descriptor.EnumValueDescriptor(
-            name="ORIGINAL_MEDIA_TYPE_UNSPECIFIED",
-            index=0,
-            number=0,
-            serialized_options=None,
-            type=None,
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="AUDIO", index=1, number=1, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="VIDEO", index=2, number=2, serialized_options=None, type=None
-        ),
-    ],
-    containing_type=None,
-    serialized_options=None,
-    serialized_start=2621,
-    serialized_end=2699,
-)
-_sym_db.RegisterEnumDescriptor(_RECOGNITIONMETADATA_ORIGINALMEDIATYPE)
-
-_RECOGNITIONMETADATA_RECORDINGDEVICETYPE = _descriptor.EnumDescriptor(
-    name="RecordingDeviceType",
-    full_name="google.cloud.speech.v1.RecognitionMetadata.RecordingDeviceType",
-    filename=None,
-    file=DESCRIPTOR,
-    values=[
-        _descriptor.EnumValueDescriptor(
-            name="RECORDING_DEVICE_TYPE_UNSPECIFIED",
-            index=0,
-            number=0,
-            serialized_options=None,
-            type=None,
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="SMARTPHONE", index=1, number=1, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="PC", index=2, number=2, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="PHONE_LINE", index=3, number=3, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="VEHICLE", index=4, number=4, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="OTHER_OUTDOOR_DEVICE",
-            index=5,
-            number=5,
-            serialized_options=None,
-            type=None,
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="OTHER_INDOOR_DEVICE",
-            index=6,
-            number=6,
-            serialized_options=None,
-            type=None,
-        ),
-    ],
-    containing_type=None,
-    serialized_options=None,
-    serialized_start=2702,
-    serialized_end=2866,
-)
-_sym_db.RegisterEnumDescriptor(_RECOGNITIONMETADATA_RECORDINGDEVICETYPE)
-
-_STREAMINGRECOGNIZERESPONSE_SPEECHEVENTTYPE = _descriptor.EnumDescriptor(
-    name="SpeechEventType",
-    full_name="google.cloud.speech.v1.StreamingRecognizeResponse.SpeechEventType",
-    filename=None,
-    file=DESCRIPTOR,
-    values=[
-        _descriptor.EnumValueDescriptor(
-            name="SPEECH_EVENT_UNSPECIFIED",
-            index=0,
-            number=0,
-            serialized_options=None,
-            type=None,
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="END_OF_SINGLE_UTTERANCE",
-            index=1,
-            number=1,
-            serialized_options=None,
-            type=None,
-        ),
-    ],
-    containing_type=None,
-    serialized_options=None,
-    serialized_start=3548,
-    serialized_end=3624,
-)
-_sym_db.RegisterEnumDescriptor(_STREAMINGRECOGNIZERESPONSE_SPEECHEVENTTYPE)
-
-
-_RECOGNIZEREQUEST = _descriptor.Descriptor(
-    name="RecognizeRequest",
-    full_name="google.cloud.speech.v1.RecognizeRequest",
-    filename=None,
-    file=DESCRIPTOR,
-    containing_type=None,
-    fields=[
-        _descriptor.FieldDescriptor(
-            name="config",
-            full_name="google.cloud.speech.v1.RecognizeRequest.config",
-            index=0,
-            number=1,
-            type=11,
-            cpp_type=10,
-            label=1,
-            has_default_value=False,
-            default_value=None,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=_b("\340A\002"),
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="audio",
-            full_name="google.cloud.speech.v1.RecognizeRequest.audio",
-            index=1,
-            number=2,
-            type=11,
-            cpp_type=10,
-            label=1,
-            has_default_value=False,
-            default_value=None,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=_b("\340A\002"),
-            file=DESCRIPTOR,
-        ),
-    ],
-    extensions=[],
-    nested_types=[],
-    enum_types=[],
-    serialized_options=None,
-    is_extendable=False,
-    syntax="proto3",
-    extension_ranges=[],
-    oneofs=[],
-    serialized_start=318,
-    serialized_end=462,
-)
-
-
-_LONGRUNNINGRECOGNIZEREQUEST = _descriptor.Descriptor(
-    name="LongRunningRecognizeRequest",
-    full_name="google.cloud.speech.v1.LongRunningRecognizeRequest",
-    filename=None,
-    file=DESCRIPTOR,
-    containing_type=None,
-    fields=[
-        _descriptor.FieldDescriptor(
-            name="config",
-            full_name="google.cloud.speech.v1.LongRunningRecognizeRequest.config",
-            index=0,
-            number=1,
-            type=11,
-            cpp_type=10,
-            label=1,
-            has_default_value=False,
-            default_value=None,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=_b("\340A\002"),
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="audio",
-            full_name="google.cloud.speech.v1.LongRunningRecognizeRequest.audio",
-            index=1,
-            number=2,
-            type=11,
-            cpp_type=10,
-            label=1,
-            has_default_value=False,
-            default_value=None,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=_b("\340A\002"),
-            file=DESCRIPTOR,
-        ),
-    ],
-    extensions=[],
-    nested_types=[],
-    enum_types=[],
-    serialized_options=None,
-    is_extendable=False,
-    syntax="proto3",
-    extension_ranges=[],
-    oneofs=[],
-    serialized_start=465,
-    serialized_end=620,
-)
-
-
-_STREAMINGRECOGNIZEREQUEST = _descriptor.Descriptor(
-    name="StreamingRecognizeRequest",
-    full_name="google.cloud.speech.v1.StreamingRecognizeRequest",
-    filename=None,
-    file=DESCRIPTOR,
-    containing_type=None,
-    fields=[
-        _descriptor.FieldDescriptor(
-            name="streaming_config",
-            full_name="google.cloud.speech.v1.StreamingRecognizeRequest.streaming_config",
-            index=0,
-            number=1,
-            type=11,
-            cpp_type=10,
-            label=1,
-            has_default_value=False,
-            default_value=None,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="audio_content",
-            full_name="google.cloud.speech.v1.StreamingRecognizeRequest.audio_content",
-            index=1,
-            number=2,
-            type=12,
-            cpp_type=9,
-            label=1,
-            has_default_value=False,
-            default_value=_b(""),
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-    ],
-    extensions=[],
-    nested_types=[],
-    enum_types=[],
-    serialized_options=None,
-    is_extendable=False,
-    syntax="proto3",
-    extension_ranges=[],
-    oneofs=[
-        _descriptor.OneofDescriptor(
-            name="streaming_request",
-            full_name="google.cloud.speech.v1.StreamingRecognizeRequest.streaming_request",
-            index=0,
-            containing_type=None,
-            fields=[],
-        )
-    ],
-    serialized_start=623,
-    serialized_end=776,
-)
-
-
-_STREAMINGRECOGNITIONCONFIG = _descriptor.Descriptor(
-    name="StreamingRecognitionConfig",
-    full_name="google.cloud.speech.v1.StreamingRecognitionConfig",
-    filename=None,
-    file=DESCRIPTOR,
-    containing_type=None,
-    fields=[
-        _descriptor.FieldDescriptor(
-            name="config",
-            full_name="google.cloud.speech.v1.StreamingRecognitionConfig.config",
-            index=0,
-            number=1,
-            type=11,
-            cpp_type=10,
-            label=1,
-            has_default_value=False,
-            default_value=None,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=_b("\340A\002"),
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="single_utterance",
-            full_name="google.cloud.speech.v1.StreamingRecognitionConfig.single_utterance",
-            index=1,
-            number=2,
-            type=8,
-            cpp_type=7,
-            label=1,
-            has_default_value=False,
-            default_value=False,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="interim_results",
-            full_name="google.cloud.speech.v1.StreamingRecognitionConfig.interim_results",
-            index=2,
-            number=3,
-            type=8,
-            cpp_type=7,
-            label=1,
-            has_default_value=False,
-            default_value=False,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-    ],
-    extensions=[],
-    nested_types=[],
-    enum_types=[],
-    serialized_options=None,
-    is_extendable=False,
-    syntax="proto3",
-    extension_ranges=[],
-    oneofs=[],
-    serialized_start=779,
-    serialized_end=922,
-)
-
-
-_RECOGNITIONCONFIG = _descriptor.Descriptor(
-    name="RecognitionConfig",
-    full_name="google.cloud.speech.v1.RecognitionConfig",
-    filename=None,
-    file=DESCRIPTOR,
-    containing_type=None,
-    fields=[
-        _descriptor.FieldDescriptor(
-            name="encoding",
-            full_name="google.cloud.speech.v1.RecognitionConfig.encoding",
-            index=0,
-            number=1,
-            type=14,
-            cpp_type=8,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="sample_rate_hertz",
-            full_name="google.cloud.speech.v1.RecognitionConfig.sample_rate_hertz",
-            index=1,
-            number=2,
-            type=5,
-            cpp_type=1,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="audio_channel_count",
-            full_name="google.cloud.speech.v1.RecognitionConfig.audio_channel_count",
-            index=2,
-            number=7,
-            type=5,
-            cpp_type=1,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="enable_separate_recognition_per_channel",
-            full_name="google.cloud.speech.v1.RecognitionConfig.enable_separate_recognition_per_channel",
-            index=3,
-            number=12,
-            type=8,
-            cpp_type=7,
-            label=1,
-            has_default_value=False,
-            default_value=False,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="language_code",
-            full_name="google.cloud.speech.v1.RecognitionConfig.language_code",
-            index=4,
-            number=3,
-            type=9,
-            cpp_type=9,
-            label=1,
-            has_default_value=False,
-            default_value=_b("").decode("utf-8"),
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=_b("\340A\002"),
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="max_alternatives",
-            full_name="google.cloud.speech.v1.RecognitionConfig.max_alternatives",
-            index=5,
-            number=4,
-            type=5,
-            cpp_type=1,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="profanity_filter",
-            full_name="google.cloud.speech.v1.RecognitionConfig.profanity_filter",
-            index=6,
-            number=5,
-            type=8,
-            cpp_type=7,
-            label=1,
-            has_default_value=False,
-            default_value=False,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="speech_contexts",
-            full_name="google.cloud.speech.v1.RecognitionConfig.speech_contexts",
-            index=7,
-            number=6,
-            type=11,
-            cpp_type=10,
-            label=3,
-            has_default_value=False,
-            default_value=[],
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="enable_word_time_offsets",
-            full_name="google.cloud.speech.v1.RecognitionConfig.enable_word_time_offsets",
-            index=8,
-            number=8,
-            type=8,
-            cpp_type=7,
-            label=1,
-            has_default_value=False,
-            default_value=False,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="enable_automatic_punctuation",
-            full_name="google.cloud.speech.v1.RecognitionConfig.enable_automatic_punctuation",
-            index=9,
-            number=11,
-            type=8,
-            cpp_type=7,
-            label=1,
-            has_default_value=False,
-            default_value=False,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="diarization_config",
-            full_name="google.cloud.speech.v1.RecognitionConfig.diarization_config",
-            index=10,
-            number=19,
-            type=11,
-            cpp_type=10,
-            label=1,
-            has_default_value=False,
-            default_value=None,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="metadata",
-            full_name="google.cloud.speech.v1.RecognitionConfig.metadata",
-            index=11,
-            number=9,
-            type=11,
-            cpp_type=10,
-            label=1,
-            has_default_value=False,
-            default_value=None,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="model",
-            full_name="google.cloud.speech.v1.RecognitionConfig.model",
-            index=12,
-            number=13,
-            type=9,
-            cpp_type=9,
-            label=1,
-            has_default_value=False,
-            default_value=_b("").decode("utf-8"),
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="use_enhanced",
-            full_name="google.cloud.speech.v1.RecognitionConfig.use_enhanced",
-            index=13,
-            number=14,
-            type=8,
-            cpp_type=7,
-            label=1,
-            has_default_value=False,
-            default_value=False,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-    ],
-    extensions=[],
-    nested_types=[],
-    enum_types=[_RECOGNITIONCONFIG_AUDIOENCODING],
-    serialized_options=None,
-    is_extendable=False,
-    syntax="proto3",
-    extension_ranges=[],
-    oneofs=[],
-    serialized_start=925,
-    serialized_end=1660,
-)
-
-
-_SPEAKERDIARIZATIONCONFIG = _descriptor.Descriptor(
-    name="SpeakerDiarizationConfig",
-    full_name="google.cloud.speech.v1.SpeakerDiarizationConfig",
-    filename=None,
-    file=DESCRIPTOR,
-    containing_type=None,
-    fields=[
-        _descriptor.FieldDescriptor(
-            name="enable_speaker_diarization",
-            full_name="google.cloud.speech.v1.SpeakerDiarizationConfig.enable_speaker_diarization",
-            index=0,
-            number=1,
-            type=8,
-            cpp_type=7,
-            label=1,
-            has_default_value=False,
-            default_value=False,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="min_speaker_count",
-            full_name="google.cloud.speech.v1.SpeakerDiarizationConfig.min_speaker_count",
-            index=1,
-            number=2,
-            type=5,
-            cpp_type=1,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="max_speaker_count",
-            full_name="google.cloud.speech.v1.SpeakerDiarizationConfig.max_speaker_count",
-            index=2,
-            number=3,
-            type=5,
-            cpp_type=1,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="speaker_tag",
-            full_name="google.cloud.speech.v1.SpeakerDiarizationConfig.speaker_tag",
-            index=3,
-            number=5,
-            type=5,
-            cpp_type=1,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=_b("\030\001\340A\003"),
-            file=DESCRIPTOR,
-        ),
-    ],
-    extensions=[],
-    nested_types=[],
-    enum_types=[],
-    serialized_options=None,
-    is_extendable=False,
-    syntax="proto3",
-    extension_ranges=[],
-    oneofs=[],
-    serialized_start=1663,
-    serialized_end=1807,
-)
-
-
-_RECOGNITIONMETADATA = _descriptor.Descriptor(
-    name="RecognitionMetadata",
-    full_name="google.cloud.speech.v1.RecognitionMetadata",
-    filename=None,
-    file=DESCRIPTOR,
-    containing_type=None,
-    fields=[
-        _descriptor.FieldDescriptor(
-            name="interaction_type",
-            full_name="google.cloud.speech.v1.RecognitionMetadata.interaction_type",
-            index=0,
-            number=1,
-            type=14,
-            cpp_type=8,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="industry_naics_code_of_audio",
-            full_name="google.cloud.speech.v1.RecognitionMetadata.industry_naics_code_of_audio",
-            index=1,
-            number=3,
-            type=13,
-            cpp_type=3,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="microphone_distance",
-            full_name="google.cloud.speech.v1.RecognitionMetadata.microphone_distance",
-            index=2,
-            number=4,
-            type=14,
-            cpp_type=8,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="original_media_type",
-            full_name="google.cloud.speech.v1.RecognitionMetadata.original_media_type",
-            index=3,
-            number=5,
-            type=14,
-            cpp_type=8,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="recording_device_type",
-            full_name="google.cloud.speech.v1.RecognitionMetadata.recording_device_type",
-            index=4,
-            number=6,
-            type=14,
-            cpp_type=8,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="recording_device_name",
-            full_name="google.cloud.speech.v1.RecognitionMetadata.recording_device_name",
-            index=5,
-            number=7,
-            type=9,
-            cpp_type=9,
-            label=1,
-            has_default_value=False,
-            default_value=_b("").decode("utf-8"),
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="original_mime_type",
-            full_name="google.cloud.speech.v1.RecognitionMetadata.original_mime_type",
-            index=6,
-            number=8,
-            type=9,
-            cpp_type=9,
-            label=1,
-            has_default_value=False,
-            default_value=_b("").decode("utf-8"),
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="audio_topic",
-            full_name="google.cloud.speech.v1.RecognitionMetadata.audio_topic",
-            index=7,
-            number=10,
-            type=9,
-            cpp_type=9,
-            label=1,
-            has_default_value=False,
-            default_value=_b("").decode("utf-8"),
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-    ],
-    extensions=[],
-    nested_types=[],
-    enum_types=[
-        _RECOGNITIONMETADATA_INTERACTIONTYPE,
-        _RECOGNITIONMETADATA_MICROPHONEDISTANCE,
-        _RECOGNITIONMETADATA_ORIGINALMEDIATYPE,
-        _RECOGNITIONMETADATA_RECORDINGDEVICETYPE,
-    ],
-    serialized_options=None,
-    is_extendable=False,
-    syntax="proto3",
-    extension_ranges=[],
-    oneofs=[],
-    serialized_start=1810,
-    serialized_end=2866,
-)
-
-
-_SPEECHCONTEXT = _descriptor.Descriptor(
-    name="SpeechContext",
-    full_name="google.cloud.speech.v1.SpeechContext",
-    filename=None,
-    file=DESCRIPTOR,
-    containing_type=None,
-    fields=[
-        _descriptor.FieldDescriptor(
-            name="phrases",
-            full_name="google.cloud.speech.v1.SpeechContext.phrases",
-            index=0,
-            number=1,
-            type=9,
-            cpp_type=9,
-            label=3,
-            has_default_value=False,
-            default_value=[],
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        )
-    ],
-    extensions=[],
-    nested_types=[],
-    enum_types=[],
-    serialized_options=None,
-    is_extendable=False,
-    syntax="proto3",
-    extension_ranges=[],
-    oneofs=[],
-    serialized_start=2868,
-    serialized_end=2900,
-)
-
-
-_RECOGNITIONAUDIO = _descriptor.Descriptor(
-    name="RecognitionAudio",
-    full_name="google.cloud.speech.v1.RecognitionAudio",
-    filename=None,
-    file=DESCRIPTOR,
-    containing_type=None,
-    fields=[
-        _descriptor.FieldDescriptor(
-            name="content",
-            full_name="google.cloud.speech.v1.RecognitionAudio.content",
-            index=0,
-            number=1,
-            type=12,
-            cpp_type=9,
-            label=1,
-            has_default_value=False,
-            default_value=_b(""),
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="uri",
-            full_name="google.cloud.speech.v1.RecognitionAudio.uri",
-            index=1,
-            number=2,
-            type=9,
-            cpp_type=9,
-            label=1,
-            has_default_value=False,
-            default_value=_b("").decode("utf-8"),
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-    ],
-    extensions=[],
-    nested_types=[],
-    enum_types=[],
-    serialized_options=None,
-    is_extendable=False,
-    syntax="proto3",
-    extension_ranges=[],
-    oneofs=[
-        _descriptor.OneofDescriptor(
-            name="audio_source",
-            full_name="google.cloud.speech.v1.RecognitionAudio.audio_source",
-            index=0,
-            containing_type=None,
-            fields=[],
-        )
-    ],
-    serialized_start=2902,
-    serialized_end=2970,
-)
-
-
-_RECOGNIZERESPONSE = _descriptor.Descriptor(
-    name="RecognizeResponse",
-    full_name="google.cloud.speech.v1.RecognizeResponse",
-    filename=None,
-    file=DESCRIPTOR,
-    containing_type=None,
-    fields=[
-        _descriptor.FieldDescriptor(
-            name="results",
-            full_name="google.cloud.speech.v1.RecognizeResponse.results",
-            index=0,
-            number=2,
-            type=11,
-            cpp_type=10,
-            label=3,
-            has_default_value=False,
-            default_value=[],
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        )
-    ],
-    extensions=[],
-    nested_types=[],
-    enum_types=[],
-    serialized_options=None,
-    is_extendable=False,
-    syntax="proto3",
-    extension_ranges=[],
-    oneofs=[],
-    serialized_start=2972,
-    serialized_end=3057,
-)
-
-
-_LONGRUNNINGRECOGNIZERESPONSE = _descriptor.Descriptor(
-    name="LongRunningRecognizeResponse",
-    full_name="google.cloud.speech.v1.LongRunningRecognizeResponse",
-    filename=None,
-    file=DESCRIPTOR,
-    containing_type=None,
-    fields=[
-        _descriptor.FieldDescriptor(
-            name="results",
-            full_name="google.cloud.speech.v1.LongRunningRecognizeResponse.results",
-            index=0,
-            number=2,
-            type=11,
-            cpp_type=10,
-            label=3,
-            has_default_value=False,
-            default_value=[],
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        )
-    ],
-    extensions=[],
-    nested_types=[],
-    enum_types=[],
-    serialized_options=None,
-    is_extendable=False,
-    syntax="proto3",
-    extension_ranges=[],
-    oneofs=[],
-    serialized_start=3059,
-    serialized_end=3155,
-)
-
-
-_LONGRUNNINGRECOGNIZEMETADATA = _descriptor.Descriptor(
-    name="LongRunningRecognizeMetadata",
-    full_name="google.cloud.speech.v1.LongRunningRecognizeMetadata",
-    filename=None,
-    file=DESCRIPTOR,
-    containing_type=None,
-    fields=[
-        _descriptor.FieldDescriptor(
-            name="progress_percent",
-            full_name="google.cloud.speech.v1.LongRunningRecognizeMetadata.progress_percent",
-            index=0,
-            number=1,
-            type=5,
-            cpp_type=1,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="start_time",
-            full_name="google.cloud.speech.v1.LongRunningRecognizeMetadata.start_time",
-            index=1,
-            number=2,
-            type=11,
-            cpp_type=10,
-            label=1,
-            has_default_value=False,
-            default_value=None,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="last_update_time",
-            full_name="google.cloud.speech.v1.LongRunningRecognizeMetadata.last_update_time",
-            index=2,
-            number=3,
-            type=11,
-            cpp_type=10,
-            label=1,
-            has_default_value=False,
-            default_value=None,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-    ],
-    extensions=[],
-    nested_types=[],
-    enum_types=[],
-    serialized_options=None,
-    is_extendable=False,
-    syntax="proto3",
-    extension_ranges=[],
-    oneofs=[],
-    serialized_start=3158,
-    serialized_end=3316,
-)
-
-
-_STREAMINGRECOGNIZERESPONSE = _descriptor.Descriptor(
-    name="StreamingRecognizeResponse",
-    full_name="google.cloud.speech.v1.StreamingRecognizeResponse",
-    filename=None,
-    file=DESCRIPTOR,
-    containing_type=None,
-    fields=[
-        _descriptor.FieldDescriptor(
-            name="error",
-            full_name="google.cloud.speech.v1.StreamingRecognizeResponse.error",
-            index=0,
-            number=1,
-            type=11,
-            cpp_type=10,
-            label=1,
-            has_default_value=False,
-            default_value=None,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="results",
-            full_name="google.cloud.speech.v1.StreamingRecognizeResponse.results",
-            index=1,
-            number=2,
-            type=11,
-            cpp_type=10,
-            label=3,
-            has_default_value=False,
-            default_value=[],
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="speech_event_type",
-            full_name="google.cloud.speech.v1.StreamingRecognizeResponse.speech_event_type",
-            index=2,
-            number=4,
-            type=14,
-            cpp_type=8,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-    ],
-    extensions=[],
-    nested_types=[],
-    enum_types=[_STREAMINGRECOGNIZERESPONSE_SPEECHEVENTTYPE],
-    serialized_options=None,
-    is_extendable=False,
-    syntax="proto3",
-    extension_ranges=[],
-    oneofs=[],
-    serialized_start=3319,
-    serialized_end=3624,
-)
-
-
-_STREAMINGRECOGNITIONRESULT = _descriptor.Descriptor(
-    name="StreamingRecognitionResult",
-    full_name="google.cloud.speech.v1.StreamingRecognitionResult",
-    filename=None,
-    file=DESCRIPTOR,
-    containing_type=None,
-    fields=[
-        _descriptor.FieldDescriptor(
-            name="alternatives",
-            full_name="google.cloud.speech.v1.StreamingRecognitionResult.alternatives",
-            index=0,
-            number=1,
-            type=11,
-            cpp_type=10,
-            label=3,
-            has_default_value=False,
-            default_value=[],
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="is_final",
-            full_name="google.cloud.speech.v1.StreamingRecognitionResult.is_final",
-            index=1,
-            number=2,
-            type=8,
-            cpp_type=7,
-            label=1,
-            has_default_value=False,
-            default_value=False,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="stability",
-            full_name="google.cloud.speech.v1.StreamingRecognitionResult.stability",
-            index=2,
-            number=3,
-            type=2,
-            cpp_type=6,
-            label=1,
-            has_default_value=False,
-            default_value=float(0),
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="result_end_time",
-            full_name="google.cloud.speech.v1.StreamingRecognitionResult.result_end_time",
-            index=3,
-            number=4,
-            type=11,
-            cpp_type=10,
-            label=1,
-            has_default_value=False,
-            default_value=None,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="channel_tag",
-            full_name="google.cloud.speech.v1.StreamingRecognitionResult.channel_tag",
-            index=4,
-            number=5,
-            type=5,
-            cpp_type=1,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="language_code",
-            full_name="google.cloud.speech.v1.StreamingRecognitionResult.language_code",
-            index=5,
-            number=6,
-            type=9,
-            cpp_type=9,
-            label=1,
-            has_default_value=False,
-            default_value=_b("").decode("utf-8"),
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=_b("\340A\003"),
-            file=DESCRIPTOR,
-        ),
-    ],
-    extensions=[],
-    nested_types=[],
-    enum_types=[],
-    serialized_options=None,
-    is_extendable=False,
-    syntax="proto3",
-    extension_ranges=[],
-    oneofs=[],
-    serialized_start=3627,
-    serialized_end=3869,
-)
-
-
-_SPEECHRECOGNITIONRESULT = _descriptor.Descriptor(
-    name="SpeechRecognitionResult",
-    full_name="google.cloud.speech.v1.SpeechRecognitionResult",
-    filename=None,
-    file=DESCRIPTOR,
-    containing_type=None,
-    fields=[
-        _descriptor.FieldDescriptor(
-            name="alternatives",
-            full_name="google.cloud.speech.v1.SpeechRecognitionResult.alternatives",
-            index=0,
-            number=1,
-            type=11,
-            cpp_type=10,
-            label=3,
-            has_default_value=False,
-            default_value=[],
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="channel_tag",
-            full_name="google.cloud.speech.v1.SpeechRecognitionResult.channel_tag",
-            index=1,
-            number=2,
-            type=5,
-            cpp_type=1,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-    ],
-    extensions=[],
-    nested_types=[],
-    enum_types=[],
-    serialized_options=None,
-    is_extendable=False,
-    syntax="proto3",
-    extension_ranges=[],
-    oneofs=[],
-    serialized_start=3871,
-    serialized_end=3993,
-)
-
-
-_SPEECHRECOGNITIONALTERNATIVE = _descriptor.Descriptor(
-    name="SpeechRecognitionAlternative",
-    full_name="google.cloud.speech.v1.SpeechRecognitionAlternative",
-    filename=None,
-    file=DESCRIPTOR,
-    containing_type=None,
-    fields=[
-        _descriptor.FieldDescriptor(
-            name="transcript",
-            full_name="google.cloud.speech.v1.SpeechRecognitionAlternative.transcript",
-            index=0,
-            number=1,
-            type=9,
-            cpp_type=9,
-            label=1,
-            has_default_value=False,
-            default_value=_b("").decode("utf-8"),
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="confidence",
-            full_name="google.cloud.speech.v1.SpeechRecognitionAlternative.confidence",
-            index=1,
-            number=2,
-            type=2,
-            cpp_type=6,
-            label=1,
-            has_default_value=False,
-            default_value=float(0),
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="words",
-            full_name="google.cloud.speech.v1.SpeechRecognitionAlternative.words",
-            index=2,
-            number=3,
-            type=11,
-            cpp_type=10,
-            label=3,
-            has_default_value=False,
-            default_value=[],
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-    ],
-    extensions=[],
-    nested_types=[],
-    enum_types=[],
-    serialized_options=None,
-    is_extendable=False,
-    syntax="proto3",
-    extension_ranges=[],
-    oneofs=[],
-    serialized_start=3995,
-    serialized_end=4114,
-)
-
-
-_WORDINFO = _descriptor.Descriptor(
-    name="WordInfo",
-    full_name="google.cloud.speech.v1.WordInfo",
-    filename=None,
-    file=DESCRIPTOR,
-    containing_type=None,
-    fields=[
-        _descriptor.FieldDescriptor(
-            name="start_time",
-            full_name="google.cloud.speech.v1.WordInfo.start_time",
-            index=0,
-            number=1,
-            type=11,
-            cpp_type=10,
-            label=1,
-            has_default_value=False,
-            default_value=None,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="end_time",
-            full_name="google.cloud.speech.v1.WordInfo.end_time",
-            index=1,
-            number=2,
-            type=11,
-            cpp_type=10,
-            label=1,
-            has_default_value=False,
-            default_value=None,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="word",
-            full_name="google.cloud.speech.v1.WordInfo.word",
-            index=2,
-            number=3,
-            type=9,
-            cpp_type=9,
-            label=1,
-            has_default_value=False,
-            default_value=_b("").decode("utf-8"),
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="speaker_tag",
-            full_name="google.cloud.speech.v1.WordInfo.speaker_tag",
-            index=3,
-            number=5,
-            type=5,
-            cpp_type=1,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=_b("\340A\003"),
-            file=DESCRIPTOR,
-        ),
-    ],
-    extensions=[],
-    nested_types=[],
-    enum_types=[],
-    serialized_options=None,
-    is_extendable=False,
-    syntax="proto3",
-    extension_ranges=[],
-    oneofs=[],
-    serialized_start=4117,
-    serialized_end=4259,
-)
-
-_RECOGNIZEREQUEST.fields_by_name["config"].message_type = _RECOGNITIONCONFIG
-_RECOGNIZEREQUEST.fields_by_name["audio"].message_type = _RECOGNITIONAUDIO
-_LONGRUNNINGRECOGNIZEREQUEST.fields_by_name["config"].message_type = _RECOGNITIONCONFIG
-_LONGRUNNINGRECOGNIZEREQUEST.fields_by_name["audio"].message_type = _RECOGNITIONAUDIO
-_STREAMINGRECOGNIZEREQUEST.fields_by_name[
-    "streaming_config"
-].message_type = _STREAMINGRECOGNITIONCONFIG
-_STREAMINGRECOGNIZEREQUEST.oneofs_by_name["streaming_request"].fields.append(
-    _STREAMINGRECOGNIZEREQUEST.fields_by_name["streaming_config"]
-)
-_STREAMINGRECOGNIZEREQUEST.fields_by_name[
-    "streaming_config"
-].containing_oneof = _STREAMINGRECOGNIZEREQUEST.oneofs_by_name["streaming_request"]
-_STREAMINGRECOGNIZEREQUEST.oneofs_by_name["streaming_request"].fields.append(
-    _STREAMINGRECOGNIZEREQUEST.fields_by_name["audio_content"]
-)
-_STREAMINGRECOGNIZEREQUEST.fields_by_name[
-    "audio_content"
-].containing_oneof = _STREAMINGRECOGNIZEREQUEST.oneofs_by_name["streaming_request"]
-_STREAMINGRECOGNITIONCONFIG.fields_by_name["config"].message_type = _RECOGNITIONCONFIG
-_RECOGNITIONCONFIG.fields_by_name[
-    "encoding"
-].enum_type = _RECOGNITIONCONFIG_AUDIOENCODING
-_RECOGNITIONCONFIG.fields_by_name["speech_contexts"].message_type = _SPEECHCONTEXT
-_RECOGNITIONCONFIG.fields_by_name[
-    "diarization_config"
-].message_type = _SPEAKERDIARIZATIONCONFIG
-_RECOGNITIONCONFIG.fields_by_name["metadata"].message_type = _RECOGNITIONMETADATA
-_RECOGNITIONCONFIG_AUDIOENCODING.containing_type = _RECOGNITIONCONFIG
-_RECOGNITIONMETADATA.fields_by_name[
-    "interaction_type"
-].enum_type = _RECOGNITIONMETADATA_INTERACTIONTYPE
-_RECOGNITIONMETADATA.fields_by_name[
-    "microphone_distance"
-].enum_type = _RECOGNITIONMETADATA_MICROPHONEDISTANCE
-_RECOGNITIONMETADATA.fields_by_name[
-    "original_media_type"
-].enum_type = _RECOGNITIONMETADATA_ORIGINALMEDIATYPE
-_RECOGNITIONMETADATA.fields_by_name[
-    "recording_device_type"
-].enum_type = _RECOGNITIONMETADATA_RECORDINGDEVICETYPE
-_RECOGNITIONMETADATA_INTERACTIONTYPE.containing_type = _RECOGNITIONMETADATA
-_RECOGNITIONMETADATA_MICROPHONEDISTANCE.containing_type = _RECOGNITIONMETADATA
-_RECOGNITIONMETADATA_ORIGINALMEDIATYPE.containing_type = _RECOGNITIONMETADATA
-_RECOGNITIONMETADATA_RECORDINGDEVICETYPE.containing_type = _RECOGNITIONMETADATA
-_RECOGNITIONAUDIO.oneofs_by_name["audio_source"].fields.append(
-    _RECOGNITIONAUDIO.fields_by_name["content"]
-)
-_RECOGNITIONAUDIO.fields_by_name[
-    "content"
-].containing_oneof = _RECOGNITIONAUDIO.oneofs_by_name["audio_source"]
-_RECOGNITIONAUDIO.oneofs_by_name["audio_source"].fields.append(
-    _RECOGNITIONAUDIO.fields_by_name["uri"]
-)
-_RECOGNITIONAUDIO.fields_by_name[
-    "uri"
-].containing_oneof = _RECOGNITIONAUDIO.oneofs_by_name["audio_source"]
-_RECOGNIZERESPONSE.fields_by_name["results"].message_type = _SPEECHRECOGNITIONRESULT
-_LONGRUNNINGRECOGNIZERESPONSE.fields_by_name[
-    "results"
-].message_type = _SPEECHRECOGNITIONRESULT
-_LONGRUNNINGRECOGNIZEMETADATA.fields_by_name[
-    "start_time"
-].message_type = google_dot_protobuf_dot_timestamp__pb2._TIMESTAMP
-_LONGRUNNINGRECOGNIZEMETADATA.fields_by_name[
-    "last_update_time"
-].message_type = google_dot_protobuf_dot_timestamp__pb2._TIMESTAMP
-_STREAMINGRECOGNIZERESPONSE.fields_by_name[
-    "error"
-].message_type = google_dot_rpc_dot_status__pb2._STATUS
-_STREAMINGRECOGNIZERESPONSE.fields_by_name[
-    "results"
-].message_type = _STREAMINGRECOGNITIONRESULT
-_STREAMINGRECOGNIZERESPONSE.fields_by_name[
-    "speech_event_type"
-].enum_type = _STREAMINGRECOGNIZERESPONSE_SPEECHEVENTTYPE
-_STREAMINGRECOGNIZERESPONSE_SPEECHEVENTTYPE.containing_type = (
-    _STREAMINGRECOGNIZERESPONSE
-)
-_STREAMINGRECOGNITIONRESULT.fields_by_name[
-    "alternatives"
-].message_type = _SPEECHRECOGNITIONALTERNATIVE
-_STREAMINGRECOGNITIONRESULT.fields_by_name[
-    "result_end_time"
-].message_type = google_dot_protobuf_dot_duration__pb2._DURATION
-_SPEECHRECOGNITIONRESULT.fields_by_name[
-    "alternatives"
-].message_type = _SPEECHRECOGNITIONALTERNATIVE
-_SPEECHRECOGNITIONALTERNATIVE.fields_by_name["words"].message_type = _WORDINFO
-_WORDINFO.fields_by_name[
-    "start_time"
-].message_type = google_dot_protobuf_dot_duration__pb2._DURATION
-_WORDINFO.fields_by_name[
-    "end_time"
-].message_type = google_dot_protobuf_dot_duration__pb2._DURATION
-DESCRIPTOR.message_types_by_name["RecognizeRequest"] = _RECOGNIZEREQUEST
-DESCRIPTOR.message_types_by_name[
-    "LongRunningRecognizeRequest"
-] = _LONGRUNNINGRECOGNIZEREQUEST
-DESCRIPTOR.message_types_by_name[
-    "StreamingRecognizeRequest"
-] = _STREAMINGRECOGNIZEREQUEST
-DESCRIPTOR.message_types_by_name[
-    "StreamingRecognitionConfig"
-] = _STREAMINGRECOGNITIONCONFIG
-DESCRIPTOR.message_types_by_name["RecognitionConfig"] = _RECOGNITIONCONFIG
-DESCRIPTOR.message_types_by_name["SpeakerDiarizationConfig"] = _SPEAKERDIARIZATIONCONFIG
-DESCRIPTOR.message_types_by_name["RecognitionMetadata"] = _RECOGNITIONMETADATA
-DESCRIPTOR.message_types_by_name["SpeechContext"] = _SPEECHCONTEXT
-DESCRIPTOR.message_types_by_name["RecognitionAudio"] = _RECOGNITIONAUDIO
-DESCRIPTOR.message_types_by_name["RecognizeResponse"] = _RECOGNIZERESPONSE
-DESCRIPTOR.message_types_by_name[
-    "LongRunningRecognizeResponse"
-] = _LONGRUNNINGRECOGNIZERESPONSE
-DESCRIPTOR.message_types_by_name[
-    "LongRunningRecognizeMetadata"
-] = _LONGRUNNINGRECOGNIZEMETADATA
-DESCRIPTOR.message_types_by_name[
-    "StreamingRecognizeResponse"
-] = _STREAMINGRECOGNIZERESPONSE
-DESCRIPTOR.message_types_by_name[
-    "StreamingRecognitionResult"
-] = _STREAMINGRECOGNITIONRESULT
-DESCRIPTOR.message_types_by_name["SpeechRecognitionResult"] = _SPEECHRECOGNITIONRESULT
-DESCRIPTOR.message_types_by_name[
-    "SpeechRecognitionAlternative"
-] = _SPEECHRECOGNITIONALTERNATIVE
-DESCRIPTOR.message_types_by_name["WordInfo"] = _WORDINFO
-_sym_db.RegisterFileDescriptor(DESCRIPTOR)
-
-RecognizeRequest = _reflection.GeneratedProtocolMessageType(
-    "RecognizeRequest",
-    (_message.Message,),
-    dict(
-        DESCRIPTOR=_RECOGNIZEREQUEST,
-        __module__="google.cloud.speech_v1.proto.cloud_speech_pb2",
-        __doc__="""The top-level message sent by the client for the
-  ``Recognize`` method.
-  
-  
-  Attributes:
-      config:
-          Required. Provides information to the recognizer that
-          specifies how to process the request.
-      audio:
-          Required. The audio data to be recognized.
-  """,
-        # @@protoc_insertion_point(class_scope:google.cloud.speech.v1.RecognizeRequest)
-    ),
-)
-_sym_db.RegisterMessage(RecognizeRequest)
-
-LongRunningRecognizeRequest = _reflection.GeneratedProtocolMessageType(
-    "LongRunningRecognizeRequest",
-    (_message.Message,),
-    dict(
-        DESCRIPTOR=_LONGRUNNINGRECOGNIZEREQUEST,
-        __module__="google.cloud.speech_v1.proto.cloud_speech_pb2",
-        __doc__="""The top-level message sent by the client for the
-  ``LongRunningRecognize`` method.
-  
-  
-  Attributes:
-      config:
-          Required. Provides information to the recognizer that
-          specifies how to process the request.
-      audio:
-          Required. The audio data to be recognized.
-  """,
-        # @@protoc_insertion_point(class_scope:google.cloud.speech.v1.LongRunningRecognizeRequest)
-    ),
-)
-_sym_db.RegisterMessage(LongRunningRecognizeRequest)
-
-StreamingRecognizeRequest = _reflection.GeneratedProtocolMessageType(
-    "StreamingRecognizeRequest",
-    (_message.Message,),
-    dict(
-        DESCRIPTOR=_STREAMINGRECOGNIZEREQUEST,
-        __module__="google.cloud.speech_v1.proto.cloud_speech_pb2",
-        __doc__="""The top-level message sent by the client for the ``StreamingRecognize``
-  method. Multiple ``StreamingRecognizeRequest`` messages are sent. The
-  first message must contain a ``streaming_config`` message and must not
-  contain ``audio_content``. All subsequent messages must contain
-  ``audio_content`` and must not contain a ``streaming_config`` message.
-  
-  
-  Attributes:
-      streaming_request:
-          The streaming request, which is either a streaming config or
-          audio content.
-      streaming_config:
-          Provides information to the recognizer that specifies how to
-          process the request. The first ``StreamingRecognizeRequest``
-          message must contain a ``streaming_config`` message.
-      audio_content:
-          The audio data to be recognized. Sequential chunks of audio
-          data are sent in sequential ``StreamingRecognizeRequest``
-          messages. The first ``StreamingRecognizeRequest`` message must
-          not contain ``audio_content`` data and all subsequent
-          ``StreamingRecognizeRequest`` messages must contain
-          ``audio_content`` data. The audio bytes must be encoded as
-          specified in ``RecognitionConfig``. Note: as with all bytes
-          fields, proto buffers use a pure binary representation (not
-          base64). See `content limits <https://cloud.google.com/speech-
-          to-text/quotas#content>`__.
-  """,
-        # @@protoc_insertion_point(class_scope:google.cloud.speech.v1.StreamingRecognizeRequest)
-    ),
-)
-_sym_db.RegisterMessage(StreamingRecognizeRequest)
-
-StreamingRecognitionConfig = _reflection.GeneratedProtocolMessageType(
-    "StreamingRecognitionConfig",
-    (_message.Message,),
-    dict(
-        DESCRIPTOR=_STREAMINGRECOGNITIONCONFIG,
-        __module__="google.cloud.speech_v1.proto.cloud_speech_pb2",
-        __doc__="""Provides information to the recognizer that specifies how
-  to process the request.
-  
-  
-  Attributes:
-      config:
-          Required. Provides information to the recognizer that
-          specifies how to process the request.
-      single_utterance:
-          If ``false`` or omitted, the recognizer will perform
-          continuous recognition (continuing to wait for and process
-          audio even if the user pauses speaking) until the client
-          closes the input stream (gRPC API) or until the maximum time
-          limit has been reached. May return multiple
-          ``StreamingRecognitionResult``\ s with the ``is_final`` flag
-          set to ``true``.  If ``true``, the recognizer will detect a
-          single spoken utterance. When it detects that the user has
-          paused or stopped speaking, it will return an
-          ``END_OF_SINGLE_UTTERANCE`` event and cease recognition. It
-          will return no more than one ``StreamingRecognitionResult``
-          with the ``is_final`` flag set to ``true``.
-      interim_results:
-          If ``true``, interim results (tentative hypotheses) may be
-          returned as they become available (these interim results are
-          indicated with the ``is_final=false`` flag). If ``false`` or
-          omitted, only ``is_final=true`` result(s) are returned.
-  """,
-        # @@protoc_insertion_point(class_scope:google.cloud.speech.v1.StreamingRecognitionConfig)
-    ),
-)
-_sym_db.RegisterMessage(StreamingRecognitionConfig)
-
-RecognitionConfig = _reflection.GeneratedProtocolMessageType(
-    "RecognitionConfig",
-    (_message.Message,),
-    dict(
-        DESCRIPTOR=_RECOGNITIONCONFIG,
-        __module__="google.cloud.speech_v1.proto.cloud_speech_pb2",
-        __doc__="""Provides information to the recognizer that specifies how
-  to process the request.
-  
-  
-  Attributes:
-      encoding:
-          Encoding of audio data sent in all ``RecognitionAudio``
-          messages. This field is optional for ``FLAC`` and ``WAV``
-          audio files and required for all other audio formats. For
-          details, see [AudioEncoding][google.cloud.speech.v1.Recognitio
-          nConfig.AudioEncoding].
-      sample_rate_hertz:
-          Sample rate in Hertz of the audio data sent in all
-          ``RecognitionAudio`` messages. Valid values are: 8000-48000.
-          16000 is optimal. For best results, set the sampling rate of
-          the audio source to 16000 Hz. If that's not possible, use the
-          native sample rate of the audio source (instead of re-
-          sampling). This field is optional for FLAC and WAV audio
-          files, but is required for all other audio formats. For
-          details, see [AudioEncoding][google.cloud.speech.v1.Recognitio
-          nConfig.AudioEncoding].
-      audio_channel_count:
-          The number of channels in the input audio data. ONLY set this
-          for MULTI-CHANNEL recognition. Valid values for LINEAR16 and
-          FLAC are ``1``-``8``. Valid values for OGG\_OPUS are
-          '1'-'254'. Valid value for MULAW, AMR, AMR\_WB and
-          SPEEX\_WITH\_HEADER\_BYTE is only ``1``. If ``0`` or omitted,
-          defaults to one channel (mono). Note: We only recognize the
-          first channel by default. To perform independent recognition
-          on each channel set
-          ``enable_separate_recognition_per_channel`` to 'true'.
-      enable_separate_recognition_per_channel:
-          This needs to be set to ``true`` explicitly and
-          ``audio_channel_count`` > 1 to get each channel recognized
-          separately. The recognition result will contain a
-          ``channel_tag`` field to state which channel that result
-          belongs to. If this is not true, we will only recognize the
-          first channel. The request is billed cumulatively for all
-          channels recognized: ``audio_channel_count`` multiplied by the
-          length of the audio.
-      language_code:
-          Required. The language of the supplied audio as a `BCP-47
-          <https://www.rfc-editor.org/rfc/bcp/bcp47.txt>`__ language
-          tag. Example: "en-US". See `Language Support
-          <https://cloud.google.com/speech-to-text/docs/languages>`__
-          for a list of the currently supported language codes.
-      max_alternatives:
-          Maximum number of recognition hypotheses to be returned.
-          Specifically, the maximum number of
-          ``SpeechRecognitionAlternative`` messages within each
-          ``SpeechRecognitionResult``. The server may return fewer than
-          ``max_alternatives``. Valid values are ``0``-``30``. A value
-          of ``0`` or ``1`` will return a maximum of one. If omitted,
-          will return a maximum of one.
-      profanity_filter:
-          If set to ``true``, the server will attempt to filter out
-          profanities, replacing all but the initial character in each
-          filtered word with asterisks, e.g. "f\*\*\*". If set to
-          ``false`` or omitted, profanities won't be filtered out.
-      speech_contexts:
-          Array of
-          [SpeechContext][google.cloud.speech.v1.SpeechContext]. A means
-          to provide context to assist the speech recognition. For more
-          information, see `speech adaptation
-          <https://cloud.google.com/speech-to-text/docs/context-
-          strength>`__.
-      enable_word_time_offsets:
-          If ``true``, the top result includes a list of words and the
-          start and end time offsets (timestamps) for those words. If
-          ``false``, no word-level time offset information is returned.
-          The default is ``false``.
-      enable_automatic_punctuation:
-          If 'true', adds punctuation to recognition result hypotheses.
-          This feature is only available in select languages. Setting
-          this for requests in other languages has no effect at all. The
-          default 'false' value does not add punctuation to result
-          hypotheses. Note: This is currently offered as an experimental
-          service, complimentary to all users. In the future this may be
-          exclusively available as a premium feature.
-      diarization_config:
-          Config to enable speaker diarization and set additional
-          parameters to make diarization better suited for your
-          application. Note: When this is enabled, we send all the words
-          from the beginning of the audio for the top alternative in
-          every consecutive STREAMING responses. This is done in order
-          to improve our speaker tags as our models learn to identify
-          the speakers in the conversation over time. For non-streaming
-          requests, the diarization results will be provided only in the
-          top alternative of the FINAL SpeechRecognitionResult.
-      metadata:
-          Metadata regarding this request.
-      model:
-          Which model to select for the given request. Select the model
-          best suited to your domain to get best results. If a model is
-          not explicitly specified, then we auto-select a model based on
-          the parameters in the RecognitionConfig.  .. raw:: html
-          <table>  .. raw:: html     <tr>  ::      <td><b>Model</b></td>
-          <td><b>Description</b></td>  .. raw:: html     </tr>  .. raw::
-          html     <tr>  ::
-          <td><code>command_and_search</code></td>     <td>Best for
-          short queries such as voice commands or voice search.</td>  ..
-          raw:: html     </tr>  .. raw:: html     <tr>  ::
-          <td><code>phone_call</code></td>     <td>Best for audio that
-          originated from a phone call (typically     recorded at an
-          8khz sampling rate).</td>  .. raw:: html     </tr>  .. raw::
-          html     <tr>  ::      <td><code>video</code></td>
-          <td>Best for audio that originated from from video or includes
-          multiple         speakers. Ideally the audio is recorded at a
-          16khz or greater         sampling rate. This is a premium
-          model that costs more than the         standard rate.</td>  ..
-          raw:: html     </tr>  .. raw:: html     <tr>  ::
-          <td><code>default</code></td>     <td>Best for audio that is
-          not one of the specific audio models.         For example,
-          long-form audio. Ideally the audio is high-fidelity,
-          recorded at a 16khz or greater sampling rate.</td>  .. raw::
-          html     </tr>  .. raw:: html     </table>
-      use_enhanced:
-          Set to true to use an enhanced model for speech recognition.
-          If ``use_enhanced`` is set to true and the ``model`` field is
-          not set, then an appropriate enhanced model is chosen if an
-          enhanced model exists for the audio.  If ``use_enhanced`` is
-          true and an enhanced version of the specified model does not
-          exist, then the speech is recognized using the standard
-          version of the specified model.
-  """,
-        # @@protoc_insertion_point(class_scope:google.cloud.speech.v1.RecognitionConfig)
-    ),
-)
-_sym_db.RegisterMessage(RecognitionConfig)
-
-SpeakerDiarizationConfig = _reflection.GeneratedProtocolMessageType(
-    "SpeakerDiarizationConfig",
-    (_message.Message,),
-    dict(
-        DESCRIPTOR=_SPEAKERDIARIZATIONCONFIG,
-        __module__="google.cloud.speech_v1.proto.cloud_speech_pb2",
-        __doc__="""Config to enable speaker diarization.
-  
-  
-  Attributes:
-      enable_speaker_diarization:
-          If 'true', enables speaker detection for each recognized word
-          in the top alternative of the recognition result using a
-          speaker\_tag provided in the WordInfo.
-      min_speaker_count:
-          Minimum number of speakers in the conversation. This range
-          gives you more flexibility by allowing the system to
-          automatically determine the correct number of speakers. If not
-          set, the default value is 2.
-      max_speaker_count:
-          Maximum number of speakers in the conversation. This range
-          gives you more flexibility by allowing the system to
-          automatically determine the correct number of speakers. If not
-          set, the default value is 6.
-      speaker_tag:
-          Unused.
-  """,
-        # @@protoc_insertion_point(class_scope:google.cloud.speech.v1.SpeakerDiarizationConfig)
-    ),
-)
-_sym_db.RegisterMessage(SpeakerDiarizationConfig)
-
-RecognitionMetadata = _reflection.GeneratedProtocolMessageType(
-    "RecognitionMetadata",
-    (_message.Message,),
-    dict(
-        DESCRIPTOR=_RECOGNITIONMETADATA,
-        __module__="google.cloud.speech_v1.proto.cloud_speech_pb2",
-        __doc__="""Description of audio data to be recognized.
-  
-  
-  Attributes:
-      interaction_type:
-          The use case most closely describing the audio content to be
-          recognized.
-      industry_naics_code_of_audio:
-          The industry vertical to which this speech recognition request
-          most closely applies. This is most indicative of the topics
-          contained in the audio. Use the 6-digit NAICS code to identify
-          the industry vertical - see https://www.naics.com/search/.
-      microphone_distance:
-          The audio type that most closely describes the audio being
-          recognized.
-      original_media_type:
-          The original media the speech was recorded on.
-      recording_device_type:
-          The type of device the speech was recorded with.
-      recording_device_name:
-          The device used to make the recording. Examples 'Nexus 5X' or
-          'Polycom SoundStation IP 6000' or 'POTS' or 'VoIP' or
-          'Cardioid Microphone'.
-      original_mime_type:
-          Mime type of the original audio file. For example
-          ``audio/m4a``, ``audio/x-alaw-basic``, ``audio/mp3``,
-          ``audio/3gpp``. A list of possible audio mime types is
-          maintained at http://www.iana.org/assignments/media-
-          types/media-types.xhtml#audio
-      audio_topic:
-          Description of the content. Eg. "Recordings of federal supreme
-          court hearings from 2012".
-  """,
-        # @@protoc_insertion_point(class_scope:google.cloud.speech.v1.RecognitionMetadata)
-    ),
-)
-_sym_db.RegisterMessage(RecognitionMetadata)
-
-SpeechContext = _reflection.GeneratedProtocolMessageType(
-    "SpeechContext",
-    (_message.Message,),
-    dict(
-        DESCRIPTOR=_SPEECHCONTEXT,
-        __module__="google.cloud.speech_v1.proto.cloud_speech_pb2",
-        __doc__="""Provides "hints" to the speech recognizer to favor
-  specific words and phrases in the results.
-  
-  
-  Attributes:
-      phrases:
-          A list of strings containing words and phrases "hints" so that
-          the speech recognition is more likely to recognize them. This
-          can be used to improve the accuracy for specific words and
-          phrases, for example, if specific commands are typically
-          spoken by the user. This can also be used to add additional
-          words to the vocabulary of the recognizer. See `usage limits
-          <https://cloud.google.com/speech-to-text/quotas#content>`__.
-          List items can also be set to classes for groups of words that
-          represent common concepts that occur in natural language. For
-          example, rather than providing phrase hints for every month of
-          the year, using the $MONTH class improves the likelihood of
-          correctly transcribing audio that includes months.
-  """,
-        # @@protoc_insertion_point(class_scope:google.cloud.speech.v1.SpeechContext)
-    ),
-)
-_sym_db.RegisterMessage(SpeechContext)
-
-RecognitionAudio = _reflection.GeneratedProtocolMessageType(
-    "RecognitionAudio",
-    (_message.Message,),
-    dict(
-        DESCRIPTOR=_RECOGNITIONAUDIO,
-        __module__="google.cloud.speech_v1.proto.cloud_speech_pb2",
-        __doc__="""Contains audio data in the encoding specified in the
-  ``RecognitionConfig``. Either ``content`` or ``uri`` must be supplied.
-  Supplying both or neither returns
-  [google.rpc.Code.INVALID\_ARGUMENT][google.rpc.Code.INVALID\_ARGUMENT].
-  See `content
-  limits <https://cloud.google.com/speech-to-text/quotas#content>`__.
-  
-  
-  Attributes:
-      audio_source:
-          The audio source, which is either inline content or a Google
-          Cloud Storage uri.
-      content:
-          The audio data bytes encoded as specified in
-          ``RecognitionConfig``. Note: as with all bytes fields, proto
-          buffers use a pure binary representation, whereas JSON
-          representations use base64.
-      uri:
-          URI that points to a file that contains audio data bytes as
-          specified in ``RecognitionConfig``. The file must not be
-          compressed (for example, gzip). Currently, only Google Cloud
-          Storage URIs are supported, which must be specified in the
-          following format: ``gs://bucket_name/object_name`` (other URI
-          formats return [google.rpc.Code.INVALID\_ARGUMENT][google.rpc.
-          Code.INVALID\_ARGUMENT]). For more information, see `Request
-          URIs <https://cloud.google.com/storage/docs/reference-
-          uris>`__.
-  """,
-        # @@protoc_insertion_point(class_scope:google.cloud.speech.v1.RecognitionAudio)
-    ),
-)
-_sym_db.RegisterMessage(RecognitionAudio)
-
-RecognizeResponse = _reflection.GeneratedProtocolMessageType(
-    "RecognizeResponse",
-    (_message.Message,),
-    dict(
-        DESCRIPTOR=_RECOGNIZERESPONSE,
-        __module__="google.cloud.speech_v1.proto.cloud_speech_pb2",
-        __doc__="""The only message returned to the client by the
-  ``Recognize`` method. It contains the result as zero or more sequential
-  ``SpeechRecognitionResult`` messages.
-  
-  
-  Attributes:
-      results:
-          Sequential list of transcription results corresponding to
-          sequential portions of audio.
-  """,
-        # @@protoc_insertion_point(class_scope:google.cloud.speech.v1.RecognizeResponse)
-    ),
-)
-_sym_db.RegisterMessage(RecognizeResponse)
-
-LongRunningRecognizeResponse = _reflection.GeneratedProtocolMessageType(
-    "LongRunningRecognizeResponse",
-    (_message.Message,),
-    dict(
-        DESCRIPTOR=_LONGRUNNINGRECOGNIZERESPONSE,
-        __module__="google.cloud.speech_v1.proto.cloud_speech_pb2",
-        __doc__="""The only message returned to the client by the
-  ``LongRunningRecognize`` method. It contains the result as zero or more
-  sequential ``SpeechRecognitionResult`` messages. It is included in the
-  ``result.response`` field of the ``Operation`` returned by the
-  ``GetOperation`` call of the ``google::longrunning::Operations``
-  service.
-  
-  
-  Attributes:
-      results:
-          Sequential list of transcription results corresponding to
-          sequential portions of audio.
-  """,
-        # @@protoc_insertion_point(class_scope:google.cloud.speech.v1.LongRunningRecognizeResponse)
-    ),
-)
-_sym_db.RegisterMessage(LongRunningRecognizeResponse)
-
-LongRunningRecognizeMetadata = _reflection.GeneratedProtocolMessageType(
-    "LongRunningRecognizeMetadata",
-    (_message.Message,),
-    dict(
-        DESCRIPTOR=_LONGRUNNINGRECOGNIZEMETADATA,
-        __module__="google.cloud.speech_v1.proto.cloud_speech_pb2",
-        __doc__="""Describes the progress of a long-running
-  ``LongRunningRecognize`` call. It is included in the ``metadata`` field
-  of the ``Operation`` returned by the ``GetOperation`` call of the
-  ``google::longrunning::Operations`` service.
-  
-  
-  Attributes:
-      progress_percent:
-          Approximate percentage of audio processed thus far. Guaranteed
-          to be 100 when the audio is fully processed and the results
-          are available.
-      start_time:
-          Time when the request was received.
-      last_update_time:
-          Time of the most recent processing update.
-  """,
-        # @@protoc_insertion_point(class_scope:google.cloud.speech.v1.LongRunningRecognizeMetadata)
-    ),
-)
-_sym_db.RegisterMessage(LongRunningRecognizeMetadata)
-
-StreamingRecognizeResponse = _reflection.GeneratedProtocolMessageType(
-    "StreamingRecognizeResponse",
-    (_message.Message,),
-    dict(
-        DESCRIPTOR=_STREAMINGRECOGNIZERESPONSE,
-        __module__="google.cloud.speech_v1.proto.cloud_speech_pb2",
-        __doc__="""\ ``StreamingRecognizeResponse`` is the only message
-  returned to the client by ``StreamingRecognize``. A series of zero or
-  more ``StreamingRecognizeResponse`` messages are streamed back to the
-  client. If there is no recognizable audio, and ``single_utterance`` is
-  set to false, then no messages are streamed back to the client.
-  
-  Here's an example of a series of ten ``StreamingRecognizeResponse``\ s
-  that might be returned while processing audio:
-  
-  1. results { alternatives { transcript: "tube" } stability: 0.01 }
-  
-  2. results { alternatives { transcript: "to be a" } stability: 0.01 }
-  
-  3. results { alternatives { transcript: "to be" } stability: 0.9 }
-     results { alternatives { transcript: " or not to be" } stability:
-     0.01 }
-  
-  4. results { alternatives { transcript: "to be or not to be" confidence:
-     0.92 } alternatives { transcript: "to bee or not to bee" } is\_final:
-     true }
-  
-  5. results { alternatives { transcript: " that's" } stability: 0.01 }
-  
-  6. results { alternatives { transcript: " that is" } stability: 0.9 }
-     results { alternatives { transcript: " the question" } stability:
-     0.01 }
-  
-  7. results { alternatives { transcript: " that is the question"
-     confidence: 0.98 } alternatives { transcript: " that was the
-     question" } is\_final: true }
-  
-  Notes:
-  
-  -  Only two of the above responses #4 and #7 contain final results; they
-     are indicated by ``is_final: true``. Concatenating these together
-     generates the full transcript: "to be or not to be that is the
-     question".
-  
-  -  The others contain interim ``results``. #3 and #6 contain two interim
-     ``results``: the first portion has a high stability and is less
-     likely to change; the second portion has a low stability and is very
-     likely to change. A UI designer might choose to show only high
-     stability ``results``.
-  
-  -  The specific ``stability`` and ``confidence`` values shown above are
-     only for illustrative purposes. Actual values may vary.
-  
-  -  In each response, only one of these fields will be set: ``error``,
-     ``speech_event_type``, or one or more (repeated) ``results``.
-  
-  
-  Attributes:
-      error:
-          If set, returns a [google.rpc.Status][google.rpc.Status]
-          message that specifies the error for the operation.
-      results:
-          This repeated list contains zero or more results that
-          correspond to consecutive portions of the audio currently
-          being processed. It contains zero or one ``is_final=true``
-          result (the newly settled portion), followed by zero or more
-          ``is_final=false`` results (the interim results).
-      speech_event_type:
-          Indicates the type of speech event.
-  """,
-        # @@protoc_insertion_point(class_scope:google.cloud.speech.v1.StreamingRecognizeResponse)
-    ),
-)
-_sym_db.RegisterMessage(StreamingRecognizeResponse)
-
-StreamingRecognitionResult = _reflection.GeneratedProtocolMessageType(
-    "StreamingRecognitionResult",
-    (_message.Message,),
-    dict(
-        DESCRIPTOR=_STREAMINGRECOGNITIONRESULT,
-        __module__="google.cloud.speech_v1.proto.cloud_speech_pb2",
-        __doc__="""A streaming speech recognition result corresponding to a
-  portion of the audio that is currently being processed.
-  
-  
-  Attributes:
-      alternatives:
-          May contain one or more recognition hypotheses (up to the
-          maximum specified in ``max_alternatives``). These alternatives
-          are ordered in terms of accuracy, with the top (first)
-          alternative being the most probable, as ranked by the
-          recognizer.
-      is_final:
-          If ``false``, this ``StreamingRecognitionResult`` represents
-          an interim result that may change. If ``true``, this is the
-          final time the speech service will return this particular
-          ``StreamingRecognitionResult``, the recognizer will not return
-          any further hypotheses for this portion of the transcript and
-          corresponding audio.
-      stability:
-          An estimate of the likelihood that the recognizer will not
-          change its guess about this interim result. Values range from
-          0.0 (completely unstable) to 1.0 (completely stable). This
-          field is only provided for interim results
-          (``is_final=false``). The default of 0.0 is a sentinel value
-          indicating ``stability`` was not set.
-      result_end_time:
-          Time offset of the end of this result relative to the
-          beginning of the audio.
-      channel_tag:
-          For multi-channel audio, this is the channel number
-          corresponding to the recognized result for the audio from that
-          channel. For audio\_channel\_count = N, its output values can
-          range from '1' to 'N'.
-      language_code:
-          The `BCP-47 <https://www.rfc-editor.org/rfc/bcp/bcp47.txt>`__
-          language tag of the language in this result. This language
-          code was detected to have the most likelihood of being spoken
-          in the audio.
-  """,
-        # @@protoc_insertion_point(class_scope:google.cloud.speech.v1.StreamingRecognitionResult)
-    ),
-)
-_sym_db.RegisterMessage(StreamingRecognitionResult)
-
-SpeechRecognitionResult = _reflection.GeneratedProtocolMessageType(
-    "SpeechRecognitionResult",
-    (_message.Message,),
-    dict(
-        DESCRIPTOR=_SPEECHRECOGNITIONRESULT,
-        __module__="google.cloud.speech_v1.proto.cloud_speech_pb2",
-        __doc__="""A speech recognition result corresponding to a portion of
-  the audio.
-  
-  
-  Attributes:
-      alternatives:
-          May contain one or more recognition hypotheses (up to the
-          maximum specified in ``max_alternatives``). These alternatives
-          are ordered in terms of accuracy, with the top (first)
-          alternative being the most probable, as ranked by the
-          recognizer.
-      channel_tag:
-          For multi-channel audio, this is the channel number
-          corresponding to the recognized result for the audio from that
-          channel. For audio\_channel\_count = N, its output values can
-          range from '1' to 'N'.
-  """,
-        # @@protoc_insertion_point(class_scope:google.cloud.speech.v1.SpeechRecognitionResult)
-    ),
-)
-_sym_db.RegisterMessage(SpeechRecognitionResult)
-
-SpeechRecognitionAlternative = _reflection.GeneratedProtocolMessageType(
-    "SpeechRecognitionAlternative",
-    (_message.Message,),
-    dict(
-        DESCRIPTOR=_SPEECHRECOGNITIONALTERNATIVE,
-        __module__="google.cloud.speech_v1.proto.cloud_speech_pb2",
-        __doc__="""Alternative hypotheses (a.k.a. n-best list).
-  
-  
-  Attributes:
-      transcript:
-          Transcript text representing the words that the user spoke.
-      confidence:
-          The confidence estimate between 0.0 and 1.0. A higher number
-          indicates an estimated greater likelihood that the recognized
-          words are correct. This field is set only for the top
-          alternative of a non-streaming result or, of a streaming
-          result where ``is_final=true``. This field is not guaranteed
-          to be accurate and users should not rely on it to be always
-          provided. The default of 0.0 is a sentinel value indicating
-          ``confidence`` was not set.
-      words:
-          A list of word-specific information for each recognized word.
-          Note: When ``enable_speaker_diarization`` is true, you will
-          see all the words from the beginning of the audio.
-  """,
-        # @@protoc_insertion_point(class_scope:google.cloud.speech.v1.SpeechRecognitionAlternative)
-    ),
-)
-_sym_db.RegisterMessage(SpeechRecognitionAlternative)
-
-WordInfo = _reflection.GeneratedProtocolMessageType(
-    "WordInfo",
-    (_message.Message,),
-    dict(
-        DESCRIPTOR=_WORDINFO,
-        __module__="google.cloud.speech_v1.proto.cloud_speech_pb2",
-        __doc__="""Word-specific information for recognized words.
-  
-  
-  Attributes:
-      start_time:
-          Time offset relative to the beginning of the audio, and
-          corresponding to the start of the spoken word. This field is
-          only set if ``enable_word_time_offsets=true`` and only in the
-          top hypothesis. This is an experimental feature and the
-          accuracy of the time offset can vary.
-      end_time:
-          Time offset relative to the beginning of the audio, and
-          corresponding to the end of the spoken word. This field is
-          only set if ``enable_word_time_offsets=true`` and only in the
-          top hypothesis. This is an experimental feature and the
-          accuracy of the time offset can vary.
-      word:
-          The word corresponding to this set of information.
-      speaker_tag:
-          A distinct integer value is assigned for every speaker within
-          the audio. This field specifies which one of those speakers
-          was detected to have spoken this word. Value ranges from '1'
-          to diarization\_speaker\_count. speaker\_tag is set if
-          enable\_speaker\_diarization = 'true' and only in the top
-          alternative.
-  """,
-        # @@protoc_insertion_point(class_scope:google.cloud.speech.v1.WordInfo)
-    ),
-)
-_sym_db.RegisterMessage(WordInfo)
-
-
-DESCRIPTOR._options = None
-_RECOGNIZEREQUEST.fields_by_name["config"]._options = None
-_RECOGNIZEREQUEST.fields_by_name["audio"]._options = None
-_LONGRUNNINGRECOGNIZEREQUEST.fields_by_name["config"]._options = None
-_LONGRUNNINGRECOGNIZEREQUEST.fields_by_name["audio"]._options = None
-_STREAMINGRECOGNITIONCONFIG.fields_by_name["config"]._options = None
-_RECOGNITIONCONFIG.fields_by_name["language_code"]._options = None
-_SPEAKERDIARIZATIONCONFIG.fields_by_name["speaker_tag"]._options = None
-_STREAMINGRECOGNITIONRESULT.fields_by_name["language_code"]._options = None
-_WORDINFO.fields_by_name["speaker_tag"]._options = None
-
-_SPEECH = _descriptor.ServiceDescriptor(
-    name="Speech",
-    full_name="google.cloud.speech.v1.Speech",
-    file=DESCRIPTOR,
-    index=0,
-    serialized_options=_b(
-        "\312A\025speech.googleapis.com\322A.https://www.googleapis.com/auth/cloud-platform"
-    ),
-    serialized_start=4262,
-    serialized_end=4855,
-    methods=[
-        _descriptor.MethodDescriptor(
-            name="Recognize",
-            full_name="google.cloud.speech.v1.Speech.Recognize",
-            index=0,
-            containing_service=None,
-            input_type=_RECOGNIZEREQUEST,
-            output_type=_RECOGNIZERESPONSE,
-            serialized_options=_b(
-                '\202\323\344\223\002\031"\024/v1/speech:recognize:\001*\332A\014config,audio'
-            ),
-        ),
-        _descriptor.MethodDescriptor(
-            name="LongRunningRecognize",
-            full_name="google.cloud.speech.v1.Speech.LongRunningRecognize",
-            index=1,
-            containing_service=None,
-            input_type=_LONGRUNNINGRECOGNIZEREQUEST,
-            output_type=google_dot_longrunning_dot_operations__pb2._OPERATION,
-            serialized_options=_b(
-                '\202\323\344\223\002$"\037/v1/speech:longrunningrecognize:\001*\332A\014config,audio\312A<\n\034LongRunningRecognizeResponse\022\034LongRunningRecognizeMetadata'
-            ),
-        ),
-        _descriptor.MethodDescriptor(
-            name="StreamingRecognize",
-            full_name="google.cloud.speech.v1.Speech.StreamingRecognize",
-            index=2,
-            containing_service=None,
-            input_type=_STREAMINGRECOGNIZEREQUEST,
-            output_type=_STREAMINGRECOGNIZERESPONSE,
-            serialized_options=None,
-        ),
-    ],
-)
-_sym_db.RegisterServiceDescriptor(_SPEECH)
-
-DESCRIPTOR.services_by_name["Speech"] = _SPEECH
-
-# @@protoc_insertion_point(module_scope)
diff --git a/google/cloud/speech_v1/proto/cloud_speech_pb2_grpc.py b/google/cloud/speech_v1/proto/cloud_speech_pb2_grpc.py
deleted file mode 100644
index a7ddcb9c..00000000
--- a/google/cloud/speech_v1/proto/cloud_speech_pb2_grpc.py
+++ /dev/null
@@ -1,93 +0,0 @@
-# Generated by the gRPC Python protocol compiler plugin. DO NOT EDIT!
-import grpc
-
-from google.cloud.speech_v1.proto import (
-    cloud_speech_pb2 as google_dot_cloud_dot_speech__v1_dot_proto_dot_cloud__speech__pb2,
-)
-from google.longrunning import (
-    operations_pb2 as google_dot_longrunning_dot_operations__pb2,
-)
-
-
-class SpeechStub(object):
-    """Service that implements Google Cloud Speech API.
-  """
-
-    def __init__(self, channel):
-        """Constructor.
-
-    Args:
-      channel: A grpc.Channel.
-    """
-        self.Recognize = channel.unary_unary(
-            "/google.cloud.speech.v1.Speech/Recognize",
-            request_serializer=google_dot_cloud_dot_speech__v1_dot_proto_dot_cloud__speech__pb2.RecognizeRequest.SerializeToString,
-            response_deserializer=google_dot_cloud_dot_speech__v1_dot_proto_dot_cloud__speech__pb2.RecognizeResponse.FromString,
-        )
-        self.LongRunningRecognize = channel.unary_unary(
-            "/google.cloud.speech.v1.Speech/LongRunningRecognize",
-            request_serializer=google_dot_cloud_dot_speech__v1_dot_proto_dot_cloud__speech__pb2.LongRunningRecognizeRequest.SerializeToString,
-            response_deserializer=google_dot_longrunning_dot_operations__pb2.Operation.FromString,
-        )
-        self.StreamingRecognize = channel.stream_stream(
-            "/google.cloud.speech.v1.Speech/StreamingRecognize",
-            request_serializer=google_dot_cloud_dot_speech__v1_dot_proto_dot_cloud__speech__pb2.StreamingRecognizeRequest.SerializeToString,
-            response_deserializer=google_dot_cloud_dot_speech__v1_dot_proto_dot_cloud__speech__pb2.StreamingRecognizeResponse.FromString,
-        )
-
-
-class SpeechServicer(object):
-    """Service that implements Google Cloud Speech API.
-  """
-
-    def Recognize(self, request, context):
-        """Performs synchronous speech recognition: receive results after all audio
-    has been sent and processed.
-    """
-        context.set_code(grpc.StatusCode.UNIMPLEMENTED)
-        context.set_details("Method not implemented!")
-        raise NotImplementedError("Method not implemented!")
-
-    def LongRunningRecognize(self, request, context):
-        """Performs asynchronous speech recognition: receive results via the
-    google.longrunning.Operations interface. Returns either an
-    `Operation.error` or an `Operation.response` which contains
-    a `LongRunningRecognizeResponse` message.
-    For more information on asynchronous speech recognition, see the
-    [how-to](https://cloud.google.com/speech-to-text/docs/async-recognize).
-    """
-        context.set_code(grpc.StatusCode.UNIMPLEMENTED)
-        context.set_details("Method not implemented!")
-        raise NotImplementedError("Method not implemented!")
-
-    def StreamingRecognize(self, request_iterator, context):
-        """Performs bidirectional streaming speech recognition: receive results while
-    sending audio. This method is only available via the gRPC API (not REST).
-    """
-        context.set_code(grpc.StatusCode.UNIMPLEMENTED)
-        context.set_details("Method not implemented!")
-        raise NotImplementedError("Method not implemented!")
-
-
-def add_SpeechServicer_to_server(servicer, server):
-    rpc_method_handlers = {
-        "Recognize": grpc.unary_unary_rpc_method_handler(
-            servicer.Recognize,
-            request_deserializer=google_dot_cloud_dot_speech__v1_dot_proto_dot_cloud__speech__pb2.RecognizeRequest.FromString,
-            response_serializer=google_dot_cloud_dot_speech__v1_dot_proto_dot_cloud__speech__pb2.RecognizeResponse.SerializeToString,
-        ),
-        "LongRunningRecognize": grpc.unary_unary_rpc_method_handler(
-            servicer.LongRunningRecognize,
-            request_deserializer=google_dot_cloud_dot_speech__v1_dot_proto_dot_cloud__speech__pb2.LongRunningRecognizeRequest.FromString,
-            response_serializer=google_dot_longrunning_dot_operations__pb2.Operation.SerializeToString,
-        ),
-        "StreamingRecognize": grpc.stream_stream_rpc_method_handler(
-            servicer.StreamingRecognize,
-            request_deserializer=google_dot_cloud_dot_speech__v1_dot_proto_dot_cloud__speech__pb2.StreamingRecognizeRequest.FromString,
-            response_serializer=google_dot_cloud_dot_speech__v1_dot_proto_dot_cloud__speech__pb2.StreamingRecognizeResponse.SerializeToString,
-        ),
-    }
-    generic_handler = grpc.method_handlers_generic_handler(
-        "google.cloud.speech.v1.Speech", rpc_method_handlers
-    )
-    server.add_generic_rpc_handlers((generic_handler,))
diff --git a/google/cloud/speech_v1/types.py b/google/cloud/speech_v1/types.py
deleted file mode 100644
index f2bd717a..00000000
--- a/google/cloud/speech_v1/types.py
+++ /dev/null
@@ -1,48 +0,0 @@
-# -*- coding: utf-8 -*-
-#
-# Copyright 2020 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-
-from __future__ import absolute_import
-import sys
-
-from google.api_core.protobuf_helpers import get_messages
-
-from google.cloud.speech_v1.proto import cloud_speech_pb2
-from google.longrunning import operations_pb2
-from google.protobuf import any_pb2
-from google.protobuf import duration_pb2
-from google.protobuf import timestamp_pb2
-from google.rpc import status_pb2
-
-
-_shared_modules = [operations_pb2, any_pb2, duration_pb2, timestamp_pb2, status_pb2]
-
-_local_modules = [cloud_speech_pb2]
-
-names = []
-
-for module in _shared_modules:  # pragma: NO COVER
-    for name, message in get_messages(module).items():
-        setattr(sys.modules[__name__], name, message)
-        names.append(name)
-for module in _local_modules:
-    for name, message in get_messages(module).items():
-        message.__module__ = "google.cloud.speech_v1.types"
-        setattr(sys.modules[__name__], name, message)
-        names.append(name)
-
-
-__all__ = tuple(sorted(names))
diff --git a/google/cloud/speech_v1p1beta1/__init__.py b/google/cloud/speech_v1p1beta1/__init__.py
deleted file mode 100644
index b68a3953..00000000
--- a/google/cloud/speech_v1p1beta1/__init__.py
+++ /dev/null
@@ -1,30 +0,0 @@
-# Copyright 2018 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-from __future__ import absolute_import
-
-from google.cloud.speech_v1p1beta1 import types
-from google.cloud.speech_v1p1beta1.gapic import enums
-from google.cloud.speech_v1p1beta1.gapic import speech_client
-
-from google.cloud.speech_v1.helpers import SpeechHelpers
-
-
-class SpeechClient(SpeechHelpers, speech_client.SpeechClient):
-    __doc__ = speech_client.SpeechClient.__doc__
-    enums = enums
-    types = types
-
-
-__all__ = ("enums", "types", "SpeechClient")
diff --git a/google/cloud/speech_v1p1beta1/gapic/__init__.py b/google/cloud/speech_v1p1beta1/gapic/__init__.py
deleted file mode 100644
index e69de29b..00000000
diff --git a/google/cloud/speech_v1p1beta1/gapic/enums.py b/google/cloud/speech_v1p1beta1/gapic/enums.py
deleted file mode 100644
index 3616356f..00000000
--- a/google/cloud/speech_v1p1beta1/gapic/enums.py
+++ /dev/null
@@ -1,200 +0,0 @@
-# -*- coding: utf-8 -*-
-#
-# Copyright 2020 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-"""Wrappers for protocol buffer enum types."""
-
-import enum
-
-
-class RecognitionConfig(object):
-    class AudioEncoding(enum.IntEnum):
-        """
-        The encoding of the audio data sent in the request.
-
-        All encodings support only 1 channel (mono) audio, unless the
-        ``audio_channel_count`` and ``enable_separate_recognition_per_channel``
-        fields are set.
-
-        For best results, the audio source should be captured and transmitted
-        using a lossless encoding (``FLAC`` or ``LINEAR16``). The accuracy of
-        the speech recognition can be reduced if lossy codecs are used to
-        capture or transmit audio, particularly if background noise is present.
-        Lossy codecs include ``MULAW``, ``AMR``, ``AMR_WB``, ``OGG_OPUS``,
-        ``SPEEX_WITH_HEADER_BYTE``, and ``MP3``.
-
-        The ``FLAC`` and ``WAV`` audio file formats include a header that
-        describes the included audio content. You can request recognition for
-        ``WAV`` files that contain either ``LINEAR16`` or ``MULAW`` encoded
-        audio. If you send ``FLAC`` or ``WAV`` audio file format in your
-        request, you do not need to specify an ``AudioEncoding``; the audio
-        encoding format is determined from the file header. If you specify an
-        ``AudioEncoding`` when you send send ``FLAC`` or ``WAV`` audio, the
-        encoding configuration must match the encoding described in the audio
-        header; otherwise the request returns an
-        ``google.rpc.Code.INVALID_ARGUMENT`` error code.
-
-        Attributes:
-          ENCODING_UNSPECIFIED (int): Not specified.
-          LINEAR16 (int): Uncompressed 16-bit signed little-endian samples (Linear PCM).
-          FLAC (int): ``FLAC`` (Free Lossless Audio Codec) is the recommended encoding because
-          it is lossless--therefore recognition is not compromised--and requires
-          only about half the bandwidth of ``LINEAR16``. ``FLAC`` stream encoding
-          supports 16-bit and 24-bit samples, however, not all fields in
-          ``STREAMINFO`` are supported.
-          MULAW (int): 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
-          AMR (int): Adaptive Multi-Rate Narrowband codec. ``sample_rate_hertz`` must be
-          8000.
-          AMR_WB (int): Adaptive Multi-Rate Wideband codec. ``sample_rate_hertz`` must be 16000.
-          OGG_OPUS (int): Opus encoded audio frames in Ogg container
-          (`OggOpus <https://wiki.xiph.org/OggOpus>`__). ``sample_rate_hertz``
-          must be one of 8000, 12000, 16000, 24000, or 48000.
-          SPEEX_WITH_HEADER_BYTE (int): Although the use of lossy encodings is not recommended, if a very low
-          bitrate encoding is required, ``OGG_OPUS`` is highly preferred over
-          Speex encoding. The `Speex <https://speex.org/>`__ encoding supported by
-          Cloud Speech API has a header byte in each block, as in MIME type
-          ``audio/x-speex-with-header-byte``. It is a variant of the RTP Speex
-          encoding defined in `RFC 5574 <https://tools.ietf.org/html/rfc5574>`__.
-          The stream is a sequence of blocks, one block per RTP packet. Each block
-          starts with a byte containing the length of the block, in bytes,
-          followed by one or more frames of Speex data, padded to an integral
-          number of bytes (octets) as specified in RFC 5574. In other words, each
-          RTP header is replaced with a single byte containing the block length.
-          Only Speex wideband is supported. ``sample_rate_hertz`` must be 16000.
-          MP3 (int): MP3 audio. Support all standard MP3 bitrates (which range from 32-320
-          kbps). When using this encoding, ``sample_rate_hertz`` can be optionally
-          unset if not known.
-        """
-
-        ENCODING_UNSPECIFIED = 0
-        LINEAR16 = 1
-        FLAC = 2
-        MULAW = 3
-        AMR = 4
-        AMR_WB = 5
-        OGG_OPUS = 6
-        SPEEX_WITH_HEADER_BYTE = 7
-        MP3 = 8
-
-
-class RecognitionMetadata(object):
-    class InteractionType(enum.IntEnum):
-        """
-        Use case categories that the audio recognition request can be described
-        by.
-
-        Attributes:
-          INTERACTION_TYPE_UNSPECIFIED (int): Use case is either unknown or is something other than one of the other
-          values below.
-          DISCUSSION (int): Multiple people in a conversation or discussion. For example in a
-          meeting with two or more people actively participating. Typically all
-          the primary people speaking would be in the same room (if not, see
-          PHONE\_CALL)
-          PRESENTATION (int): One or more persons lecturing or presenting to others, mostly
-          uninterrupted.
-          PHONE_CALL (int): A phone-call or video-conference in which two or more people, who are
-          not in the same room, are actively participating.
-          VOICEMAIL (int): A recorded message intended for another person to listen to.
-          PROFESSIONALLY_PRODUCED (int): Professionally produced audio (eg. TV Show, Podcast).
-          VOICE_SEARCH (int): Transcribe spoken questions and queries into text.
-          VOICE_COMMAND (int): Transcribe voice commands, such as for controlling a device.
-          DICTATION (int): Transcribe speech to text to create a written document, such as a
-          text-message, email or report.
-        """
-
-        INTERACTION_TYPE_UNSPECIFIED = 0
-        DISCUSSION = 1
-        PRESENTATION = 2
-        PHONE_CALL = 3
-        VOICEMAIL = 4
-        PROFESSIONALLY_PRODUCED = 5
-        VOICE_SEARCH = 6
-        VOICE_COMMAND = 7
-        DICTATION = 8
-
-    class MicrophoneDistance(enum.IntEnum):
-        """
-        Enumerates the types of capture settings describing an audio file.
-
-        Attributes:
-          MICROPHONE_DISTANCE_UNSPECIFIED (int): Audio type is not known.
-          NEARFIELD (int): The audio was captured from a closely placed microphone. Eg. phone,
-          dictaphone, or handheld microphone. Generally if there speaker is within
-          1 meter of the microphone.
-          MIDFIELD (int): The speaker if within 3 meters of the microphone.
-          FARFIELD (int): The speaker is more than 3 meters away from the microphone.
-        """
-
-        MICROPHONE_DISTANCE_UNSPECIFIED = 0
-        NEARFIELD = 1
-        MIDFIELD = 2
-        FARFIELD = 3
-
-    class OriginalMediaType(enum.IntEnum):
-        """
-        The original media the speech was recorded on.
-
-        Attributes:
-          ORIGINAL_MEDIA_TYPE_UNSPECIFIED (int): Unknown original media type.
-          AUDIO (int): The speech data is an audio recording.
-          VIDEO (int): The speech data originally recorded on a video.
-        """
-
-        ORIGINAL_MEDIA_TYPE_UNSPECIFIED = 0
-        AUDIO = 1
-        VIDEO = 2
-
-    class RecordingDeviceType(enum.IntEnum):
-        """
-        The type of device the speech was recorded with.
-
-        Attributes:
-          RECORDING_DEVICE_TYPE_UNSPECIFIED (int): The recording device is unknown.
-          SMARTPHONE (int): Speech was recorded on a smartphone.
-          PC (int): Speech was recorded using a personal computer or tablet.
-          PHONE_LINE (int): Speech was recorded over a phone line.
-          VEHICLE (int): Speech was recorded in a vehicle.
-          OTHER_OUTDOOR_DEVICE (int): Speech was recorded outdoors.
-          OTHER_INDOOR_DEVICE (int): Speech was recorded indoors.
-        """
-
-        RECORDING_DEVICE_TYPE_UNSPECIFIED = 0
-        SMARTPHONE = 1
-        PC = 2
-        PHONE_LINE = 3
-        VEHICLE = 4
-        OTHER_OUTDOOR_DEVICE = 5
-        OTHER_INDOOR_DEVICE = 6
-
-
-class StreamingRecognizeResponse(object):
-    class SpeechEventType(enum.IntEnum):
-        """
-        Indicates the type of speech event.
-
-        Attributes:
-          SPEECH_EVENT_UNSPECIFIED (int): No speech event specified.
-          END_OF_SINGLE_UTTERANCE (int): This event indicates that the server has detected the end of the user's
-          speech utterance and expects no additional speech. Therefore, the server
-          will not process additional audio (although it may subsequently return
-          additional results). The client should stop sending additional audio
-          data, half-close the gRPC connection, and wait for any additional
-          results until the server closes the gRPC connection. This event is only
-          sent if ``single_utterance`` was set to ``true``, and is not used
-          otherwise.
-        """
-
-        SPEECH_EVENT_UNSPECIFIED = 0
-        END_OF_SINGLE_UTTERANCE = 1
diff --git a/google/cloud/speech_v1p1beta1/gapic/speech_client.py b/google/cloud/speech_v1p1beta1/gapic/speech_client.py
deleted file mode 100644
index 5703df5b..00000000
--- a/google/cloud/speech_v1p1beta1/gapic/speech_client.py
+++ /dev/null
@@ -1,412 +0,0 @@
-# -*- coding: utf-8 -*-
-#
-# Copyright 2020 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-"""Accesses the google.cloud.speech.v1p1beta1 Speech API."""
-
-import pkg_resources
-import warnings
-
-from google.oauth2 import service_account
-import google.api_core.client_options
-import google.api_core.gapic_v1.client_info
-import google.api_core.gapic_v1.config
-import google.api_core.gapic_v1.method
-import google.api_core.grpc_helpers
-import google.api_core.operation
-import google.api_core.operations_v1
-import google.api_core.protobuf_helpers
-import grpc
-
-from google.cloud.speech_v1p1beta1.gapic import enums
-from google.cloud.speech_v1p1beta1.gapic import speech_client_config
-from google.cloud.speech_v1p1beta1.gapic.transports import speech_grpc_transport
-from google.cloud.speech_v1p1beta1.proto import cloud_speech_pb2
-from google.cloud.speech_v1p1beta1.proto import cloud_speech_pb2_grpc
-from google.longrunning import operations_pb2
-
-
-_GAPIC_LIBRARY_VERSION = pkg_resources.get_distribution("google-cloud-speech").version
-
-
-class SpeechClient(object):
-    """Service that implements Google Cloud Speech API."""
-
-    SERVICE_ADDRESS = "speech.googleapis.com:443"
-    """The default address of the service."""
-
-    # The name of the interface for this client. This is the key used to
-    # find the method configuration in the client_config dictionary.
-    _INTERFACE_NAME = "google.cloud.speech.v1p1beta1.Speech"
-
-    @classmethod
-    def from_service_account_file(cls, filename, *args, **kwargs):
-        """Creates an instance of this client using the provided credentials
-        file.
-
-        Args:
-            filename (str): The path to the service account private key json
-                file.
-            args: Additional arguments to pass to the constructor.
-            kwargs: Additional arguments to pass to the constructor.
-
-        Returns:
-            SpeechClient: The constructed client.
-        """
-        credentials = service_account.Credentials.from_service_account_file(filename)
-        kwargs["credentials"] = credentials
-        return cls(*args, **kwargs)
-
-    from_service_account_json = from_service_account_file
-
-    def __init__(
-        self,
-        transport=None,
-        channel=None,
-        credentials=None,
-        client_config=None,
-        client_info=None,
-        client_options=None,
-    ):
-        """Constructor.
-
-        Args:
-            transport (Union[~.SpeechGrpcTransport,
-                    Callable[[~.Credentials, type], ~.SpeechGrpcTransport]): A transport
-                instance, responsible for actually making the API calls.
-                The default transport uses the gRPC protocol.
-                This argument may also be a callable which returns a
-                transport instance. Callables will be sent the credentials
-                as the first argument and the default transport class as
-                the second argument.
-            channel (grpc.Channel): DEPRECATED. A ``Channel`` instance
-                through which to make calls. This argument is mutually exclusive
-                with ``credentials``; providing both will raise an exception.
-            credentials (google.auth.credentials.Credentials): The
-                authorization credentials to attach to requests. These
-                credentials identify this application to the service. If none
-                are specified, the client will attempt to ascertain the
-                credentials from the environment.
-                This argument is mutually exclusive with providing a
-                transport instance to ``transport``; doing so will raise
-                an exception.
-            client_config (dict): DEPRECATED. A dictionary of call options for
-                each method. If not specified, the default configuration is used.
-            client_info (google.api_core.gapic_v1.client_info.ClientInfo):
-                The client info used to send a user-agent string along with
-                API requests. If ``None``, then default info will be used.
-                Generally, you only need to set this if you're developing
-                your own client library.
-            client_options (Union[dict, google.api_core.client_options.ClientOptions]):
-                Client options used to set user options on the client. API Endpoint
-                should be set through client_options.
-        """
-        # Raise deprecation warnings for things we want to go away.
-        if client_config is not None:
-            warnings.warn(
-                "The `client_config` argument is deprecated.",
-                PendingDeprecationWarning,
-                stacklevel=2,
-            )
-        else:
-            client_config = speech_client_config.config
-
-        if channel:
-            warnings.warn(
-                "The `channel` argument is deprecated; use " "`transport` instead.",
-                PendingDeprecationWarning,
-                stacklevel=2,
-            )
-
-        api_endpoint = self.SERVICE_ADDRESS
-        if client_options:
-            if type(client_options) == dict:
-                client_options = google.api_core.client_options.from_dict(
-                    client_options
-                )
-            if client_options.api_endpoint:
-                api_endpoint = client_options.api_endpoint
-
-        # Instantiate the transport.
-        # The transport is responsible for handling serialization and
-        # deserialization and actually sending data to the service.
-        if transport:
-            if callable(transport):
-                self.transport = transport(
-                    credentials=credentials,
-                    default_class=speech_grpc_transport.SpeechGrpcTransport,
-                    address=api_endpoint,
-                )
-            else:
-                if credentials:
-                    raise ValueError(
-                        "Received both a transport instance and "
-                        "credentials; these are mutually exclusive."
-                    )
-                self.transport = transport
-        else:
-            self.transport = speech_grpc_transport.SpeechGrpcTransport(
-                address=api_endpoint, channel=channel, credentials=credentials
-            )
-
-        if client_info is None:
-            client_info = google.api_core.gapic_v1.client_info.ClientInfo(
-                gapic_version=_GAPIC_LIBRARY_VERSION
-            )
-        else:
-            client_info.gapic_version = _GAPIC_LIBRARY_VERSION
-        self._client_info = client_info
-
-        # Parse out the default settings for retry and timeout for each RPC
-        # from the client configuration.
-        # (Ordinarily, these are the defaults specified in the `*_config.py`
-        # file next to this one.)
-        self._method_configs = google.api_core.gapic_v1.config.parse_method_configs(
-            client_config["interfaces"][self._INTERFACE_NAME]
-        )
-
-        # Save a dictionary of cached API call functions.
-        # These are the actual callables which invoke the proper
-        # transport methods, wrapped with `wrap_method` to add retry,
-        # timeout, and the like.
-        self._inner_api_calls = {}
-
-    # Service calls
-    def recognize(
-        self,
-        config,
-        audio,
-        retry=google.api_core.gapic_v1.method.DEFAULT,
-        timeout=google.api_core.gapic_v1.method.DEFAULT,
-        metadata=None,
-    ):
-        """
-        Performs synchronous speech recognition: receive results after all audio
-        has been sent and processed.
-
-        Example:
-            >>> from google.cloud import speech_v1p1beta1
-            >>> from google.cloud.speech_v1p1beta1 import enums
-            >>>
-            >>> client = speech_v1p1beta1.SpeechClient()
-            >>>
-            >>> encoding = enums.RecognitionConfig.AudioEncoding.FLAC
-            >>> sample_rate_hertz = 44100
-            >>> language_code = 'en-US'
-            >>> config = {'encoding': encoding, 'sample_rate_hertz': sample_rate_hertz, 'language_code': language_code}
-            >>> uri = 'gs://bucket_name/file_name.flac'
-            >>> audio = {'uri': uri}
-            >>>
-            >>> response = client.recognize(config, audio)
-
-        Args:
-            config (Union[dict, ~google.cloud.speech_v1p1beta1.types.RecognitionConfig]): Required. Provides information to the recognizer that specifies how to
-                process the request.
-
-                If a dict is provided, it must be of the same form as the protobuf
-                message :class:`~google.cloud.speech_v1p1beta1.types.RecognitionConfig`
-            audio (Union[dict, ~google.cloud.speech_v1p1beta1.types.RecognitionAudio]): Required. The audio data to be recognized.
-
-                If a dict is provided, it must be of the same form as the protobuf
-                message :class:`~google.cloud.speech_v1p1beta1.types.RecognitionAudio`
-            retry (Optional[google.api_core.retry.Retry]):  A retry object used
-                to retry requests. If ``None`` is specified, requests will
-                be retried using a default configuration.
-            timeout (Optional[float]): The amount of time, in seconds, to wait
-                for the request to complete. Note that if ``retry`` is
-                specified, the timeout applies to each individual attempt.
-            metadata (Optional[Sequence[Tuple[str, str]]]): Additional metadata
-                that is provided to the method.
-
-        Returns:
-            A :class:`~google.cloud.speech_v1p1beta1.types.RecognizeResponse` instance.
-
-        Raises:
-            google.api_core.exceptions.GoogleAPICallError: If the request
-                    failed for any reason.
-            google.api_core.exceptions.RetryError: If the request failed due
-                    to a retryable error and retry attempts failed.
-            ValueError: If the parameters are invalid.
-        """
-        # Wrap the transport method to add retry and timeout logic.
-        if "recognize" not in self._inner_api_calls:
-            self._inner_api_calls[
-                "recognize"
-            ] = google.api_core.gapic_v1.method.wrap_method(
-                self.transport.recognize,
-                default_retry=self._method_configs["Recognize"].retry,
-                default_timeout=self._method_configs["Recognize"].timeout,
-                client_info=self._client_info,
-            )
-
-        request = cloud_speech_pb2.RecognizeRequest(config=config, audio=audio)
-        return self._inner_api_calls["recognize"](
-            request, retry=retry, timeout=timeout, metadata=metadata
-        )
-
-    def long_running_recognize(
-        self,
-        config,
-        audio,
-        retry=google.api_core.gapic_v1.method.DEFAULT,
-        timeout=google.api_core.gapic_v1.method.DEFAULT,
-        metadata=None,
-    ):
-        """
-        Performs asynchronous speech recognition: receive results via the
-        google.longrunning.Operations interface. Returns either an
-        ``Operation.error`` or an ``Operation.response`` which contains a
-        ``LongRunningRecognizeResponse`` message. For more information on
-        asynchronous speech recognition, see the
-        `how-to <https://cloud.google.com/speech-to-text/docs/async-recognize>`__.
-
-        Example:
-            >>> from google.cloud import speech_v1p1beta1
-            >>> from google.cloud.speech_v1p1beta1 import enums
-            >>>
-            >>> client = speech_v1p1beta1.SpeechClient()
-            >>>
-            >>> encoding = enums.RecognitionConfig.AudioEncoding.FLAC
-            >>> sample_rate_hertz = 44100
-            >>> language_code = 'en-US'
-            >>> config = {'encoding': encoding, 'sample_rate_hertz': sample_rate_hertz, 'language_code': language_code}
-            >>> uri = 'gs://bucket_name/file_name.flac'
-            >>> audio = {'uri': uri}
-            >>>
-            >>> response = client.long_running_recognize(config, audio)
-            >>>
-            >>> def callback(operation_future):
-            ...     # Handle result.
-            ...     result = operation_future.result()
-            >>>
-            >>> response.add_done_callback(callback)
-            >>>
-            >>> # Handle metadata.
-            >>> metadata = response.metadata()
-
-        Args:
-            config (Union[dict, ~google.cloud.speech_v1p1beta1.types.RecognitionConfig]): Required. Provides information to the recognizer that specifies how to
-                process the request.
-
-                If a dict is provided, it must be of the same form as the protobuf
-                message :class:`~google.cloud.speech_v1p1beta1.types.RecognitionConfig`
-            audio (Union[dict, ~google.cloud.speech_v1p1beta1.types.RecognitionAudio]): Required. The audio data to be recognized.
-
-                If a dict is provided, it must be of the same form as the protobuf
-                message :class:`~google.cloud.speech_v1p1beta1.types.RecognitionAudio`
-            retry (Optional[google.api_core.retry.Retry]):  A retry object used
-                to retry requests. If ``None`` is specified, requests will
-                be retried using a default configuration.
-            timeout (Optional[float]): The amount of time, in seconds, to wait
-                for the request to complete. Note that if ``retry`` is
-                specified, the timeout applies to each individual attempt.
-            metadata (Optional[Sequence[Tuple[str, str]]]): Additional metadata
-                that is provided to the method.
-
-        Returns:
-            A :class:`~google.cloud.speech_v1p1beta1.types._OperationFuture` instance.
-
-        Raises:
-            google.api_core.exceptions.GoogleAPICallError: If the request
-                    failed for any reason.
-            google.api_core.exceptions.RetryError: If the request failed due
-                    to a retryable error and retry attempts failed.
-            ValueError: If the parameters are invalid.
-        """
-        # Wrap the transport method to add retry and timeout logic.
-        if "long_running_recognize" not in self._inner_api_calls:
-            self._inner_api_calls[
-                "long_running_recognize"
-            ] = google.api_core.gapic_v1.method.wrap_method(
-                self.transport.long_running_recognize,
-                default_retry=self._method_configs["LongRunningRecognize"].retry,
-                default_timeout=self._method_configs["LongRunningRecognize"].timeout,
-                client_info=self._client_info,
-            )
-
-        request = cloud_speech_pb2.LongRunningRecognizeRequest(
-            config=config, audio=audio
-        )
-        operation = self._inner_api_calls["long_running_recognize"](
-            request, retry=retry, timeout=timeout, metadata=metadata
-        )
-        return google.api_core.operation.from_gapic(
-            operation,
-            self.transport._operations_client,
-            cloud_speech_pb2.LongRunningRecognizeResponse,
-            metadata_type=cloud_speech_pb2.LongRunningRecognizeMetadata,
-        )
-
-    def streaming_recognize(
-        self,
-        requests,
-        retry=google.api_core.gapic_v1.method.DEFAULT,
-        timeout=google.api_core.gapic_v1.method.DEFAULT,
-        metadata=None,
-    ):
-        """
-        Performs bidirectional streaming speech recognition: receive results while
-        sending audio. This method is only available via the gRPC API (not REST).
-
-        EXPERIMENTAL: This method interface might change in the future.
-
-        Example:
-            >>> from google.cloud import speech_v1p1beta1
-            >>>
-            >>> client = speech_v1p1beta1.SpeechClient()
-            >>>
-            >>> request = {}
-            >>>
-            >>> requests = [request]
-            >>> for element in client.streaming_recognize(requests):
-            ...     # process element
-            ...     pass
-
-        Args:
-            requests (iterator[dict|google.cloud.speech_v1p1beta1.proto.cloud_speech_pb2.StreamingRecognizeRequest]): The input objects. If a dict is provided, it must be of the
-                same form as the protobuf message :class:`~google.cloud.speech_v1p1beta1.types.StreamingRecognizeRequest`
-            retry (Optional[google.api_core.retry.Retry]):  A retry object used
-                to retry requests. If ``None`` is specified, requests will
-                be retried using a default configuration.
-            timeout (Optional[float]): The amount of time, in seconds, to wait
-                for the request to complete. Note that if ``retry`` is
-                specified, the timeout applies to each individual attempt.
-            metadata (Optional[Sequence[Tuple[str, str]]]): Additional metadata
-                that is provided to the method.
-
-        Returns:
-            Iterable[~google.cloud.speech_v1p1beta1.types.StreamingRecognizeResponse].
-
-        Raises:
-            google.api_core.exceptions.GoogleAPICallError: If the request
-                    failed for any reason.
-            google.api_core.exceptions.RetryError: If the request failed due
-                    to a retryable error and retry attempts failed.
-            ValueError: If the parameters are invalid.
-        """
-        # Wrap the transport method to add retry and timeout logic.
-        if "streaming_recognize" not in self._inner_api_calls:
-            self._inner_api_calls[
-                "streaming_recognize"
-            ] = google.api_core.gapic_v1.method.wrap_method(
-                self.transport.streaming_recognize,
-                default_retry=self._method_configs["StreamingRecognize"].retry,
-                default_timeout=self._method_configs["StreamingRecognize"].timeout,
-                client_info=self._client_info,
-            )
-
-        return self._inner_api_calls["streaming_recognize"](
-            requests, retry=retry, timeout=timeout, metadata=metadata
-        )
diff --git a/google/cloud/speech_v1p1beta1/gapic/speech_client_config.py b/google/cloud/speech_v1p1beta1/gapic/speech_client_config.py
deleted file mode 100644
index c405dd03..00000000
--- a/google/cloud/speech_v1p1beta1/gapic/speech_client_config.py
+++ /dev/null
@@ -1,38 +0,0 @@
-config = {
-    "interfaces": {
-        "google.cloud.speech.v1p1beta1.Speech": {
-            "retry_codes": {
-                "idempotent": ["DEADLINE_EXCEEDED", "UNAVAILABLE"],
-                "non_idempotent": [],
-            },
-            "retry_params": {
-                "default": {
-                    "initial_retry_delay_millis": 100,
-                    "retry_delay_multiplier": 1.3,
-                    "max_retry_delay_millis": 60000,
-                    "initial_rpc_timeout_millis": 1000000,
-                    "rpc_timeout_multiplier": 1.0,
-                    "max_rpc_timeout_millis": 1000000,
-                    "total_timeout_millis": 5000000,
-                }
-            },
-            "methods": {
-                "Recognize": {
-                    "timeout_millis": 60000,
-                    "retry_codes_name": "non_idempotent",
-                    "retry_params_name": "default",
-                },
-                "LongRunningRecognize": {
-                    "timeout_millis": 60000,
-                    "retry_codes_name": "non_idempotent",
-                    "retry_params_name": "default",
-                },
-                "StreamingRecognize": {
-                    "timeout_millis": 60000,
-                    "retry_codes_name": "non_idempotent",
-                    "retry_params_name": "default",
-                },
-            },
-        }
-    }
-}
diff --git a/google/cloud/speech_v1p1beta1/gapic/transports/__init__.py b/google/cloud/speech_v1p1beta1/gapic/transports/__init__.py
deleted file mode 100644
index e69de29b..00000000
diff --git a/google/cloud/speech_v1p1beta1/gapic/transports/speech_grpc_transport.py b/google/cloud/speech_v1p1beta1/gapic/transports/speech_grpc_transport.py
deleted file mode 100644
index a8e42b09..00000000
--- a/google/cloud/speech_v1p1beta1/gapic/transports/speech_grpc_transport.py
+++ /dev/null
@@ -1,160 +0,0 @@
-# -*- coding: utf-8 -*-
-#
-# Copyright 2020 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-
-import google.api_core.grpc_helpers
-import google.api_core.operations_v1
-
-from google.cloud.speech_v1p1beta1.proto import cloud_speech_pb2_grpc
-
-
-class SpeechGrpcTransport(object):
-    """gRPC transport class providing stubs for
-    google.cloud.speech.v1p1beta1 Speech API.
-
-    The transport provides access to the raw gRPC stubs,
-    which can be used to take advantage of advanced
-    features of gRPC.
-    """
-
-    # The scopes needed to make gRPC calls to all of the methods defined
-    # in this service.
-    _OAUTH_SCOPES = ("https://www.googleapis.com/auth/cloud-platform",)
-
-    def __init__(
-        self, channel=None, credentials=None, address="speech.googleapis.com:443"
-    ):
-        """Instantiate the transport class.
-
-        Args:
-            channel (grpc.Channel): A ``Channel`` instance through
-                which to make calls. This argument is mutually exclusive
-                with ``credentials``; providing both will raise an exception.
-            credentials (google.auth.credentials.Credentials): The
-                authorization credentials to attach to requests. These
-                credentials identify this application to the service. If none
-                are specified, the client will attempt to ascertain the
-                credentials from the environment.
-            address (str): The address where the service is hosted.
-        """
-        # If both `channel` and `credentials` are specified, raise an
-        # exception (channels come with credentials baked in already).
-        if channel is not None and credentials is not None:
-            raise ValueError(
-                "The `channel` and `credentials` arguments are mutually " "exclusive."
-            )
-
-        # Create the channel.
-        if channel is None:
-            channel = self.create_channel(
-                address=address,
-                credentials=credentials,
-                options={
-                    "grpc.max_send_message_length": -1,
-                    "grpc.max_receive_message_length": -1,
-                }.items(),
-            )
-
-        self._channel = channel
-
-        # gRPC uses objects called "stubs" that are bound to the
-        # channel and provide a basic method for each RPC.
-        self._stubs = {"speech_stub": cloud_speech_pb2_grpc.SpeechStub(channel)}
-
-        # Because this API includes a method that returns a
-        # long-running operation (proto: google.longrunning.Operation),
-        # instantiate an LRO client.
-        self._operations_client = google.api_core.operations_v1.OperationsClient(
-            channel
-        )
-
-    @classmethod
-    def create_channel(
-        cls, address="speech.googleapis.com:443", credentials=None, **kwargs
-    ):
-        """Create and return a gRPC channel object.
-
-        Args:
-            address (str): The host for the channel to use.
-            credentials (~.Credentials): The
-                authorization credentials to attach to requests. These
-                credentials identify this application to the service. If
-                none are specified, the client will attempt to ascertain
-                the credentials from the environment.
-            kwargs (dict): Keyword arguments, which are passed to the
-                channel creation.
-
-        Returns:
-            grpc.Channel: A gRPC channel object.
-        """
-        return google.api_core.grpc_helpers.create_channel(
-            address, credentials=credentials, scopes=cls._OAUTH_SCOPES, **kwargs
-        )
-
-    @property
-    def channel(self):
-        """The gRPC channel used by the transport.
-
-        Returns:
-            grpc.Channel: A gRPC channel object.
-        """
-        return self._channel
-
-    @property
-    def recognize(self):
-        """Return the gRPC stub for :meth:`SpeechClient.recognize`.
-
-        Performs synchronous speech recognition: receive results after all audio
-        has been sent and processed.
-
-        Returns:
-            Callable: A callable which accepts the appropriate
-                deserialized request object and returns a
-                deserialized response object.
-        """
-        return self._stubs["speech_stub"].Recognize
-
-    @property
-    def long_running_recognize(self):
-        """Return the gRPC stub for :meth:`SpeechClient.long_running_recognize`.
-
-        Performs asynchronous speech recognition: receive results via the
-        google.longrunning.Operations interface. Returns either an
-        ``Operation.error`` or an ``Operation.response`` which contains a
-        ``LongRunningRecognizeResponse`` message. For more information on
-        asynchronous speech recognition, see the
-        `how-to <https://cloud.google.com/speech-to-text/docs/async-recognize>`__.
-
-        Returns:
-            Callable: A callable which accepts the appropriate
-                deserialized request object and returns a
-                deserialized response object.
-        """
-        return self._stubs["speech_stub"].LongRunningRecognize
-
-    @property
-    def streaming_recognize(self):
-        """Return the gRPC stub for :meth:`SpeechClient.streaming_recognize`.
-
-        Performs bidirectional streaming speech recognition: receive results while
-        sending audio. This method is only available via the gRPC API (not REST).
-
-        Returns:
-            Callable: A callable which accepts the appropriate
-                deserialized request object and returns a
-                deserialized response object.
-        """
-        return self._stubs["speech_stub"].StreamingRecognize
diff --git a/google/cloud/speech_v1p1beta1/proto/__init__.py b/google/cloud/speech_v1p1beta1/proto/__init__.py
deleted file mode 100644
index e69de29b..00000000
diff --git a/google/cloud/speech_v1p1beta1/proto/cloud_speech.proto b/google/cloud/speech_v1p1beta1/proto/cloud_speech.proto
deleted file mode 100644
index 7718b0eb..00000000
--- a/google/cloud/speech_v1p1beta1/proto/cloud_speech.proto
+++ /dev/null
@@ -1,816 +0,0 @@
-// Copyright 2019 Google LLC.
-//
-// Licensed under the Apache License, Version 2.0 (the "License");
-// you may not use this file except in compliance with the License.
-// You may obtain a copy of the License at
-//
-//     http://www.apache.org/licenses/LICENSE-2.0
-//
-// Unless required by applicable law or agreed to in writing, software
-// distributed under the License is distributed on an "AS IS" BASIS,
-// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-// See the License for the specific language governing permissions and
-// limitations under the License.
-//
-
-syntax = "proto3";
-
-package google.cloud.speech.v1p1beta1;
-
-import "google/api/annotations.proto";
-import "google/api/client.proto";
-import "google/api/field_behavior.proto";
-import "google/longrunning/operations.proto";
-import "google/protobuf/any.proto";
-import "google/protobuf/duration.proto";
-import "google/protobuf/timestamp.proto";
-import "google/rpc/status.proto";
-
-option cc_enable_arenas = true;
-option go_package = "google.golang.org/genproto/googleapis/cloud/speech/v1p1beta1;speech";
-option java_multiple_files = true;
-option java_outer_classname = "SpeechProto";
-option java_package = "com.google.cloud.speech.v1p1beta1";
-option objc_class_prefix = "GCS";
-
-// Service that implements Google Cloud Speech API.
-service Speech {
-  option (google.api.default_host) = "speech.googleapis.com";
-  option (google.api.oauth_scopes) = "https://www.googleapis.com/auth/cloud-platform";
-
-  // Performs synchronous speech recognition: receive results after all audio
-  // has been sent and processed.
-  rpc Recognize(RecognizeRequest) returns (RecognizeResponse) {
-    option (google.api.http) = {
-      post: "/v1p1beta1/speech:recognize"
-      body: "*"
-    };
-    option (google.api.method_signature) = "config,audio";
-  }
-
-  // Performs asynchronous speech recognition: receive results via the
-  // google.longrunning.Operations interface. Returns either an
-  // `Operation.error` or an `Operation.response` which contains
-  // a `LongRunningRecognizeResponse` message.
-  // For more information on asynchronous speech recognition, see the
-  // [how-to](https://cloud.google.com/speech-to-text/docs/async-recognize).
-  rpc LongRunningRecognize(LongRunningRecognizeRequest) returns (google.longrunning.Operation) {
-    option (google.api.http) = {
-      post: "/v1p1beta1/speech:longrunningrecognize"
-      body: "*"
-    };
-    option (google.api.method_signature) = "config,audio";
-    option (google.longrunning.operation_info) = {
-      response_type: "LongRunningRecognizeResponse"
-      metadata_type: "LongRunningRecognizeMetadata"
-    };
-  }
-
-  // Performs bidirectional streaming speech recognition: receive results while
-  // sending audio. This method is only available via the gRPC API (not REST).
-  rpc StreamingRecognize(stream StreamingRecognizeRequest) returns (stream StreamingRecognizeResponse) {
-  }
-}
-
-// The top-level message sent by the client for the `Recognize` method.
-message RecognizeRequest {
-  // Required. Provides information to the recognizer that specifies how to
-  // process the request.
-  RecognitionConfig config = 1 [(google.api.field_behavior) = REQUIRED];
-
-  // Required. The audio data to be recognized.
-  RecognitionAudio audio = 2 [(google.api.field_behavior) = REQUIRED];
-}
-
-// The top-level message sent by the client for the `LongRunningRecognize`
-// method.
-message LongRunningRecognizeRequest {
-  // Required. Provides information to the recognizer that specifies how to
-  // process the request.
-  RecognitionConfig config = 1 [(google.api.field_behavior) = REQUIRED];
-
-  // Required. The audio data to be recognized.
-  RecognitionAudio audio = 2 [(google.api.field_behavior) = REQUIRED];
-}
-
-// The top-level message sent by the client for the `StreamingRecognize` method.
-// Multiple `StreamingRecognizeRequest` messages are sent. The first message
-// must contain a `streaming_config` message and must not contain
-// `audio_content`. All subsequent messages must contain `audio_content` and
-// must not contain a `streaming_config` message.
-message StreamingRecognizeRequest {
-  // The streaming request, which is either a streaming config or audio content.
-  oneof streaming_request {
-    // Provides information to the recognizer that specifies how to process the
-    // request. The first `StreamingRecognizeRequest` message must contain a
-    // `streaming_config`  message.
-    StreamingRecognitionConfig streaming_config = 1;
-
-    // The audio data to be recognized. Sequential chunks of audio data are sent
-    // in sequential `StreamingRecognizeRequest` messages. The first
-    // `StreamingRecognizeRequest` message must not contain `audio_content` data
-    // and all subsequent `StreamingRecognizeRequest` messages must contain
-    // `audio_content` data. The audio bytes must be encoded as specified in
-    // `RecognitionConfig`. Note: as with all bytes fields, proto buffers use a
-    // pure binary representation (not base64). See
-    // [content limits](https://cloud.google.com/speech-to-text/quotas#content).
-    bytes audio_content = 2;
-  }
-}
-
-// Provides information to the recognizer that specifies how to process the
-// request.
-message StreamingRecognitionConfig {
-  // Required. Provides information to the recognizer that specifies how to
-  // process the request.
-  RecognitionConfig config = 1 [(google.api.field_behavior) = REQUIRED];
-
-  // If `false` or omitted, the recognizer will perform continuous
-  // recognition (continuing to wait for and process audio even if the user
-  // pauses speaking) until the client closes the input stream (gRPC API) or
-  // until the maximum time limit has been reached. May return multiple
-  // `StreamingRecognitionResult`s with the `is_final` flag set to `true`.
-  //
-  // If `true`, the recognizer will detect a single spoken utterance. When it
-  // detects that the user has paused or stopped speaking, it will return an
-  // `END_OF_SINGLE_UTTERANCE` event and cease recognition. It will return no
-  // more than one `StreamingRecognitionResult` with the `is_final` flag set to
-  // `true`.
-  bool single_utterance = 2;
-
-  // If `true`, interim results (tentative hypotheses) may be
-  // returned as they become available (these interim results are indicated with
-  // the `is_final=false` flag).
-  // If `false` or omitted, only `is_final=true` result(s) are returned.
-  bool interim_results = 3;
-}
-
-// Provides information to the recognizer that specifies how to process the
-// request.
-message RecognitionConfig {
-  // The encoding of the audio data sent in the request.
-  //
-  // All encodings support only 1 channel (mono) audio, unless the
-  // `audio_channel_count` and `enable_separate_recognition_per_channel` fields
-  // are set.
-  //
-  // For best results, the audio source should be captured and transmitted using
-  // a lossless encoding (`FLAC` or `LINEAR16`). The accuracy of the speech
-  // recognition can be reduced if lossy codecs are used to capture or transmit
-  // audio, particularly if background noise is present. Lossy codecs include
-  // `MULAW`, `AMR`, `AMR_WB`, `OGG_OPUS`, `SPEEX_WITH_HEADER_BYTE`, and `MP3`.
-  //
-  // The `FLAC` and `WAV` audio file formats include a header that describes the
-  // included audio content. You can request recognition for `WAV` files that
-  // contain either `LINEAR16` or `MULAW` encoded audio.
-  // If you send `FLAC` or `WAV` audio file format in
-  // your request, you do not need to specify an `AudioEncoding`; the audio
-  // encoding format is determined from the file header. If you specify
-  // an `AudioEncoding` when you send  send `FLAC` or `WAV` audio, the
-  // encoding configuration must match the encoding described in the audio
-  // header; otherwise the request returns an
-  // [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT] error code.
-  enum AudioEncoding {
-    // Not specified.
-    ENCODING_UNSPECIFIED = 0;
-
-    // Uncompressed 16-bit signed little-endian samples (Linear PCM).
-    LINEAR16 = 1;
-
-    // `FLAC` (Free Lossless Audio
-    // Codec) is the recommended encoding because it is
-    // lossless--therefore recognition is not compromised--and
-    // requires only about half the bandwidth of `LINEAR16`. `FLAC` stream
-    // encoding supports 16-bit and 24-bit samples, however, not all fields in
-    // `STREAMINFO` are supported.
-    FLAC = 2;
-
-    // 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
-    MULAW = 3;
-
-    // Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
-    AMR = 4;
-
-    // Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000.
-    AMR_WB = 5;
-
-    // Opus encoded audio frames in Ogg container
-    // ([OggOpus](https://wiki.xiph.org/OggOpus)).
-    // `sample_rate_hertz` must be one of 8000, 12000, 16000, 24000, or 48000.
-    OGG_OPUS = 6;
-
-    // Although the use of lossy encodings is not recommended, if a very low
-    // bitrate encoding is required, `OGG_OPUS` is highly preferred over
-    // Speex encoding. The [Speex](https://speex.org/)  encoding supported by
-    // Cloud Speech API has a header byte in each block, as in MIME type
-    // `audio/x-speex-with-header-byte`.
-    // It is a variant of the RTP Speex encoding defined in
-    // [RFC 5574](https://tools.ietf.org/html/rfc5574).
-    // The stream is a sequence of blocks, one block per RTP packet. Each block
-    // starts with a byte containing the length of the block, in bytes, followed
-    // by one or more frames of Speex data, padded to an integral number of
-    // bytes (octets) as specified in RFC 5574. In other words, each RTP header
-    // is replaced with a single byte containing the block length. Only Speex
-    // wideband is supported. `sample_rate_hertz` must be 16000.
-    SPEEX_WITH_HEADER_BYTE = 7;
-
-    // MP3 audio. Support all standard MP3 bitrates (which range from 32-320
-    // kbps). When using this encoding, `sample_rate_hertz` can be optionally
-    // unset if not known.
-    MP3 = 8;
-  }
-
-  // Encoding of audio data sent in all `RecognitionAudio` messages.
-  // This field is optional for `FLAC` and `WAV` audio files and required
-  // for all other audio formats. For details, see [AudioEncoding][google.cloud.speech.v1p1beta1.RecognitionConfig.AudioEncoding].
-  AudioEncoding encoding = 1;
-
-  // Sample rate in Hertz of the audio data sent in all
-  // `RecognitionAudio` messages. Valid values are: 8000-48000.
-  // 16000 is optimal. For best results, set the sampling rate of the audio
-  // source to 16000 Hz. If that's not possible, use the native sample rate of
-  // the audio source (instead of re-sampling).
-  // This field is optional for FLAC and WAV audio files, but is
-  // required for all other audio formats. For details, see [AudioEncoding][google.cloud.speech.v1p1beta1.RecognitionConfig.AudioEncoding].
-  int32 sample_rate_hertz = 2;
-
-  // The number of channels in the input audio data.
-  // ONLY set this for MULTI-CHANNEL recognition.
-  // Valid values for LINEAR16 and FLAC are `1`-`8`.
-  // Valid values for OGG_OPUS are '1'-'254'.
-  // Valid value for MULAW, AMR, AMR_WB and SPEEX_WITH_HEADER_BYTE is only `1`.
-  // If `0` or omitted, defaults to one channel (mono).
-  // Note: We only recognize the first channel by default.
-  // To perform independent recognition on each channel set
-  // `enable_separate_recognition_per_channel` to 'true'.
-  int32 audio_channel_count = 7;
-
-  // This needs to be set to `true` explicitly and `audio_channel_count` > 1
-  // to get each channel recognized separately. The recognition result will
-  // contain a `channel_tag` field to state which channel that result belongs
-  // to. If this is not true, we will only recognize the first channel. The
-  // request is billed cumulatively for all channels recognized:
-  // `audio_channel_count` multiplied by the length of the audio.
-  bool enable_separate_recognition_per_channel = 12;
-
-  // Required. The language of the supplied audio as a
-  // [BCP-47](https://www.rfc-editor.org/rfc/bcp/bcp47.txt) language tag.
-  // Example: "en-US".
-  // See [Language
-  // Support](https://cloud.google.com/speech-to-text/docs/languages) for a list
-  // of the currently supported language codes.
-  string language_code = 3 [(google.api.field_behavior) = REQUIRED];
-
-  // A list of up to 3 additional
-  // [BCP-47](https://www.rfc-editor.org/rfc/bcp/bcp47.txt) language tags,
-  // listing possible alternative languages of the supplied audio.
-  // See [Language
-  // Support](https://cloud.google.com/speech-to-text/docs/languages) for a list
-  // of the currently supported language codes. If alternative languages are
-  // listed, recognition result will contain recognition in the most likely
-  // language detected including the main language_code. The recognition result
-  // will include the language tag of the language detected in the audio. Note:
-  // This feature is only supported for Voice Command and Voice Search use cases
-  // and performance may vary for other use cases (e.g., phone call
-  // transcription).
-  repeated string alternative_language_codes = 18;
-
-  // Maximum number of recognition hypotheses to be returned.
-  // Specifically, the maximum number of `SpeechRecognitionAlternative` messages
-  // within each `SpeechRecognitionResult`.
-  // The server may return fewer than `max_alternatives`.
-  // Valid values are `0`-`30`. A value of `0` or `1` will return a maximum of
-  // one. If omitted, will return a maximum of one.
-  int32 max_alternatives = 4;
-
-  // If set to `true`, the server will attempt to filter out
-  // profanities, replacing all but the initial character in each filtered word
-  // with asterisks, e.g. "f***". If set to `false` or omitted, profanities
-  // won't be filtered out.
-  bool profanity_filter = 5;
-
-  // Array of [SpeechContext][google.cloud.speech.v1p1beta1.SpeechContext].
-  // A means to provide context to assist the speech recognition. For more
-  // information, see
-  // [speech
-  // adaptation](https://cloud.google.com/speech-to-text/docs/context-strength).
-  repeated SpeechContext speech_contexts = 6;
-
-  // If `true`, the top result includes a list of words and
-  // the start and end time offsets (timestamps) for those words. If
-  // `false`, no word-level time offset information is returned. The default is
-  // `false`.
-  bool enable_word_time_offsets = 8;
-
-  // If `true`, the top result includes a list of words and the
-  // confidence for those words. If `false`, no word-level confidence
-  // information is returned. The default is `false`.
-  bool enable_word_confidence = 15;
-
-  // If 'true', adds punctuation to recognition result hypotheses.
-  // This feature is only available in select languages. Setting this for
-  // requests in other languages has no effect at all.
-  // The default 'false' value does not add punctuation to result hypotheses.
-  // Note: This is currently offered as an experimental service, complimentary
-  // to all users. In the future this may be exclusively available as a
-  // premium feature.
-  bool enable_automatic_punctuation = 11;
-
-  // If 'true', enables speaker detection for each recognized word in
-  // the top alternative of the recognition result using a speaker_tag provided
-  // in the WordInfo.
-  // Note: Use diarization_config instead.
-  bool enable_speaker_diarization = 16 [deprecated = true];
-
-  // If set, specifies the estimated number of speakers in the conversation.
-  // Defaults to '2'. Ignored unless enable_speaker_diarization is set to true.
-  // Note: Use diarization_config instead.
-  int32 diarization_speaker_count = 17 [deprecated = true];
-
-  // Config to enable speaker diarization and set additional
-  // parameters to make diarization better suited for your application.
-  // Note: When this is enabled, we send all the words from the beginning of the
-  // audio for the top alternative in every consecutive STREAMING responses.
-  // This is done in order to improve our speaker tags as our models learn to
-  // identify the speakers in the conversation over time.
-  // For non-streaming requests, the diarization results will be provided only
-  // in the top alternative of the FINAL SpeechRecognitionResult.
-  SpeakerDiarizationConfig diarization_config = 19;
-
-  // Metadata regarding this request.
-  RecognitionMetadata metadata = 9;
-
-  // Which model to select for the given request. Select the model
-  // best suited to your domain to get best results. If a model is not
-  // explicitly specified, then we auto-select a model based on the parameters
-  // in the RecognitionConfig.
-  // <table>
-  //   <tr>
-  //     <td><b>Model</b></td>
-  //     <td><b>Description</b></td>
-  //   </tr>
-  //   <tr>
-  //     <td><code>command_and_search</code></td>
-  //     <td>Best for short queries such as voice commands or voice search.</td>
-  //   </tr>
-  //   <tr>
-  //     <td><code>phone_call</code></td>
-  //     <td>Best for audio that originated from a phone call (typically
-  //     recorded at an 8khz sampling rate).</td>
-  //   </tr>
-  //   <tr>
-  //     <td><code>video</code></td>
-  //     <td>Best for audio that originated from from video or includes multiple
-  //         speakers. Ideally the audio is recorded at a 16khz or greater
-  //         sampling rate. This is a premium model that costs more than the
-  //         standard rate.</td>
-  //   </tr>
-  //   <tr>
-  //     <td><code>default</code></td>
-  //     <td>Best for audio that is not one of the specific audio models.
-  //         For example, long-form audio. Ideally the audio is high-fidelity,
-  //         recorded at a 16khz or greater sampling rate.</td>
-  //   </tr>
-  // </table>
-  string model = 13;
-
-  // Set to true to use an enhanced model for speech recognition.
-  // If `use_enhanced` is set to true and the `model` field is not set, then
-  // an appropriate enhanced model is chosen if an enhanced model exists for
-  // the audio.
-  //
-  // If `use_enhanced` is true and an enhanced version of the specified model
-  // does not exist, then the speech is recognized using the standard version
-  // of the specified model.
-  bool use_enhanced = 14;
-}
-
-// Config to enable speaker diarization.
-message SpeakerDiarizationConfig {
-  // If 'true', enables speaker detection for each recognized word in
-  // the top alternative of the recognition result using a speaker_tag provided
-  // in the WordInfo.
-  bool enable_speaker_diarization = 1;
-
-  // Minimum number of speakers in the conversation. This range gives you more
-  // flexibility by allowing the system to automatically determine the correct
-  // number of speakers. If not set, the default value is 2.
-  int32 min_speaker_count = 2;
-
-  // Maximum number of speakers in the conversation. This range gives you more
-  // flexibility by allowing the system to automatically determine the correct
-  // number of speakers. If not set, the default value is 6.
-  int32 max_speaker_count = 3;
-}
-
-// Description of audio data to be recognized.
-message RecognitionMetadata {
-  // Use case categories that the audio recognition request can be described
-  // by.
-  enum InteractionType {
-    // Use case is either unknown or is something other than one of the other
-    // values below.
-    INTERACTION_TYPE_UNSPECIFIED = 0;
-
-    // Multiple people in a conversation or discussion. For example in a
-    // meeting with two or more people actively participating. Typically
-    // all the primary people speaking would be in the same room (if not,
-    // see PHONE_CALL)
-    DISCUSSION = 1;
-
-    // One or more persons lecturing or presenting to others, mostly
-    // uninterrupted.
-    PRESENTATION = 2;
-
-    // A phone-call or video-conference in which two or more people, who are
-    // not in the same room, are actively participating.
-    PHONE_CALL = 3;
-
-    // A recorded message intended for another person to listen to.
-    VOICEMAIL = 4;
-
-    // Professionally produced audio (eg. TV Show, Podcast).
-    PROFESSIONALLY_PRODUCED = 5;
-
-    // Transcribe spoken questions and queries into text.
-    VOICE_SEARCH = 6;
-
-    // Transcribe voice commands, such as for controlling a device.
-    VOICE_COMMAND = 7;
-
-    // Transcribe speech to text to create a written document, such as a
-    // text-message, email or report.
-    DICTATION = 8;
-  }
-
-  // Enumerates the types of capture settings describing an audio file.
-  enum MicrophoneDistance {
-    // Audio type is not known.
-    MICROPHONE_DISTANCE_UNSPECIFIED = 0;
-
-    // The audio was captured from a closely placed microphone. Eg. phone,
-    // dictaphone, or handheld microphone. Generally if there speaker is within
-    // 1 meter of the microphone.
-    NEARFIELD = 1;
-
-    // The speaker if within 3 meters of the microphone.
-    MIDFIELD = 2;
-
-    // The speaker is more than 3 meters away from the microphone.
-    FARFIELD = 3;
-  }
-
-  // The original media the speech was recorded on.
-  enum OriginalMediaType {
-    // Unknown original media type.
-    ORIGINAL_MEDIA_TYPE_UNSPECIFIED = 0;
-
-    // The speech data is an audio recording.
-    AUDIO = 1;
-
-    // The speech data originally recorded on a video.
-    VIDEO = 2;
-  }
-
-  // The type of device the speech was recorded with.
-  enum RecordingDeviceType {
-    // The recording device is unknown.
-    RECORDING_DEVICE_TYPE_UNSPECIFIED = 0;
-
-    // Speech was recorded on a smartphone.
-    SMARTPHONE = 1;
-
-    // Speech was recorded using a personal computer or tablet.
-    PC = 2;
-
-    // Speech was recorded over a phone line.
-    PHONE_LINE = 3;
-
-    // Speech was recorded in a vehicle.
-    VEHICLE = 4;
-
-    // Speech was recorded outdoors.
-    OTHER_OUTDOOR_DEVICE = 5;
-
-    // Speech was recorded indoors.
-    OTHER_INDOOR_DEVICE = 6;
-  }
-
-  // The use case most closely describing the audio content to be recognized.
-  InteractionType interaction_type = 1;
-
-  // The industry vertical to which this speech recognition request most
-  // closely applies. This is most indicative of the topics contained
-  // in the audio.  Use the 6-digit NAICS code to identify the industry
-  // vertical - see https://www.naics.com/search/.
-  uint32 industry_naics_code_of_audio = 3;
-
-  // The audio type that most closely describes the audio being recognized.
-  MicrophoneDistance microphone_distance = 4;
-
-  // The original media the speech was recorded on.
-  OriginalMediaType original_media_type = 5;
-
-  // The type of device the speech was recorded with.
-  RecordingDeviceType recording_device_type = 6;
-
-  // The device used to make the recording.  Examples 'Nexus 5X' or
-  // 'Polycom SoundStation IP 6000' or 'POTS' or 'VoIP' or
-  // 'Cardioid Microphone'.
-  string recording_device_name = 7;
-
-  // Mime type of the original audio file.  For example `audio/m4a`,
-  // `audio/x-alaw-basic`, `audio/mp3`, `audio/3gpp`.
-  // A list of possible audio mime types is maintained at
-  // http://www.iana.org/assignments/media-types/media-types.xhtml#audio
-  string original_mime_type = 8;
-
-  // Obfuscated (privacy-protected) ID of the user, to identify number of
-  // unique users using the service.
-  int64 obfuscated_id = 9 [deprecated = true];
-
-  // Description of the content. Eg. "Recordings of federal supreme court
-  // hearings from 2012".
-  string audio_topic = 10;
-}
-
-// Provides "hints" to the speech recognizer to favor specific words and phrases
-// in the results.
-message SpeechContext {
-  // A list of strings containing words and phrases "hints" so that
-  // the speech recognition is more likely to recognize them. This can be used
-  // to improve the accuracy for specific words and phrases, for example, if
-  // specific commands are typically spoken by the user. This can also be used
-  // to add additional words to the vocabulary of the recognizer. See
-  // [usage limits](https://cloud.google.com/speech-to-text/quotas#content).
-  //
-  // List items can also be set to classes for groups of words that represent
-  // common concepts that occur in natural language. For example, rather than
-  // providing phrase hints for every month of the year, using the $MONTH class
-  // improves the likelihood of correctly transcribing audio that includes
-  // months.
-  repeated string phrases = 1;
-
-  // Hint Boost. Positive value will increase the probability that a specific
-  // phrase will be recognized over other similar sounding phrases. The higher
-  // the boost, the higher the chance of false positive recognition as well.
-  // Negative boost values would correspond to anti-biasing. Anti-biasing is not
-  // enabled, so negative boost will simply be ignored. Though `boost` can
-  // accept a wide range of positive values, most use cases are best served with
-  // values between 0 and 20. We recommend using a binary search approach to
-  // finding the optimal value for your use case.
-  float boost = 4;
-}
-
-// Contains audio data in the encoding specified in the `RecognitionConfig`.
-// Either `content` or `uri` must be supplied. Supplying both or neither
-// returns [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT]. See
-// [content limits](https://cloud.google.com/speech-to-text/quotas#content).
-message RecognitionAudio {
-  // The audio source, which is either inline content or a Google Cloud
-  // Storage uri.
-  oneof audio_source {
-    // The audio data bytes encoded as specified in
-    // `RecognitionConfig`. Note: as with all bytes fields, proto buffers use a
-    // pure binary representation, whereas JSON representations use base64.
-    bytes content = 1;
-
-    // URI that points to a file that contains audio data bytes as specified in
-    // `RecognitionConfig`. The file must not be compressed (for example, gzip).
-    // Currently, only Google Cloud Storage URIs are
-    // supported, which must be specified in the following format:
-    // `gs://bucket_name/object_name` (other URI formats return
-    // [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT]). For more information, see
-    // [Request URIs](https://cloud.google.com/storage/docs/reference-uris).
-    string uri = 2;
-  }
-}
-
-// The only message returned to the client by the `Recognize` method. It
-// contains the result as zero or more sequential `SpeechRecognitionResult`
-// messages.
-message RecognizeResponse {
-  // Sequential list of transcription results corresponding to
-  // sequential portions of audio.
-  repeated SpeechRecognitionResult results = 2;
-}
-
-// The only message returned to the client by the `LongRunningRecognize` method.
-// It contains the result as zero or more sequential `SpeechRecognitionResult`
-// messages. It is included in the `result.response` field of the `Operation`
-// returned by the `GetOperation` call of the `google::longrunning::Operations`
-// service.
-message LongRunningRecognizeResponse {
-  // Sequential list of transcription results corresponding to
-  // sequential portions of audio.
-  repeated SpeechRecognitionResult results = 2;
-}
-
-// Describes the progress of a long-running `LongRunningRecognize` call. It is
-// included in the `metadata` field of the `Operation` returned by the
-// `GetOperation` call of the `google::longrunning::Operations` service.
-message LongRunningRecognizeMetadata {
-  // Approximate percentage of audio processed thus far. Guaranteed to be 100
-  // when the audio is fully processed and the results are available.
-  int32 progress_percent = 1;
-
-  // Time when the request was received.
-  google.protobuf.Timestamp start_time = 2;
-
-  // Time of the most recent processing update.
-  google.protobuf.Timestamp last_update_time = 3;
-}
-
-// `StreamingRecognizeResponse` is the only message returned to the client by
-// `StreamingRecognize`. A series of zero or more `StreamingRecognizeResponse`
-// messages are streamed back to the client. If there is no recognizable
-// audio, and `single_utterance` is set to false, then no messages are streamed
-// back to the client.
-//
-// Here's an example of a series of ten `StreamingRecognizeResponse`s that might
-// be returned while processing audio:
-//
-// 1. results { alternatives { transcript: "tube" } stability: 0.01 }
-//
-// 2. results { alternatives { transcript: "to be a" } stability: 0.01 }
-//
-// 3. results { alternatives { transcript: "to be" } stability: 0.9 }
-//    results { alternatives { transcript: " or not to be" } stability: 0.01 }
-//
-// 4. results { alternatives { transcript: "to be or not to be"
-//                             confidence: 0.92 }
-//              alternatives { transcript: "to bee or not to bee" }
-//              is_final: true }
-//
-// 5. results { alternatives { transcript: " that's" } stability: 0.01 }
-//
-// 6. results { alternatives { transcript: " that is" } stability: 0.9 }
-//    results { alternatives { transcript: " the question" } stability: 0.01 }
-//
-// 7. results { alternatives { transcript: " that is the question"
-//                             confidence: 0.98 }
-//              alternatives { transcript: " that was the question" }
-//              is_final: true }
-//
-// Notes:
-//
-// - Only two of the above responses #4 and #7 contain final results; they are
-//   indicated by `is_final: true`. Concatenating these together generates the
-//   full transcript: "to be or not to be that is the question".
-//
-// - The others contain interim `results`. #3 and #6 contain two interim
-//   `results`: the first portion has a high stability and is less likely to
-//   change; the second portion has a low stability and is very likely to
-//   change. A UI designer might choose to show only high stability `results`.
-//
-// - The specific `stability` and `confidence` values shown above are only for
-//   illustrative purposes. Actual values may vary.
-//
-// - In each response, only one of these fields will be set:
-//     `error`,
-//     `speech_event_type`, or
-//     one or more (repeated) `results`.
-message StreamingRecognizeResponse {
-  // Indicates the type of speech event.
-  enum SpeechEventType {
-    // No speech event specified.
-    SPEECH_EVENT_UNSPECIFIED = 0;
-
-    // This event indicates that the server has detected the end of the user's
-    // speech utterance and expects no additional speech. Therefore, the server
-    // will not process additional audio (although it may subsequently return
-    // additional results). The client should stop sending additional audio
-    // data, half-close the gRPC connection, and wait for any additional results
-    // until the server closes the gRPC connection. This event is only sent if
-    // `single_utterance` was set to `true`, and is not used otherwise.
-    END_OF_SINGLE_UTTERANCE = 1;
-  }
-
-  // If set, returns a [google.rpc.Status][google.rpc.Status] message that
-  // specifies the error for the operation.
-  google.rpc.Status error = 1;
-
-  // This repeated list contains zero or more results that
-  // correspond to consecutive portions of the audio currently being processed.
-  // It contains zero or one `is_final=true` result (the newly settled portion),
-  // followed by zero or more `is_final=false` results (the interim results).
-  repeated StreamingRecognitionResult results = 2;
-
-  // Indicates the type of speech event.
-  SpeechEventType speech_event_type = 4;
-}
-
-// A streaming speech recognition result corresponding to a portion of the audio
-// that is currently being processed.
-message StreamingRecognitionResult {
-  // May contain one or more recognition hypotheses (up to the
-  // maximum specified in `max_alternatives`).
-  // These alternatives are ordered in terms of accuracy, with the top (first)
-  // alternative being the most probable, as ranked by the recognizer.
-  repeated SpeechRecognitionAlternative alternatives = 1;
-
-  // If `false`, this `StreamingRecognitionResult` represents an
-  // interim result that may change. If `true`, this is the final time the
-  // speech service will return this particular `StreamingRecognitionResult`,
-  // the recognizer will not return any further hypotheses for this portion of
-  // the transcript and corresponding audio.
-  bool is_final = 2;
-
-  // An estimate of the likelihood that the recognizer will not
-  // change its guess about this interim result. Values range from 0.0
-  // (completely unstable) to 1.0 (completely stable).
-  // This field is only provided for interim results (`is_final=false`).
-  // The default of 0.0 is a sentinel value indicating `stability` was not set.
-  float stability = 3;
-
-  // Time offset of the end of this result relative to the
-  // beginning of the audio.
-  google.protobuf.Duration result_end_time = 4;
-
-  // For multi-channel audio, this is the channel number corresponding to the
-  // recognized result for the audio from that channel.
-  // For audio_channel_count = N, its output values can range from '1' to 'N'.
-  int32 channel_tag = 5;
-
-  // The [BCP-47](https://www.rfc-editor.org/rfc/bcp/bcp47.txt) language tag
-  // of the language in this result. This language code was detected to have
-  // the most likelihood of being spoken in the audio.
-  string language_code = 6;
-}
-
-// A speech recognition result corresponding to a portion of the audio.
-message SpeechRecognitionResult {
-  // May contain one or more recognition hypotheses (up to the
-  // maximum specified in `max_alternatives`).
-  // These alternatives are ordered in terms of accuracy, with the top (first)
-  // alternative being the most probable, as ranked by the recognizer.
-  repeated SpeechRecognitionAlternative alternatives = 1;
-
-  // For multi-channel audio, this is the channel number corresponding to the
-  // recognized result for the audio from that channel.
-  // For audio_channel_count = N, its output values can range from '1' to 'N'.
-  int32 channel_tag = 2;
-
-  // The [BCP-47](https://www.rfc-editor.org/rfc/bcp/bcp47.txt) language tag
-  // of the language in this result. This language code was detected to have
-  // the most likelihood of being spoken in the audio.
-  string language_code = 5;
-}
-
-// Alternative hypotheses (a.k.a. n-best list).
-message SpeechRecognitionAlternative {
-  // Transcript text representing the words that the user spoke.
-  string transcript = 1;
-
-  // The confidence estimate between 0.0 and 1.0. A higher number
-  // indicates an estimated greater likelihood that the recognized words are
-  // correct. This field is set only for the top alternative of a non-streaming
-  // result or, of a streaming result where `is_final=true`.
-  // This field is not guaranteed to be accurate and users should not rely on it
-  // to be always provided.
-  // The default of 0.0 is a sentinel value indicating `confidence` was not set.
-  float confidence = 2;
-
-  // A list of word-specific information for each recognized word.
-  // Note: When `enable_speaker_diarization` is true, you will see all the words
-  // from the beginning of the audio.
-  repeated WordInfo words = 3;
-}
-
-// Word-specific information for recognized words.
-message WordInfo {
-  // Time offset relative to the beginning of the audio,
-  // and corresponding to the start of the spoken word.
-  // This field is only set if `enable_word_time_offsets=true` and only
-  // in the top hypothesis.
-  // This is an experimental feature and the accuracy of the time offset can
-  // vary.
-  google.protobuf.Duration start_time = 1;
-
-  // Time offset relative to the beginning of the audio,
-  // and corresponding to the end of the spoken word.
-  // This field is only set if `enable_word_time_offsets=true` and only
-  // in the top hypothesis.
-  // This is an experimental feature and the accuracy of the time offset can
-  // vary.
-  google.protobuf.Duration end_time = 2;
-
-  // The word corresponding to this set of information.
-  string word = 3;
-
-  // The confidence estimate between 0.0 and 1.0. A higher number
-  // indicates an estimated greater likelihood that the recognized words are
-  // correct. This field is set only for the top alternative of a non-streaming
-  // result or, of a streaming result where `is_final=true`.
-  // This field is not guaranteed to be accurate and users should not rely on it
-  // to be always provided.
-  // The default of 0.0 is a sentinel value indicating `confidence` was not set.
-  float confidence = 4;
-
-  // A distinct integer value is assigned for every speaker within
-  // the audio. This field specifies which one of those speakers was detected to
-  // have spoken this word. Value ranges from '1' to diarization_speaker_count.
-  // speaker_tag is set if enable_speaker_diarization = 'true' and only in the
-  // top alternative.
-  int32 speaker_tag = 5;
-}
diff --git a/google/cloud/speech_v1p1beta1/proto/cloud_speech_pb2.py b/google/cloud/speech_v1p1beta1/proto/cloud_speech_pb2.py
deleted file mode 100644
index 1fd8de4f..00000000
--- a/google/cloud/speech_v1p1beta1/proto/cloud_speech_pb2.py
+++ /dev/null
@@ -1,2823 +0,0 @@
-# -*- coding: utf-8 -*-
-# Generated by the protocol buffer compiler.  DO NOT EDIT!
-# source: google/cloud/speech_v1p1beta1/proto/cloud_speech.proto
-
-import sys
-
-_b = sys.version_info[0] < 3 and (lambda x: x) or (lambda x: x.encode("latin1"))
-from google.protobuf import descriptor as _descriptor
-from google.protobuf import message as _message
-from google.protobuf import reflection as _reflection
-from google.protobuf import symbol_database as _symbol_database
-
-# @@protoc_insertion_point(imports)
-
-_sym_db = _symbol_database.Default()
-
-
-from google.api import annotations_pb2 as google_dot_api_dot_annotations__pb2
-from google.api import client_pb2 as google_dot_api_dot_client__pb2
-from google.api import field_behavior_pb2 as google_dot_api_dot_field__behavior__pb2
-from google.longrunning import (
-    operations_pb2 as google_dot_longrunning_dot_operations__pb2,
-)
-from google.protobuf import any_pb2 as google_dot_protobuf_dot_any__pb2
-from google.protobuf import duration_pb2 as google_dot_protobuf_dot_duration__pb2
-from google.protobuf import timestamp_pb2 as google_dot_protobuf_dot_timestamp__pb2
-from google.rpc import status_pb2 as google_dot_rpc_dot_status__pb2
-
-
-DESCRIPTOR = _descriptor.FileDescriptor(
-    name="google/cloud/speech_v1p1beta1/proto/cloud_speech.proto",
-    package="google.cloud.speech.v1p1beta1",
-    syntax="proto3",
-    serialized_options=_b(
-        "\n!com.google.cloud.speech.v1p1beta1B\013SpeechProtoP\001ZCgoogle.golang.org/genproto/googleapis/cloud/speech/v1p1beta1;speech\370\001\001\242\002\003GCS"
-    ),
-    serialized_pb=_b(
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-    ),
-    dependencies=[
-        google_dot_api_dot_annotations__pb2.DESCRIPTOR,
-        google_dot_api_dot_client__pb2.DESCRIPTOR,
-        google_dot_api_dot_field__behavior__pb2.DESCRIPTOR,
-        google_dot_longrunning_dot_operations__pb2.DESCRIPTOR,
-        google_dot_protobuf_dot_any__pb2.DESCRIPTOR,
-        google_dot_protobuf_dot_duration__pb2.DESCRIPTOR,
-        google_dot_protobuf_dot_timestamp__pb2.DESCRIPTOR,
-        google_dot_rpc_dot_status__pb2.DESCRIPTOR,
-    ],
-)
-
-
-_RECOGNITIONCONFIG_AUDIOENCODING = _descriptor.EnumDescriptor(
-    name="AudioEncoding",
-    full_name="google.cloud.speech.v1p1beta1.RecognitionConfig.AudioEncoding",
-    filename=None,
-    file=DESCRIPTOR,
-    values=[
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-            name="ENCODING_UNSPECIFIED",
-            index=0,
-            number=0,
-            serialized_options=None,
-            type=None,
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="LINEAR16", index=1, number=1, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="FLAC", index=2, number=2, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="MULAW", index=3, number=3, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="AMR", index=4, number=4, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="AMR_WB", index=5, number=5, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="OGG_OPUS", index=6, number=6, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="SPEEX_WITH_HEADER_BYTE",
-            index=7,
-            number=7,
-            serialized_options=None,
-            type=None,
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="MP3", index=8, number=8, serialized_options=None, type=None
-        ),
-    ],
-    containing_type=None,
-    serialized_options=None,
-    serialized_start=1752,
-    serialized_end=1900,
-)
-_sym_db.RegisterEnumDescriptor(_RECOGNITIONCONFIG_AUDIOENCODING)
-
-_RECOGNITIONMETADATA_INTERACTIONTYPE = _descriptor.EnumDescriptor(
-    name="InteractionType",
-    full_name="google.cloud.speech.v1p1beta1.RecognitionMetadata.InteractionType",
-    filename=None,
-    file=DESCRIPTOR,
-    values=[
-        _descriptor.EnumValueDescriptor(
-            name="INTERACTION_TYPE_UNSPECIFIED",
-            index=0,
-            number=0,
-            serialized_options=None,
-            type=None,
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="DISCUSSION", index=1, number=1, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="PRESENTATION", index=2, number=2, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="PHONE_CALL", index=3, number=3, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="VOICEMAIL", index=4, number=4, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="PROFESSIONALLY_PRODUCED",
-            index=5,
-            number=5,
-            serialized_options=None,
-            type=None,
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="VOICE_SEARCH", index=6, number=6, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="VOICE_COMMAND", index=7, number=7, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="DICTATION", index=8, number=8, serialized_options=None, type=None
-        ),
-    ],
-    containing_type=None,
-    serialized_options=None,
-    serialized_start=2586,
-    serialized_end=2783,
-)
-_sym_db.RegisterEnumDescriptor(_RECOGNITIONMETADATA_INTERACTIONTYPE)
-
-_RECOGNITIONMETADATA_MICROPHONEDISTANCE = _descriptor.EnumDescriptor(
-    name="MicrophoneDistance",
-    full_name="google.cloud.speech.v1p1beta1.RecognitionMetadata.MicrophoneDistance",
-    filename=None,
-    file=DESCRIPTOR,
-    values=[
-        _descriptor.EnumValueDescriptor(
-            name="MICROPHONE_DISTANCE_UNSPECIFIED",
-            index=0,
-            number=0,
-            serialized_options=None,
-            type=None,
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="NEARFIELD", index=1, number=1, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="MIDFIELD", index=2, number=2, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="FARFIELD", index=3, number=3, serialized_options=None, type=None
-        ),
-    ],
-    containing_type=None,
-    serialized_options=None,
-    serialized_start=2785,
-    serialized_end=2885,
-)
-_sym_db.RegisterEnumDescriptor(_RECOGNITIONMETADATA_MICROPHONEDISTANCE)
-
-_RECOGNITIONMETADATA_ORIGINALMEDIATYPE = _descriptor.EnumDescriptor(
-    name="OriginalMediaType",
-    full_name="google.cloud.speech.v1p1beta1.RecognitionMetadata.OriginalMediaType",
-    filename=None,
-    file=DESCRIPTOR,
-    values=[
-        _descriptor.EnumValueDescriptor(
-            name="ORIGINAL_MEDIA_TYPE_UNSPECIFIED",
-            index=0,
-            number=0,
-            serialized_options=None,
-            type=None,
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="AUDIO", index=1, number=1, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="VIDEO", index=2, number=2, serialized_options=None, type=None
-        ),
-    ],
-    containing_type=None,
-    serialized_options=None,
-    serialized_start=2887,
-    serialized_end=2965,
-)
-_sym_db.RegisterEnumDescriptor(_RECOGNITIONMETADATA_ORIGINALMEDIATYPE)
-
-_RECOGNITIONMETADATA_RECORDINGDEVICETYPE = _descriptor.EnumDescriptor(
-    name="RecordingDeviceType",
-    full_name="google.cloud.speech.v1p1beta1.RecognitionMetadata.RecordingDeviceType",
-    filename=None,
-    file=DESCRIPTOR,
-    values=[
-        _descriptor.EnumValueDescriptor(
-            name="RECORDING_DEVICE_TYPE_UNSPECIFIED",
-            index=0,
-            number=0,
-            serialized_options=None,
-            type=None,
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="SMARTPHONE", index=1, number=1, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="PC", index=2, number=2, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="PHONE_LINE", index=3, number=3, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="VEHICLE", index=4, number=4, serialized_options=None, type=None
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="OTHER_OUTDOOR_DEVICE",
-            index=5,
-            number=5,
-            serialized_options=None,
-            type=None,
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="OTHER_INDOOR_DEVICE",
-            index=6,
-            number=6,
-            serialized_options=None,
-            type=None,
-        ),
-    ],
-    containing_type=None,
-    serialized_options=None,
-    serialized_start=2968,
-    serialized_end=3132,
-)
-_sym_db.RegisterEnumDescriptor(_RECOGNITIONMETADATA_RECORDINGDEVICETYPE)
-
-_STREAMINGRECOGNIZERESPONSE_SPEECHEVENTTYPE = _descriptor.EnumDescriptor(
-    name="SpeechEventType",
-    full_name="google.cloud.speech.v1p1beta1.StreamingRecognizeResponse.SpeechEventType",
-    filename=None,
-    file=DESCRIPTOR,
-    values=[
-        _descriptor.EnumValueDescriptor(
-            name="SPEECH_EVENT_UNSPECIFIED",
-            index=0,
-            number=0,
-            serialized_options=None,
-            type=None,
-        ),
-        _descriptor.EnumValueDescriptor(
-            name="END_OF_SINGLE_UTTERANCE",
-            index=1,
-            number=1,
-            serialized_options=None,
-            type=None,
-        ),
-    ],
-    containing_type=None,
-    serialized_options=None,
-    serialized_start=3857,
-    serialized_end=3933,
-)
-_sym_db.RegisterEnumDescriptor(_STREAMINGRECOGNIZERESPONSE_SPEECHEVENTTYPE)
-
-
-_RECOGNIZEREQUEST = _descriptor.Descriptor(
-    name="RecognizeRequest",
-    full_name="google.cloud.speech.v1p1beta1.RecognizeRequest",
-    filename=None,
-    file=DESCRIPTOR,
-    containing_type=None,
-    fields=[
-        _descriptor.FieldDescriptor(
-            name="config",
-            full_name="google.cloud.speech.v1p1beta1.RecognizeRequest.config",
-            index=0,
-            number=1,
-            type=11,
-            cpp_type=10,
-            label=1,
-            has_default_value=False,
-            default_value=None,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=_b("\340A\002"),
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="audio",
-            full_name="google.cloud.speech.v1p1beta1.RecognizeRequest.audio",
-            index=1,
-            number=2,
-            type=11,
-            cpp_type=10,
-            label=1,
-            has_default_value=False,
-            default_value=None,
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-            extension_scope=None,
-            serialized_options=_b("\340A\002"),
-            file=DESCRIPTOR,
-        ),
-    ],
-    extensions=[],
-    nested_types=[],
-    enum_types=[],
-    serialized_options=None,
-    is_extendable=False,
-    syntax="proto3",
-    extension_ranges=[],
-    oneofs=[],
-    serialized_start=332,
-    serialized_end=490,
-)
-
-
-_LONGRUNNINGRECOGNIZEREQUEST = _descriptor.Descriptor(
-    name="LongRunningRecognizeRequest",
-    full_name="google.cloud.speech.v1p1beta1.LongRunningRecognizeRequest",
-    filename=None,
-    file=DESCRIPTOR,
-    containing_type=None,
-    fields=[
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-            name="config",
-            full_name="google.cloud.speech.v1p1beta1.LongRunningRecognizeRequest.config",
-            index=0,
-            number=1,
-            type=11,
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-            label=1,
-            has_default_value=False,
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-            extension_scope=None,
-            serialized_options=_b("\340A\002"),
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="audio",
-            full_name="google.cloud.speech.v1p1beta1.LongRunningRecognizeRequest.audio",
-            index=1,
-            number=2,
-            type=11,
-            cpp_type=10,
-            label=1,
-            has_default_value=False,
-            default_value=None,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=_b("\340A\002"),
-            file=DESCRIPTOR,
-        ),
-    ],
-    extensions=[],
-    nested_types=[],
-    enum_types=[],
-    serialized_options=None,
-    is_extendable=False,
-    syntax="proto3",
-    extension_ranges=[],
-    oneofs=[],
-    serialized_start=493,
-    serialized_end=662,
-)
-
-
-_STREAMINGRECOGNIZEREQUEST = _descriptor.Descriptor(
-    name="StreamingRecognizeRequest",
-    full_name="google.cloud.speech.v1p1beta1.StreamingRecognizeRequest",
-    filename=None,
-    file=DESCRIPTOR,
-    containing_type=None,
-    fields=[
-        _descriptor.FieldDescriptor(
-            name="streaming_config",
-            full_name="google.cloud.speech.v1p1beta1.StreamingRecognizeRequest.streaming_config",
-            index=0,
-            number=1,
-            type=11,
-            cpp_type=10,
-            label=1,
-            has_default_value=False,
-            default_value=None,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="audio_content",
-            full_name="google.cloud.speech.v1p1beta1.StreamingRecognizeRequest.audio_content",
-            index=1,
-            number=2,
-            type=12,
-            cpp_type=9,
-            label=1,
-            has_default_value=False,
-            default_value=_b(""),
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-    ],
-    extensions=[],
-    nested_types=[],
-    enum_types=[],
-    serialized_options=None,
-    is_extendable=False,
-    syntax="proto3",
-    extension_ranges=[],
-    oneofs=[
-        _descriptor.OneofDescriptor(
-            name="streaming_request",
-            full_name="google.cloud.speech.v1p1beta1.StreamingRecognizeRequest.streaming_request",
-            index=0,
-            containing_type=None,
-            fields=[],
-        )
-    ],
-    serialized_start=665,
-    serialized_end=825,
-)
-
-
-_STREAMINGRECOGNITIONCONFIG = _descriptor.Descriptor(
-    name="StreamingRecognitionConfig",
-    full_name="google.cloud.speech.v1p1beta1.StreamingRecognitionConfig",
-    filename=None,
-    file=DESCRIPTOR,
-    containing_type=None,
-    fields=[
-        _descriptor.FieldDescriptor(
-            name="config",
-            full_name="google.cloud.speech.v1p1beta1.StreamingRecognitionConfig.config",
-            index=0,
-            number=1,
-            type=11,
-            cpp_type=10,
-            label=1,
-            has_default_value=False,
-            default_value=None,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=_b("\340A\002"),
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="single_utterance",
-            full_name="google.cloud.speech.v1p1beta1.StreamingRecognitionConfig.single_utterance",
-            index=1,
-            number=2,
-            type=8,
-            cpp_type=7,
-            label=1,
-            has_default_value=False,
-            default_value=False,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="interim_results",
-            full_name="google.cloud.speech.v1p1beta1.StreamingRecognitionConfig.interim_results",
-            index=2,
-            number=3,
-            type=8,
-            cpp_type=7,
-            label=1,
-            has_default_value=False,
-            default_value=False,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-    ],
-    extensions=[],
-    nested_types=[],
-    enum_types=[],
-    serialized_options=None,
-    is_extendable=False,
-    syntax="proto3",
-    extension_ranges=[],
-    oneofs=[],
-    serialized_start=828,
-    serialized_end=978,
-)
-
-
-_RECOGNITIONCONFIG = _descriptor.Descriptor(
-    name="RecognitionConfig",
-    full_name="google.cloud.speech.v1p1beta1.RecognitionConfig",
-    filename=None,
-    file=DESCRIPTOR,
-    containing_type=None,
-    fields=[
-        _descriptor.FieldDescriptor(
-            name="encoding",
-            full_name="google.cloud.speech.v1p1beta1.RecognitionConfig.encoding",
-            index=0,
-            number=1,
-            type=14,
-            cpp_type=8,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="sample_rate_hertz",
-            full_name="google.cloud.speech.v1p1beta1.RecognitionConfig.sample_rate_hertz",
-            index=1,
-            number=2,
-            type=5,
-            cpp_type=1,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="audio_channel_count",
-            full_name="google.cloud.speech.v1p1beta1.RecognitionConfig.audio_channel_count",
-            index=2,
-            number=7,
-            type=5,
-            cpp_type=1,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="enable_separate_recognition_per_channel",
-            full_name="google.cloud.speech.v1p1beta1.RecognitionConfig.enable_separate_recognition_per_channel",
-            index=3,
-            number=12,
-            type=8,
-            cpp_type=7,
-            label=1,
-            has_default_value=False,
-            default_value=False,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="language_code",
-            full_name="google.cloud.speech.v1p1beta1.RecognitionConfig.language_code",
-            index=4,
-            number=3,
-            type=9,
-            cpp_type=9,
-            label=1,
-            has_default_value=False,
-            default_value=_b("").decode("utf-8"),
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=_b("\340A\002"),
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="alternative_language_codes",
-            full_name="google.cloud.speech.v1p1beta1.RecognitionConfig.alternative_language_codes",
-            index=5,
-            number=18,
-            type=9,
-            cpp_type=9,
-            label=3,
-            has_default_value=False,
-            default_value=[],
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="max_alternatives",
-            full_name="google.cloud.speech.v1p1beta1.RecognitionConfig.max_alternatives",
-            index=6,
-            number=4,
-            type=5,
-            cpp_type=1,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="profanity_filter",
-            full_name="google.cloud.speech.v1p1beta1.RecognitionConfig.profanity_filter",
-            index=7,
-            number=5,
-            type=8,
-            cpp_type=7,
-            label=1,
-            has_default_value=False,
-            default_value=False,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="speech_contexts",
-            full_name="google.cloud.speech.v1p1beta1.RecognitionConfig.speech_contexts",
-            index=8,
-            number=6,
-            type=11,
-            cpp_type=10,
-            label=3,
-            has_default_value=False,
-            default_value=[],
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="enable_word_time_offsets",
-            full_name="google.cloud.speech.v1p1beta1.RecognitionConfig.enable_word_time_offsets",
-            index=9,
-            number=8,
-            type=8,
-            cpp_type=7,
-            label=1,
-            has_default_value=False,
-            default_value=False,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="enable_word_confidence",
-            full_name="google.cloud.speech.v1p1beta1.RecognitionConfig.enable_word_confidence",
-            index=10,
-            number=15,
-            type=8,
-            cpp_type=7,
-            label=1,
-            has_default_value=False,
-            default_value=False,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="enable_automatic_punctuation",
-            full_name="google.cloud.speech.v1p1beta1.RecognitionConfig.enable_automatic_punctuation",
-            index=11,
-            number=11,
-            type=8,
-            cpp_type=7,
-            label=1,
-            has_default_value=False,
-            default_value=False,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="enable_speaker_diarization",
-            full_name="google.cloud.speech.v1p1beta1.RecognitionConfig.enable_speaker_diarization",
-            index=12,
-            number=16,
-            type=8,
-            cpp_type=7,
-            label=1,
-            has_default_value=False,
-            default_value=False,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=_b("\030\001"),
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="diarization_speaker_count",
-            full_name="google.cloud.speech.v1p1beta1.RecognitionConfig.diarization_speaker_count",
-            index=13,
-            number=17,
-            type=5,
-            cpp_type=1,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=_b("\030\001"),
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="diarization_config",
-            full_name="google.cloud.speech.v1p1beta1.RecognitionConfig.diarization_config",
-            index=14,
-            number=19,
-            type=11,
-            cpp_type=10,
-            label=1,
-            has_default_value=False,
-            default_value=None,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="metadata",
-            full_name="google.cloud.speech.v1p1beta1.RecognitionConfig.metadata",
-            index=15,
-            number=9,
-            type=11,
-            cpp_type=10,
-            label=1,
-            has_default_value=False,
-            default_value=None,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="model",
-            full_name="google.cloud.speech.v1p1beta1.RecognitionConfig.model",
-            index=16,
-            number=13,
-            type=9,
-            cpp_type=9,
-            label=1,
-            has_default_value=False,
-            default_value=_b("").decode("utf-8"),
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="use_enhanced",
-            full_name="google.cloud.speech.v1p1beta1.RecognitionConfig.use_enhanced",
-            index=17,
-            number=14,
-            type=8,
-            cpp_type=7,
-            label=1,
-            has_default_value=False,
-            default_value=False,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-    ],
-    extensions=[],
-    nested_types=[],
-    enum_types=[_RECOGNITIONCONFIG_AUDIOENCODING],
-    serialized_options=None,
-    is_extendable=False,
-    syntax="proto3",
-    extension_ranges=[],
-    oneofs=[],
-    serialized_start=981,
-    serialized_end=1900,
-)
-
-
-_SPEAKERDIARIZATIONCONFIG = _descriptor.Descriptor(
-    name="SpeakerDiarizationConfig",
-    full_name="google.cloud.speech.v1p1beta1.SpeakerDiarizationConfig",
-    filename=None,
-    file=DESCRIPTOR,
-    containing_type=None,
-    fields=[
-        _descriptor.FieldDescriptor(
-            name="enable_speaker_diarization",
-            full_name="google.cloud.speech.v1p1beta1.SpeakerDiarizationConfig.enable_speaker_diarization",
-            index=0,
-            number=1,
-            type=8,
-            cpp_type=7,
-            label=1,
-            has_default_value=False,
-            default_value=False,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="min_speaker_count",
-            full_name="google.cloud.speech.v1p1beta1.SpeakerDiarizationConfig.min_speaker_count",
-            index=1,
-            number=2,
-            type=5,
-            cpp_type=1,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="max_speaker_count",
-            full_name="google.cloud.speech.v1p1beta1.SpeakerDiarizationConfig.max_speaker_count",
-            index=2,
-            number=3,
-            type=5,
-            cpp_type=1,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-    ],
-    extensions=[],
-    nested_types=[],
-    enum_types=[],
-    serialized_options=None,
-    is_extendable=False,
-    syntax="proto3",
-    extension_ranges=[],
-    oneofs=[],
-    serialized_start=1902,
-    serialized_end=2018,
-)
-
-
-_RECOGNITIONMETADATA = _descriptor.Descriptor(
-    name="RecognitionMetadata",
-    full_name="google.cloud.speech.v1p1beta1.RecognitionMetadata",
-    filename=None,
-    file=DESCRIPTOR,
-    containing_type=None,
-    fields=[
-        _descriptor.FieldDescriptor(
-            name="interaction_type",
-            full_name="google.cloud.speech.v1p1beta1.RecognitionMetadata.interaction_type",
-            index=0,
-            number=1,
-            type=14,
-            cpp_type=8,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="industry_naics_code_of_audio",
-            full_name="google.cloud.speech.v1p1beta1.RecognitionMetadata.industry_naics_code_of_audio",
-            index=1,
-            number=3,
-            type=13,
-            cpp_type=3,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="microphone_distance",
-            full_name="google.cloud.speech.v1p1beta1.RecognitionMetadata.microphone_distance",
-            index=2,
-            number=4,
-            type=14,
-            cpp_type=8,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="original_media_type",
-            full_name="google.cloud.speech.v1p1beta1.RecognitionMetadata.original_media_type",
-            index=3,
-            number=5,
-            type=14,
-            cpp_type=8,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="recording_device_type",
-            full_name="google.cloud.speech.v1p1beta1.RecognitionMetadata.recording_device_type",
-            index=4,
-            number=6,
-            type=14,
-            cpp_type=8,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="recording_device_name",
-            full_name="google.cloud.speech.v1p1beta1.RecognitionMetadata.recording_device_name",
-            index=5,
-            number=7,
-            type=9,
-            cpp_type=9,
-            label=1,
-            has_default_value=False,
-            default_value=_b("").decode("utf-8"),
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="original_mime_type",
-            full_name="google.cloud.speech.v1p1beta1.RecognitionMetadata.original_mime_type",
-            index=6,
-            number=8,
-            type=9,
-            cpp_type=9,
-            label=1,
-            has_default_value=False,
-            default_value=_b("").decode("utf-8"),
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="obfuscated_id",
-            full_name="google.cloud.speech.v1p1beta1.RecognitionMetadata.obfuscated_id",
-            index=7,
-            number=9,
-            type=3,
-            cpp_type=2,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=_b("\030\001"),
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="audio_topic",
-            full_name="google.cloud.speech.v1p1beta1.RecognitionMetadata.audio_topic",
-            index=8,
-            number=10,
-            type=9,
-            cpp_type=9,
-            label=1,
-            has_default_value=False,
-            default_value=_b("").decode("utf-8"),
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-    ],
-    extensions=[],
-    nested_types=[],
-    enum_types=[
-        _RECOGNITIONMETADATA_INTERACTIONTYPE,
-        _RECOGNITIONMETADATA_MICROPHONEDISTANCE,
-        _RECOGNITIONMETADATA_ORIGINALMEDIATYPE,
-        _RECOGNITIONMETADATA_RECORDINGDEVICETYPE,
-    ],
-    serialized_options=None,
-    is_extendable=False,
-    syntax="proto3",
-    extension_ranges=[],
-    oneofs=[],
-    serialized_start=2021,
-    serialized_end=3132,
-)
-
-
-_SPEECHCONTEXT = _descriptor.Descriptor(
-    name="SpeechContext",
-    full_name="google.cloud.speech.v1p1beta1.SpeechContext",
-    filename=None,
-    file=DESCRIPTOR,
-    containing_type=None,
-    fields=[
-        _descriptor.FieldDescriptor(
-            name="phrases",
-            full_name="google.cloud.speech.v1p1beta1.SpeechContext.phrases",
-            index=0,
-            number=1,
-            type=9,
-            cpp_type=9,
-            label=3,
-            has_default_value=False,
-            default_value=[],
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="boost",
-            full_name="google.cloud.speech.v1p1beta1.SpeechContext.boost",
-            index=1,
-            number=4,
-            type=2,
-            cpp_type=6,
-            label=1,
-            has_default_value=False,
-            default_value=float(0),
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-    ],
-    extensions=[],
-    nested_types=[],
-    enum_types=[],
-    serialized_options=None,
-    is_extendable=False,
-    syntax="proto3",
-    extension_ranges=[],
-    oneofs=[],
-    serialized_start=3134,
-    serialized_end=3181,
-)
-
-
-_RECOGNITIONAUDIO = _descriptor.Descriptor(
-    name="RecognitionAudio",
-    full_name="google.cloud.speech.v1p1beta1.RecognitionAudio",
-    filename=None,
-    file=DESCRIPTOR,
-    containing_type=None,
-    fields=[
-        _descriptor.FieldDescriptor(
-            name="content",
-            full_name="google.cloud.speech.v1p1beta1.RecognitionAudio.content",
-            index=0,
-            number=1,
-            type=12,
-            cpp_type=9,
-            label=1,
-            has_default_value=False,
-            default_value=_b(""),
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="uri",
-            full_name="google.cloud.speech.v1p1beta1.RecognitionAudio.uri",
-            index=1,
-            number=2,
-            type=9,
-            cpp_type=9,
-            label=1,
-            has_default_value=False,
-            default_value=_b("").decode("utf-8"),
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-    ],
-    extensions=[],
-    nested_types=[],
-    enum_types=[],
-    serialized_options=None,
-    is_extendable=False,
-    syntax="proto3",
-    extension_ranges=[],
-    oneofs=[
-        _descriptor.OneofDescriptor(
-            name="audio_source",
-            full_name="google.cloud.speech.v1p1beta1.RecognitionAudio.audio_source",
-            index=0,
-            containing_type=None,
-            fields=[],
-        )
-    ],
-    serialized_start=3183,
-    serialized_end=3251,
-)
-
-
-_RECOGNIZERESPONSE = _descriptor.Descriptor(
-    name="RecognizeResponse",
-    full_name="google.cloud.speech.v1p1beta1.RecognizeResponse",
-    filename=None,
-    file=DESCRIPTOR,
-    containing_type=None,
-    fields=[
-        _descriptor.FieldDescriptor(
-            name="results",
-            full_name="google.cloud.speech.v1p1beta1.RecognizeResponse.results",
-            index=0,
-            number=2,
-            type=11,
-            cpp_type=10,
-            label=3,
-            has_default_value=False,
-            default_value=[],
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        )
-    ],
-    extensions=[],
-    nested_types=[],
-    enum_types=[],
-    serialized_options=None,
-    is_extendable=False,
-    syntax="proto3",
-    extension_ranges=[],
-    oneofs=[],
-    serialized_start=3253,
-    serialized_end=3345,
-)
-
-
-_LONGRUNNINGRECOGNIZERESPONSE = _descriptor.Descriptor(
-    name="LongRunningRecognizeResponse",
-    full_name="google.cloud.speech.v1p1beta1.LongRunningRecognizeResponse",
-    filename=None,
-    file=DESCRIPTOR,
-    containing_type=None,
-    fields=[
-        _descriptor.FieldDescriptor(
-            name="results",
-            full_name="google.cloud.speech.v1p1beta1.LongRunningRecognizeResponse.results",
-            index=0,
-            number=2,
-            type=11,
-            cpp_type=10,
-            label=3,
-            has_default_value=False,
-            default_value=[],
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        )
-    ],
-    extensions=[],
-    nested_types=[],
-    enum_types=[],
-    serialized_options=None,
-    is_extendable=False,
-    syntax="proto3",
-    extension_ranges=[],
-    oneofs=[],
-    serialized_start=3347,
-    serialized_end=3450,
-)
-
-
-_LONGRUNNINGRECOGNIZEMETADATA = _descriptor.Descriptor(
-    name="LongRunningRecognizeMetadata",
-    full_name="google.cloud.speech.v1p1beta1.LongRunningRecognizeMetadata",
-    filename=None,
-    file=DESCRIPTOR,
-    containing_type=None,
-    fields=[
-        _descriptor.FieldDescriptor(
-            name="progress_percent",
-            full_name="google.cloud.speech.v1p1beta1.LongRunningRecognizeMetadata.progress_percent",
-            index=0,
-            number=1,
-            type=5,
-            cpp_type=1,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="start_time",
-            full_name="google.cloud.speech.v1p1beta1.LongRunningRecognizeMetadata.start_time",
-            index=1,
-            number=2,
-            type=11,
-            cpp_type=10,
-            label=1,
-            has_default_value=False,
-            default_value=None,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="last_update_time",
-            full_name="google.cloud.speech.v1p1beta1.LongRunningRecognizeMetadata.last_update_time",
-            index=2,
-            number=3,
-            type=11,
-            cpp_type=10,
-            label=1,
-            has_default_value=False,
-            default_value=None,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-    ],
-    extensions=[],
-    nested_types=[],
-    enum_types=[],
-    serialized_options=None,
-    is_extendable=False,
-    syntax="proto3",
-    extension_ranges=[],
-    oneofs=[],
-    serialized_start=3453,
-    serialized_end=3611,
-)
-
-
-_STREAMINGRECOGNIZERESPONSE = _descriptor.Descriptor(
-    name="StreamingRecognizeResponse",
-    full_name="google.cloud.speech.v1p1beta1.StreamingRecognizeResponse",
-    filename=None,
-    file=DESCRIPTOR,
-    containing_type=None,
-    fields=[
-        _descriptor.FieldDescriptor(
-            name="error",
-            full_name="google.cloud.speech.v1p1beta1.StreamingRecognizeResponse.error",
-            index=0,
-            number=1,
-            type=11,
-            cpp_type=10,
-            label=1,
-            has_default_value=False,
-            default_value=None,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="results",
-            full_name="google.cloud.speech.v1p1beta1.StreamingRecognizeResponse.results",
-            index=1,
-            number=2,
-            type=11,
-            cpp_type=10,
-            label=3,
-            has_default_value=False,
-            default_value=[],
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="speech_event_type",
-            full_name="google.cloud.speech.v1p1beta1.StreamingRecognizeResponse.speech_event_type",
-            index=2,
-            number=4,
-            type=14,
-            cpp_type=8,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-    ],
-    extensions=[],
-    nested_types=[],
-    enum_types=[_STREAMINGRECOGNIZERESPONSE_SPEECHEVENTTYPE],
-    serialized_options=None,
-    is_extendable=False,
-    syntax="proto3",
-    extension_ranges=[],
-    oneofs=[],
-    serialized_start=3614,
-    serialized_end=3933,
-)
-
-
-_STREAMINGRECOGNITIONRESULT = _descriptor.Descriptor(
-    name="StreamingRecognitionResult",
-    full_name="google.cloud.speech.v1p1beta1.StreamingRecognitionResult",
-    filename=None,
-    file=DESCRIPTOR,
-    containing_type=None,
-    fields=[
-        _descriptor.FieldDescriptor(
-            name="alternatives",
-            full_name="google.cloud.speech.v1p1beta1.StreamingRecognitionResult.alternatives",
-            index=0,
-            number=1,
-            type=11,
-            cpp_type=10,
-            label=3,
-            has_default_value=False,
-            default_value=[],
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="is_final",
-            full_name="google.cloud.speech.v1p1beta1.StreamingRecognitionResult.is_final",
-            index=1,
-            number=2,
-            type=8,
-            cpp_type=7,
-            label=1,
-            has_default_value=False,
-            default_value=False,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="stability",
-            full_name="google.cloud.speech.v1p1beta1.StreamingRecognitionResult.stability",
-            index=2,
-            number=3,
-            type=2,
-            cpp_type=6,
-            label=1,
-            has_default_value=False,
-            default_value=float(0),
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="result_end_time",
-            full_name="google.cloud.speech.v1p1beta1.StreamingRecognitionResult.result_end_time",
-            index=3,
-            number=4,
-            type=11,
-            cpp_type=10,
-            label=1,
-            has_default_value=False,
-            default_value=None,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="channel_tag",
-            full_name="google.cloud.speech.v1p1beta1.StreamingRecognitionResult.channel_tag",
-            index=4,
-            number=5,
-            type=5,
-            cpp_type=1,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="language_code",
-            full_name="google.cloud.speech.v1p1beta1.StreamingRecognitionResult.language_code",
-            index=5,
-            number=6,
-            type=9,
-            cpp_type=9,
-            label=1,
-            has_default_value=False,
-            default_value=_b("").decode("utf-8"),
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-    ],
-    extensions=[],
-    nested_types=[],
-    enum_types=[],
-    serialized_options=None,
-    is_extendable=False,
-    syntax="proto3",
-    extension_ranges=[],
-    oneofs=[],
-    serialized_start=3936,
-    serialized_end=4180,
-)
-
-
-_SPEECHRECOGNITIONRESULT = _descriptor.Descriptor(
-    name="SpeechRecognitionResult",
-    full_name="google.cloud.speech.v1p1beta1.SpeechRecognitionResult",
-    filename=None,
-    file=DESCRIPTOR,
-    containing_type=None,
-    fields=[
-        _descriptor.FieldDescriptor(
-            name="alternatives",
-            full_name="google.cloud.speech.v1p1beta1.SpeechRecognitionResult.alternatives",
-            index=0,
-            number=1,
-            type=11,
-            cpp_type=10,
-            label=3,
-            has_default_value=False,
-            default_value=[],
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="channel_tag",
-            full_name="google.cloud.speech.v1p1beta1.SpeechRecognitionResult.channel_tag",
-            index=1,
-            number=2,
-            type=5,
-            cpp_type=1,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="language_code",
-            full_name="google.cloud.speech.v1p1beta1.SpeechRecognitionResult.language_code",
-            index=2,
-            number=5,
-            type=9,
-            cpp_type=9,
-            label=1,
-            has_default_value=False,
-            default_value=_b("").decode("utf-8"),
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-    ],
-    extensions=[],
-    nested_types=[],
-    enum_types=[],
-    serialized_options=None,
-    is_extendable=False,
-    syntax="proto3",
-    extension_ranges=[],
-    oneofs=[],
-    serialized_start=4183,
-    serialized_end=4335,
-)
-
-
-_SPEECHRECOGNITIONALTERNATIVE = _descriptor.Descriptor(
-    name="SpeechRecognitionAlternative",
-    full_name="google.cloud.speech.v1p1beta1.SpeechRecognitionAlternative",
-    filename=None,
-    file=DESCRIPTOR,
-    containing_type=None,
-    fields=[
-        _descriptor.FieldDescriptor(
-            name="transcript",
-            full_name="google.cloud.speech.v1p1beta1.SpeechRecognitionAlternative.transcript",
-            index=0,
-            number=1,
-            type=9,
-            cpp_type=9,
-            label=1,
-            has_default_value=False,
-            default_value=_b("").decode("utf-8"),
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="confidence",
-            full_name="google.cloud.speech.v1p1beta1.SpeechRecognitionAlternative.confidence",
-            index=1,
-            number=2,
-            type=2,
-            cpp_type=6,
-            label=1,
-            has_default_value=False,
-            default_value=float(0),
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="words",
-            full_name="google.cloud.speech.v1p1beta1.SpeechRecognitionAlternative.words",
-            index=2,
-            number=3,
-            type=11,
-            cpp_type=10,
-            label=3,
-            has_default_value=False,
-            default_value=[],
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-    ],
-    extensions=[],
-    nested_types=[],
-    enum_types=[],
-    serialized_options=None,
-    is_extendable=False,
-    syntax="proto3",
-    extension_ranges=[],
-    oneofs=[],
-    serialized_start=4337,
-    serialized_end=4463,
-)
-
-
-_WORDINFO = _descriptor.Descriptor(
-    name="WordInfo",
-    full_name="google.cloud.speech.v1p1beta1.WordInfo",
-    filename=None,
-    file=DESCRIPTOR,
-    containing_type=None,
-    fields=[
-        _descriptor.FieldDescriptor(
-            name="start_time",
-            full_name="google.cloud.speech.v1p1beta1.WordInfo.start_time",
-            index=0,
-            number=1,
-            type=11,
-            cpp_type=10,
-            label=1,
-            has_default_value=False,
-            default_value=None,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="end_time",
-            full_name="google.cloud.speech.v1p1beta1.WordInfo.end_time",
-            index=1,
-            number=2,
-            type=11,
-            cpp_type=10,
-            label=1,
-            has_default_value=False,
-            default_value=None,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="word",
-            full_name="google.cloud.speech.v1p1beta1.WordInfo.word",
-            index=2,
-            number=3,
-            type=9,
-            cpp_type=9,
-            label=1,
-            has_default_value=False,
-            default_value=_b("").decode("utf-8"),
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="confidence",
-            full_name="google.cloud.speech.v1p1beta1.WordInfo.confidence",
-            index=3,
-            number=4,
-            type=2,
-            cpp_type=6,
-            label=1,
-            has_default_value=False,
-            default_value=float(0),
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-        _descriptor.FieldDescriptor(
-            name="speaker_tag",
-            full_name="google.cloud.speech.v1p1beta1.WordInfo.speaker_tag",
-            index=4,
-            number=5,
-            type=5,
-            cpp_type=1,
-            label=1,
-            has_default_value=False,
-            default_value=0,
-            message_type=None,
-            enum_type=None,
-            containing_type=None,
-            is_extension=False,
-            extension_scope=None,
-            serialized_options=None,
-            file=DESCRIPTOR,
-        ),
-    ],
-    extensions=[],
-    nested_types=[],
-    enum_types=[],
-    serialized_options=None,
-    is_extendable=False,
-    syntax="proto3",
-    extension_ranges=[],
-    oneofs=[],
-    serialized_start=4466,
-    serialized_end=4623,
-)
-
-_RECOGNIZEREQUEST.fields_by_name["config"].message_type = _RECOGNITIONCONFIG
-_RECOGNIZEREQUEST.fields_by_name["audio"].message_type = _RECOGNITIONAUDIO
-_LONGRUNNINGRECOGNIZEREQUEST.fields_by_name["config"].message_type = _RECOGNITIONCONFIG
-_LONGRUNNINGRECOGNIZEREQUEST.fields_by_name["audio"].message_type = _RECOGNITIONAUDIO
-_STREAMINGRECOGNIZEREQUEST.fields_by_name[
-    "streaming_config"
-].message_type = _STREAMINGRECOGNITIONCONFIG
-_STREAMINGRECOGNIZEREQUEST.oneofs_by_name["streaming_request"].fields.append(
-    _STREAMINGRECOGNIZEREQUEST.fields_by_name["streaming_config"]
-)
-_STREAMINGRECOGNIZEREQUEST.fields_by_name[
-    "streaming_config"
-].containing_oneof = _STREAMINGRECOGNIZEREQUEST.oneofs_by_name["streaming_request"]
-_STREAMINGRECOGNIZEREQUEST.oneofs_by_name["streaming_request"].fields.append(
-    _STREAMINGRECOGNIZEREQUEST.fields_by_name["audio_content"]
-)
-_STREAMINGRECOGNIZEREQUEST.fields_by_name[
-    "audio_content"
-].containing_oneof = _STREAMINGRECOGNIZEREQUEST.oneofs_by_name["streaming_request"]
-_STREAMINGRECOGNITIONCONFIG.fields_by_name["config"].message_type = _RECOGNITIONCONFIG
-_RECOGNITIONCONFIG.fields_by_name[
-    "encoding"
-].enum_type = _RECOGNITIONCONFIG_AUDIOENCODING
-_RECOGNITIONCONFIG.fields_by_name["speech_contexts"].message_type = _SPEECHCONTEXT
-_RECOGNITIONCONFIG.fields_by_name[
-    "diarization_config"
-].message_type = _SPEAKERDIARIZATIONCONFIG
-_RECOGNITIONCONFIG.fields_by_name["metadata"].message_type = _RECOGNITIONMETADATA
-_RECOGNITIONCONFIG_AUDIOENCODING.containing_type = _RECOGNITIONCONFIG
-_RECOGNITIONMETADATA.fields_by_name[
-    "interaction_type"
-].enum_type = _RECOGNITIONMETADATA_INTERACTIONTYPE
-_RECOGNITIONMETADATA.fields_by_name[
-    "microphone_distance"
-].enum_type = _RECOGNITIONMETADATA_MICROPHONEDISTANCE
-_RECOGNITIONMETADATA.fields_by_name[
-    "original_media_type"
-].enum_type = _RECOGNITIONMETADATA_ORIGINALMEDIATYPE
-_RECOGNITIONMETADATA.fields_by_name[
-    "recording_device_type"
-].enum_type = _RECOGNITIONMETADATA_RECORDINGDEVICETYPE
-_RECOGNITIONMETADATA_INTERACTIONTYPE.containing_type = _RECOGNITIONMETADATA
-_RECOGNITIONMETADATA_MICROPHONEDISTANCE.containing_type = _RECOGNITIONMETADATA
-_RECOGNITIONMETADATA_ORIGINALMEDIATYPE.containing_type = _RECOGNITIONMETADATA
-_RECOGNITIONMETADATA_RECORDINGDEVICETYPE.containing_type = _RECOGNITIONMETADATA
-_RECOGNITIONAUDIO.oneofs_by_name["audio_source"].fields.append(
-    _RECOGNITIONAUDIO.fields_by_name["content"]
-)
-_RECOGNITIONAUDIO.fields_by_name[
-    "content"
-].containing_oneof = _RECOGNITIONAUDIO.oneofs_by_name["audio_source"]
-_RECOGNITIONAUDIO.oneofs_by_name["audio_source"].fields.append(
-    _RECOGNITIONAUDIO.fields_by_name["uri"]
-)
-_RECOGNITIONAUDIO.fields_by_name[
-    "uri"
-].containing_oneof = _RECOGNITIONAUDIO.oneofs_by_name["audio_source"]
-_RECOGNIZERESPONSE.fields_by_name["results"].message_type = _SPEECHRECOGNITIONRESULT
-_LONGRUNNINGRECOGNIZERESPONSE.fields_by_name[
-    "results"
-].message_type = _SPEECHRECOGNITIONRESULT
-_LONGRUNNINGRECOGNIZEMETADATA.fields_by_name[
-    "start_time"
-].message_type = google_dot_protobuf_dot_timestamp__pb2._TIMESTAMP
-_LONGRUNNINGRECOGNIZEMETADATA.fields_by_name[
-    "last_update_time"
-].message_type = google_dot_protobuf_dot_timestamp__pb2._TIMESTAMP
-_STREAMINGRECOGNIZERESPONSE.fields_by_name[
-    "error"
-].message_type = google_dot_rpc_dot_status__pb2._STATUS
-_STREAMINGRECOGNIZERESPONSE.fields_by_name[
-    "results"
-].message_type = _STREAMINGRECOGNITIONRESULT
-_STREAMINGRECOGNIZERESPONSE.fields_by_name[
-    "speech_event_type"
-].enum_type = _STREAMINGRECOGNIZERESPONSE_SPEECHEVENTTYPE
-_STREAMINGRECOGNIZERESPONSE_SPEECHEVENTTYPE.containing_type = (
-    _STREAMINGRECOGNIZERESPONSE
-)
-_STREAMINGRECOGNITIONRESULT.fields_by_name[
-    "alternatives"
-].message_type = _SPEECHRECOGNITIONALTERNATIVE
-_STREAMINGRECOGNITIONRESULT.fields_by_name[
-    "result_end_time"
-].message_type = google_dot_protobuf_dot_duration__pb2._DURATION
-_SPEECHRECOGNITIONRESULT.fields_by_name[
-    "alternatives"
-].message_type = _SPEECHRECOGNITIONALTERNATIVE
-_SPEECHRECOGNITIONALTERNATIVE.fields_by_name["words"].message_type = _WORDINFO
-_WORDINFO.fields_by_name[
-    "start_time"
-].message_type = google_dot_protobuf_dot_duration__pb2._DURATION
-_WORDINFO.fields_by_name[
-    "end_time"
-].message_type = google_dot_protobuf_dot_duration__pb2._DURATION
-DESCRIPTOR.message_types_by_name["RecognizeRequest"] = _RECOGNIZEREQUEST
-DESCRIPTOR.message_types_by_name[
-    "LongRunningRecognizeRequest"
-] = _LONGRUNNINGRECOGNIZEREQUEST
-DESCRIPTOR.message_types_by_name[
-    "StreamingRecognizeRequest"
-] = _STREAMINGRECOGNIZEREQUEST
-DESCRIPTOR.message_types_by_name[
-    "StreamingRecognitionConfig"
-] = _STREAMINGRECOGNITIONCONFIG
-DESCRIPTOR.message_types_by_name["RecognitionConfig"] = _RECOGNITIONCONFIG
-DESCRIPTOR.message_types_by_name["SpeakerDiarizationConfig"] = _SPEAKERDIARIZATIONCONFIG
-DESCRIPTOR.message_types_by_name["RecognitionMetadata"] = _RECOGNITIONMETADATA
-DESCRIPTOR.message_types_by_name["SpeechContext"] = _SPEECHCONTEXT
-DESCRIPTOR.message_types_by_name["RecognitionAudio"] = _RECOGNITIONAUDIO
-DESCRIPTOR.message_types_by_name["RecognizeResponse"] = _RECOGNIZERESPONSE
-DESCRIPTOR.message_types_by_name[
-    "LongRunningRecognizeResponse"
-] = _LONGRUNNINGRECOGNIZERESPONSE
-DESCRIPTOR.message_types_by_name[
-    "LongRunningRecognizeMetadata"
-] = _LONGRUNNINGRECOGNIZEMETADATA
-DESCRIPTOR.message_types_by_name[
-    "StreamingRecognizeResponse"
-] = _STREAMINGRECOGNIZERESPONSE
-DESCRIPTOR.message_types_by_name[
-    "StreamingRecognitionResult"
-] = _STREAMINGRECOGNITIONRESULT
-DESCRIPTOR.message_types_by_name["SpeechRecognitionResult"] = _SPEECHRECOGNITIONRESULT
-DESCRIPTOR.message_types_by_name[
-    "SpeechRecognitionAlternative"
-] = _SPEECHRECOGNITIONALTERNATIVE
-DESCRIPTOR.message_types_by_name["WordInfo"] = _WORDINFO
-_sym_db.RegisterFileDescriptor(DESCRIPTOR)
-
-RecognizeRequest = _reflection.GeneratedProtocolMessageType(
-    "RecognizeRequest",
-    (_message.Message,),
-    dict(
-        DESCRIPTOR=_RECOGNIZEREQUEST,
-        __module__="google.cloud.speech_v1p1beta1.proto.cloud_speech_pb2",
-        __doc__="""The top-level message sent by the client for the
-  ``Recognize`` method.
-  
-  
-  Attributes:
-      config:
-          Required. Provides information to the recognizer that
-          specifies how to process the request.
-      audio:
-          Required. The audio data to be recognized.
-  """,
-        # @@protoc_insertion_point(class_scope:google.cloud.speech.v1p1beta1.RecognizeRequest)
-    ),
-)
-_sym_db.RegisterMessage(RecognizeRequest)
-
-LongRunningRecognizeRequest = _reflection.GeneratedProtocolMessageType(
-    "LongRunningRecognizeRequest",
-    (_message.Message,),
-    dict(
-        DESCRIPTOR=_LONGRUNNINGRECOGNIZEREQUEST,
-        __module__="google.cloud.speech_v1p1beta1.proto.cloud_speech_pb2",
-        __doc__="""The top-level message sent by the client for the
-  ``LongRunningRecognize`` method.
-  
-  
-  Attributes:
-      config:
-          Required. Provides information to the recognizer that
-          specifies how to process the request.
-      audio:
-          Required. The audio data to be recognized.
-  """,
-        # @@protoc_insertion_point(class_scope:google.cloud.speech.v1p1beta1.LongRunningRecognizeRequest)
-    ),
-)
-_sym_db.RegisterMessage(LongRunningRecognizeRequest)
-
-StreamingRecognizeRequest = _reflection.GeneratedProtocolMessageType(
-    "StreamingRecognizeRequest",
-    (_message.Message,),
-    dict(
-        DESCRIPTOR=_STREAMINGRECOGNIZEREQUEST,
-        __module__="google.cloud.speech_v1p1beta1.proto.cloud_speech_pb2",
-        __doc__="""The top-level message sent by the client for the
-  ``StreamingRecognize`` method. Multiple ``StreamingRecognizeRequest``
-  messages are sent. The first message must contain a ``streaming_config``
-  message and must not contain ``audio_content``. All subsequent messages
-  must contain ``audio_content`` and must not contain a
-  ``streaming_config`` message.
-  
-  
-  Attributes:
-      streaming_request:
-          The streaming request, which is either a streaming config or
-          audio content.
-      streaming_config:
-          Provides information to the recognizer that specifies how to
-          process the request. The first ``StreamingRecognizeRequest``
-          message must contain a ``streaming_config`` message.
-      audio_content:
-          The audio data to be recognized. Sequential chunks of audio
-          data are sent in sequential ``StreamingRecognizeRequest``
-          messages. The first ``StreamingRecognizeRequest`` message must
-          not contain ``audio_content`` data and all subsequent
-          ``StreamingRecognizeRequest`` messages must contain
-          ``audio_content`` data. The audio bytes must be encoded as
-          specified in ``RecognitionConfig``. Note: as with all bytes
-          fields, proto buffers use a pure binary representation (not
-          base64). See `content limits <https://cloud.google.com/speech-
-          to-text/quotas#content>`__.
-  """,
-        # @@protoc_insertion_point(class_scope:google.cloud.speech.v1p1beta1.StreamingRecognizeRequest)
-    ),
-)
-_sym_db.RegisterMessage(StreamingRecognizeRequest)
-
-StreamingRecognitionConfig = _reflection.GeneratedProtocolMessageType(
-    "StreamingRecognitionConfig",
-    (_message.Message,),
-    dict(
-        DESCRIPTOR=_STREAMINGRECOGNITIONCONFIG,
-        __module__="google.cloud.speech_v1p1beta1.proto.cloud_speech_pb2",
-        __doc__="""Provides information to the recognizer that specifies how
-  to process the request.
-  
-  
-  Attributes:
-      config:
-          Required. Provides information to the recognizer that
-          specifies how to process the request.
-      single_utterance:
-          If ``false`` or omitted, the recognizer will perform
-          continuous recognition (continuing to wait for and process
-          audio even if the user pauses speaking) until the client
-          closes the input stream (gRPC API) or until the maximum time
-          limit has been reached. May return multiple
-          ``StreamingRecognitionResult``\ s with the ``is_final`` flag
-          set to ``true``.  If ``true``, the recognizer will detect a
-          single spoken utterance. When it detects that the user has
-          paused or stopped speaking, it will return an
-          ``END_OF_SINGLE_UTTERANCE`` event and cease recognition. It
-          will return no more than one ``StreamingRecognitionResult``
-          with the ``is_final`` flag set to ``true``.
-      interim_results:
-          If ``true``, interim results (tentative hypotheses) may be
-          returned as they become available (these interim results are
-          indicated with the ``is_final=false`` flag). If ``false`` or
-          omitted, only ``is_final=true`` result(s) are returned.
-  """,
-        # @@protoc_insertion_point(class_scope:google.cloud.speech.v1p1beta1.StreamingRecognitionConfig)
-    ),
-)
-_sym_db.RegisterMessage(StreamingRecognitionConfig)
-
-RecognitionConfig = _reflection.GeneratedProtocolMessageType(
-    "RecognitionConfig",
-    (_message.Message,),
-    dict(
-        DESCRIPTOR=_RECOGNITIONCONFIG,
-        __module__="google.cloud.speech_v1p1beta1.proto.cloud_speech_pb2",
-        __doc__="""Provides information to the recognizer that specifies how
-  to process the request.
-  
-  
-  Attributes:
-      encoding:
-          Encoding of audio data sent in all ``RecognitionAudio``
-          messages. This field is optional for ``FLAC`` and ``WAV``
-          audio files and required for all other audio formats. For
-          details, see [AudioEncoding][google.cloud.speech.v1p1beta1.Rec
-          ognitionConfig.AudioEncoding].
-      sample_rate_hertz:
-          Sample rate in Hertz of the audio data sent in all
-          ``RecognitionAudio`` messages. Valid values are: 8000-48000.
-          16000 is optimal. For best results, set the sampling rate of
-          the audio source to 16000 Hz. If that's not possible, use the
-          native sample rate of the audio source (instead of re-
-          sampling). This field is optional for FLAC and WAV audio
-          files, but is required for all other audio formats. For
-          details, see [AudioEncoding][google.cloud.speech.v1p1beta1.Rec
-          ognitionConfig.AudioEncoding].
-      audio_channel_count:
-          The number of channels in the input audio data. ONLY set this
-          for MULTI-CHANNEL recognition. Valid values for LINEAR16 and
-          FLAC are ``1``-``8``. Valid values for OGG\_OPUS are
-          '1'-'254'. Valid value for MULAW, AMR, AMR\_WB and
-          SPEEX\_WITH\_HEADER\_BYTE is only ``1``. If ``0`` or omitted,
-          defaults to one channel (mono). Note: We only recognize the
-          first channel by default. To perform independent recognition
-          on each channel set
-          ``enable_separate_recognition_per_channel`` to 'true'.
-      enable_separate_recognition_per_channel:
-          This needs to be set to ``true`` explicitly and
-          ``audio_channel_count`` > 1 to get each channel recognized
-          separately. The recognition result will contain a
-          ``channel_tag`` field to state which channel that result
-          belongs to. If this is not true, we will only recognize the
-          first channel. The request is billed cumulatively for all
-          channels recognized: ``audio_channel_count`` multiplied by the
-          length of the audio.
-      language_code:
-          Required. The language of the supplied audio as a `BCP-47
-          <https://www.rfc-editor.org/rfc/bcp/bcp47.txt>`__ language
-          tag. Example: "en-US". See `Language Support
-          <https://cloud.google.com/speech-to-text/docs/languages>`__
-          for a list of the currently supported language codes.
-      alternative_language_codes:
-          A list of up to 3 additional `BCP-47 <https://www.rfc-
-          editor.org/rfc/bcp/bcp47.txt>`__ language tags, listing
-          possible alternative languages of the supplied audio. See
-          `Language Support <https://cloud.google.com/speech-to-
-          text/docs/languages>`__ for a list of the currently supported
-          language codes. If alternative languages are listed,
-          recognition result will contain recognition in the most likely
-          language detected including the main language\_code. The
-          recognition result will include the language tag of the
-          language detected in the audio. Note: This feature is only
-          supported for Voice Command and Voice Search use cases and
-          performance may vary for other use cases (e.g., phone call
-          transcription).
-      max_alternatives:
-          Maximum number of recognition hypotheses to be returned.
-          Specifically, the maximum number of
-          ``SpeechRecognitionAlternative`` messages within each
-          ``SpeechRecognitionResult``. The server may return fewer than
-          ``max_alternatives``. Valid values are ``0``-``30``. A value
-          of ``0`` or ``1`` will return a maximum of one. If omitted,
-          will return a maximum of one.
-      profanity_filter:
-          If set to ``true``, the server will attempt to filter out
-          profanities, replacing all but the initial character in each
-          filtered word with asterisks, e.g. "f\*\*\*". If set to
-          ``false`` or omitted, profanities won't be filtered out.
-      speech_contexts:
-          Array of
-          [SpeechContext][google.cloud.speech.v1p1beta1.SpeechContext].
-          A means to provide context to assist the speech recognition.
-          For more information, see `speech adaptation
-          <https://cloud.google.com/speech-to-text/docs/context-
-          strength>`__.
-      enable_word_time_offsets:
-          If ``true``, the top result includes a list of words and the
-          start and end time offsets (timestamps) for those words. If
-          ``false``, no word-level time offset information is returned.
-          The default is ``false``.
-      enable_word_confidence:
-          If ``true``, the top result includes a list of words and the
-          confidence for those words. If ``false``, no word-level
-          confidence information is returned. The default is ``false``.
-      enable_automatic_punctuation:
-          If 'true', adds punctuation to recognition result hypotheses.
-          This feature is only available in select languages. Setting
-          this for requests in other languages has no effect at all. The
-          default 'false' value does not add punctuation to result
-          hypotheses. Note: This is currently offered as an experimental
-          service, complimentary to all users. In the future this may be
-          exclusively available as a premium feature.
-      enable_speaker_diarization:
-          If 'true', enables speaker detection for each recognized word
-          in the top alternative of the recognition result using a
-          speaker\_tag provided in the WordInfo. Note: Use
-          diarization\_config instead.
-      diarization_speaker_count:
-          If set, specifies the estimated number of speakers in the
-          conversation. Defaults to '2'. Ignored unless
-          enable\_speaker\_diarization is set to true. Note: Use
-          diarization\_config instead.
-      diarization_config:
-          Config to enable speaker diarization and set additional
-          parameters to make diarization better suited for your
-          application. Note: When this is enabled, we send all the words
-          from the beginning of the audio for the top alternative in
-          every consecutive STREAMING responses. This is done in order
-          to improve our speaker tags as our models learn to identify
-          the speakers in the conversation over time. For non-streaming
-          requests, the diarization results will be provided only in the
-          top alternative of the FINAL SpeechRecognitionResult.
-      metadata:
-          Metadata regarding this request.
-      model:
-          Which model to select for the given request. Select the model
-          best suited to your domain to get best results. If a model is
-          not explicitly specified, then we auto-select a model based on
-          the parameters in the RecognitionConfig.  .. raw:: html
-          <table>  .. raw:: html     <tr>  ::      <td><b>Model</b></td>
-          <td><b>Description</b></td>  .. raw:: html     </tr>  .. raw::
-          html     <tr>  ::
-          <td><code>command_and_search</code></td>     <td>Best for
-          short queries such as voice commands or voice search.</td>  ..
-          raw:: html     </tr>  .. raw:: html     <tr>  ::
-          <td><code>phone_call</code></td>     <td>Best for audio that
-          originated from a phone call (typically     recorded at an
-          8khz sampling rate).</td>  .. raw:: html     </tr>  .. raw::
-          html     <tr>  ::      <td><code>video</code></td>
-          <td>Best for audio that originated from from video or includes
-          multiple         speakers. Ideally the audio is recorded at a
-          16khz or greater         sampling rate. This is a premium
-          model that costs more than the         standard rate.</td>  ..
-          raw:: html     </tr>  .. raw:: html     <tr>  ::
-          <td><code>default</code></td>     <td>Best for audio that is
-          not one of the specific audio models.         For example,
-          long-form audio. Ideally the audio is high-fidelity,
-          recorded at a 16khz or greater sampling rate.</td>  .. raw::
-          html     </tr>  .. raw:: html     </table>
-      use_enhanced:
-          Set to true to use an enhanced model for speech recognition.
-          If ``use_enhanced`` is set to true and the ``model`` field is
-          not set, then an appropriate enhanced model is chosen if an
-          enhanced model exists for the audio.  If ``use_enhanced`` is
-          true and an enhanced version of the specified model does not
-          exist, then the speech is recognized using the standard
-          version of the specified model.
-  """,
-        # @@protoc_insertion_point(class_scope:google.cloud.speech.v1p1beta1.RecognitionConfig)
-    ),
-)
-_sym_db.RegisterMessage(RecognitionConfig)
-
-SpeakerDiarizationConfig = _reflection.GeneratedProtocolMessageType(
-    "SpeakerDiarizationConfig",
-    (_message.Message,),
-    dict(
-        DESCRIPTOR=_SPEAKERDIARIZATIONCONFIG,
-        __module__="google.cloud.speech_v1p1beta1.proto.cloud_speech_pb2",
-        __doc__="""Config to enable speaker diarization.
-  
-  
-  Attributes:
-      enable_speaker_diarization:
-          If 'true', enables speaker detection for each recognized word
-          in the top alternative of the recognition result using a
-          speaker\_tag provided in the WordInfo.
-      min_speaker_count:
-          Minimum number of speakers in the conversation. This range
-          gives you more flexibility by allowing the system to
-          automatically determine the correct number of speakers. If not
-          set, the default value is 2.
-      max_speaker_count:
-          Maximum number of speakers in the conversation. This range
-          gives you more flexibility by allowing the system to
-          automatically determine the correct number of speakers. If not
-          set, the default value is 6.
-  """,
-        # @@protoc_insertion_point(class_scope:google.cloud.speech.v1p1beta1.SpeakerDiarizationConfig)
-    ),
-)
-_sym_db.RegisterMessage(SpeakerDiarizationConfig)
-
-RecognitionMetadata = _reflection.GeneratedProtocolMessageType(
-    "RecognitionMetadata",
-    (_message.Message,),
-    dict(
-        DESCRIPTOR=_RECOGNITIONMETADATA,
-        __module__="google.cloud.speech_v1p1beta1.proto.cloud_speech_pb2",
-        __doc__="""Description of audio data to be recognized.
-  
-  
-  Attributes:
-      interaction_type:
-          The use case most closely describing the audio content to be
-          recognized.
-      industry_naics_code_of_audio:
-          The industry vertical to which this speech recognition request
-          most closely applies. This is most indicative of the topics
-          contained in the audio. Use the 6-digit NAICS code to identify
-          the industry vertical - see https://www.naics.com/search/.
-      microphone_distance:
-          The audio type that most closely describes the audio being
-          recognized.
-      original_media_type:
-          The original media the speech was recorded on.
-      recording_device_type:
-          The type of device the speech was recorded with.
-      recording_device_name:
-          The device used to make the recording. Examples 'Nexus 5X' or
-          'Polycom SoundStation IP 6000' or 'POTS' or 'VoIP' or
-          'Cardioid Microphone'.
-      original_mime_type:
-          Mime type of the original audio file. For example
-          ``audio/m4a``, ``audio/x-alaw-basic``, ``audio/mp3``,
-          ``audio/3gpp``. A list of possible audio mime types is
-          maintained at http://www.iana.org/assignments/media-
-          types/media-types.xhtml#audio
-      obfuscated_id:
-          Obfuscated (privacy-protected) ID of the user, to identify
-          number of unique users using the service.
-      audio_topic:
-          Description of the content. Eg. "Recordings of federal supreme
-          court hearings from 2012".
-  """,
-        # @@protoc_insertion_point(class_scope:google.cloud.speech.v1p1beta1.RecognitionMetadata)
-    ),
-)
-_sym_db.RegisterMessage(RecognitionMetadata)
-
-SpeechContext = _reflection.GeneratedProtocolMessageType(
-    "SpeechContext",
-    (_message.Message,),
-    dict(
-        DESCRIPTOR=_SPEECHCONTEXT,
-        __module__="google.cloud.speech_v1p1beta1.proto.cloud_speech_pb2",
-        __doc__="""Provides "hints" to the speech recognizer to favor
-  specific words and phrases in the results.
-  
-  
-  Attributes:
-      phrases:
-          A list of strings containing words and phrases "hints" so that
-          the speech recognition is more likely to recognize them. This
-          can be used to improve the accuracy for specific words and
-          phrases, for example, if specific commands are typically
-          spoken by the user. This can also be used to add additional
-          words to the vocabulary of the recognizer. See `usage limits
-          <https://cloud.google.com/speech-to-text/quotas#content>`__.
-          List items can also be set to classes for groups of words that
-          represent common concepts that occur in natural language. For
-          example, rather than providing phrase hints for every month of
-          the year, using the $MONTH class improves the likelihood of
-          correctly transcribing audio that includes months.
-      boost:
-          Hint Boost. Positive value will increase the probability that
-          a specific phrase will be recognized over other similar
-          sounding phrases. The higher the boost, the higher the chance
-          of false positive recognition as well. Negative boost values
-          would correspond to anti-biasing. Anti-biasing is not enabled,
-          so negative boost will simply be ignored. Though ``boost`` can
-          accept a wide range of positive values, most use cases are
-          best served with values between 0 and 20. We recommend using a
-          binary search approach to finding the optimal value for your
-          use case.
-  """,
-        # @@protoc_insertion_point(class_scope:google.cloud.speech.v1p1beta1.SpeechContext)
-    ),
-)
-_sym_db.RegisterMessage(SpeechContext)
-
-RecognitionAudio = _reflection.GeneratedProtocolMessageType(
-    "RecognitionAudio",
-    (_message.Message,),
-    dict(
-        DESCRIPTOR=_RECOGNITIONAUDIO,
-        __module__="google.cloud.speech_v1p1beta1.proto.cloud_speech_pb2",
-        __doc__="""Contains audio data in the encoding specified in the
-  ``RecognitionConfig``. Either ``content`` or ``uri`` must be supplied.
-  Supplying both or neither returns
-  [google.rpc.Code.INVALID\_ARGUMENT][google.rpc.Code.INVALID\_ARGUMENT].
-  See `content
-  limits <https://cloud.google.com/speech-to-text/quotas#content>`__.
-  
-  
-  Attributes:
-      audio_source:
-          The audio source, which is either inline content or a Google
-          Cloud Storage uri.
-      content:
-          The audio data bytes encoded as specified in
-          ``RecognitionConfig``. Note: as with all bytes fields, proto
-          buffers use a pure binary representation, whereas JSON
-          representations use base64.
-      uri:
-          URI that points to a file that contains audio data bytes as
-          specified in ``RecognitionConfig``. The file must not be
-          compressed (for example, gzip). Currently, only Google Cloud
-          Storage URIs are supported, which must be specified in the
-          following format: ``gs://bucket_name/object_name`` (other URI
-          formats return [google.rpc.Code.INVALID\_ARGUMENT][google.rpc.
-          Code.INVALID\_ARGUMENT]). For more information, see `Request
-          URIs <https://cloud.google.com/storage/docs/reference-
-          uris>`__.
-  """,
-        # @@protoc_insertion_point(class_scope:google.cloud.speech.v1p1beta1.RecognitionAudio)
-    ),
-)
-_sym_db.RegisterMessage(RecognitionAudio)
-
-RecognizeResponse = _reflection.GeneratedProtocolMessageType(
-    "RecognizeResponse",
-    (_message.Message,),
-    dict(
-        DESCRIPTOR=_RECOGNIZERESPONSE,
-        __module__="google.cloud.speech_v1p1beta1.proto.cloud_speech_pb2",
-        __doc__="""The only message returned to the client by the
-  ``Recognize`` method. It contains the result as zero or more sequential
-  ``SpeechRecognitionResult`` messages.
-  
-  
-  Attributes:
-      results:
-          Sequential list of transcription results corresponding to
-          sequential portions of audio.
-  """,
-        # @@protoc_insertion_point(class_scope:google.cloud.speech.v1p1beta1.RecognizeResponse)
-    ),
-)
-_sym_db.RegisterMessage(RecognizeResponse)
-
-LongRunningRecognizeResponse = _reflection.GeneratedProtocolMessageType(
-    "LongRunningRecognizeResponse",
-    (_message.Message,),
-    dict(
-        DESCRIPTOR=_LONGRUNNINGRECOGNIZERESPONSE,
-        __module__="google.cloud.speech_v1p1beta1.proto.cloud_speech_pb2",
-        __doc__="""The only message returned to the client by the
-  ``LongRunningRecognize`` method. It contains the result as zero or more
-  sequential ``SpeechRecognitionResult`` messages. It is included in the
-  ``result.response`` field of the ``Operation`` returned by the
-  ``GetOperation`` call of the ``google::longrunning::Operations``
-  service.
-  
-  
-  Attributes:
-      results:
-          Sequential list of transcription results corresponding to
-          sequential portions of audio.
-  """,
-        # @@protoc_insertion_point(class_scope:google.cloud.speech.v1p1beta1.LongRunningRecognizeResponse)
-    ),
-)
-_sym_db.RegisterMessage(LongRunningRecognizeResponse)
-
-LongRunningRecognizeMetadata = _reflection.GeneratedProtocolMessageType(
-    "LongRunningRecognizeMetadata",
-    (_message.Message,),
-    dict(
-        DESCRIPTOR=_LONGRUNNINGRECOGNIZEMETADATA,
-        __module__="google.cloud.speech_v1p1beta1.proto.cloud_speech_pb2",
-        __doc__="""Describes the progress of a long-running
-  ``LongRunningRecognize`` call. It is included in the ``metadata`` field
-  of the ``Operation`` returned by the ``GetOperation`` call of the
-  ``google::longrunning::Operations`` service.
-  
-  
-  Attributes:
-      progress_percent:
-          Approximate percentage of audio processed thus far. Guaranteed
-          to be 100 when the audio is fully processed and the results
-          are available.
-      start_time:
-          Time when the request was received.
-      last_update_time:
-          Time of the most recent processing update.
-  """,
-        # @@protoc_insertion_point(class_scope:google.cloud.speech.v1p1beta1.LongRunningRecognizeMetadata)
-    ),
-)
-_sym_db.RegisterMessage(LongRunningRecognizeMetadata)
-
-StreamingRecognizeResponse = _reflection.GeneratedProtocolMessageType(
-    "StreamingRecognizeResponse",
-    (_message.Message,),
-    dict(
-        DESCRIPTOR=_STREAMINGRECOGNIZERESPONSE,
-        __module__="google.cloud.speech_v1p1beta1.proto.cloud_speech_pb2",
-        __doc__="""\ ``StreamingRecognizeResponse`` is the only message
-  returned to the client by ``StreamingRecognize``. A series of zero or
-  more ``StreamingRecognizeResponse`` messages are streamed back to the
-  client. If there is no recognizable audio, and ``single_utterance`` is
-  set to false, then no messages are streamed back to the client.
-  
-  Here's an example of a series of ten ``StreamingRecognizeResponse``\ s
-  that might be returned while processing audio:
-  
-  1. results { alternatives { transcript: "tube" } stability: 0.01 }
-  
-  2. results { alternatives { transcript: "to be a" } stability: 0.01 }
-  
-  3. results { alternatives { transcript: "to be" } stability: 0.9 }
-     results { alternatives { transcript: " or not to be" } stability:
-     0.01 }
-  
-  4. results { alternatives { transcript: "to be or not to be" confidence:
-     0.92 } alternatives { transcript: "to bee or not to bee" } is\_final:
-     true }
-  
-  5. results { alternatives { transcript: " that's" } stability: 0.01 }
-  
-  6. results { alternatives { transcript: " that is" } stability: 0.9 }
-     results { alternatives { transcript: " the question" } stability:
-     0.01 }
-  
-  7. results { alternatives { transcript: " that is the question"
-     confidence: 0.98 } alternatives { transcript: " that was the
-     question" } is\_final: true }
-  
-  Notes:
-  
-  -  Only two of the above responses #4 and #7 contain final results; they
-     are indicated by ``is_final: true``. Concatenating these together
-     generates the full transcript: "to be or not to be that is the
-     question".
-  
-  -  The others contain interim ``results``. #3 and #6 contain two interim
-     ``results``: the first portion has a high stability and is less
-     likely to change; the second portion has a low stability and is very
-     likely to change. A UI designer might choose to show only high
-     stability ``results``.
-  
-  -  The specific ``stability`` and ``confidence`` values shown above are
-     only for illustrative purposes. Actual values may vary.
-  
-  -  In each response, only one of these fields will be set: ``error``,
-     ``speech_event_type``, or one or more (repeated) ``results``.
-  
-  
-  Attributes:
-      error:
-          If set, returns a [google.rpc.Status][google.rpc.Status]
-          message that specifies the error for the operation.
-      results:
-          This repeated list contains zero or more results that
-          correspond to consecutive portions of the audio currently
-          being processed. It contains zero or one ``is_final=true``
-          result (the newly settled portion), followed by zero or more
-          ``is_final=false`` results (the interim results).
-      speech_event_type:
-          Indicates the type of speech event.
-  """,
-        # @@protoc_insertion_point(class_scope:google.cloud.speech.v1p1beta1.StreamingRecognizeResponse)
-    ),
-)
-_sym_db.RegisterMessage(StreamingRecognizeResponse)
-
-StreamingRecognitionResult = _reflection.GeneratedProtocolMessageType(
-    "StreamingRecognitionResult",
-    (_message.Message,),
-    dict(
-        DESCRIPTOR=_STREAMINGRECOGNITIONRESULT,
-        __module__="google.cloud.speech_v1p1beta1.proto.cloud_speech_pb2",
-        __doc__="""A streaming speech recognition result corresponding to a
-  portion of the audio that is currently being processed.
-  
-  
-  Attributes:
-      alternatives:
-          May contain one or more recognition hypotheses (up to the
-          maximum specified in ``max_alternatives``). These alternatives
-          are ordered in terms of accuracy, with the top (first)
-          alternative being the most probable, as ranked by the
-          recognizer.
-      is_final:
-          If ``false``, this ``StreamingRecognitionResult`` represents
-          an interim result that may change. If ``true``, this is the
-          final time the speech service will return this particular
-          ``StreamingRecognitionResult``, the recognizer will not return
-          any further hypotheses for this portion of the transcript and
-          corresponding audio.
-      stability:
-          An estimate of the likelihood that the recognizer will not
-          change its guess about this interim result. Values range from
-          0.0 (completely unstable) to 1.0 (completely stable). This
-          field is only provided for interim results
-          (``is_final=false``). The default of 0.0 is a sentinel value
-          indicating ``stability`` was not set.
-      result_end_time:
-          Time offset of the end of this result relative to the
-          beginning of the audio.
-      channel_tag:
-          For multi-channel audio, this is the channel number
-          corresponding to the recognized result for the audio from that
-          channel. For audio\_channel\_count = N, its output values can
-          range from '1' to 'N'.
-      language_code:
-          The `BCP-47 <https://www.rfc-editor.org/rfc/bcp/bcp47.txt>`__
-          language tag of the language in this result. This language
-          code was detected to have the most likelihood of being spoken
-          in the audio.
-  """,
-        # @@protoc_insertion_point(class_scope:google.cloud.speech.v1p1beta1.StreamingRecognitionResult)
-    ),
-)
-_sym_db.RegisterMessage(StreamingRecognitionResult)
-
-SpeechRecognitionResult = _reflection.GeneratedProtocolMessageType(
-    "SpeechRecognitionResult",
-    (_message.Message,),
-    dict(
-        DESCRIPTOR=_SPEECHRECOGNITIONRESULT,
-        __module__="google.cloud.speech_v1p1beta1.proto.cloud_speech_pb2",
-        __doc__="""A speech recognition result corresponding to a portion of the audio.
-  
-  
-  Attributes:
-      alternatives:
-          May contain one or more recognition hypotheses (up to the
-          maximum specified in ``max_alternatives``). These alternatives
-          are ordered in terms of accuracy, with the top (first)
-          alternative being the most probable, as ranked by the
-          recognizer.
-      channel_tag:
-          For multi-channel audio, this is the channel number
-          corresponding to the recognized result for the audio from that
-          channel. For audio\_channel\_count = N, its output values can
-          range from '1' to 'N'.
-      language_code:
-          The `BCP-47 <https://www.rfc-editor.org/rfc/bcp/bcp47.txt>`__
-          language tag of the language in this result. This language
-          code was detected to have the most likelihood of being spoken
-          in the audio.
-  """,
-        # @@protoc_insertion_point(class_scope:google.cloud.speech.v1p1beta1.SpeechRecognitionResult)
-    ),
-)
-_sym_db.RegisterMessage(SpeechRecognitionResult)
-
-SpeechRecognitionAlternative = _reflection.GeneratedProtocolMessageType(
-    "SpeechRecognitionAlternative",
-    (_message.Message,),
-    dict(
-        DESCRIPTOR=_SPEECHRECOGNITIONALTERNATIVE,
-        __module__="google.cloud.speech_v1p1beta1.proto.cloud_speech_pb2",
-        __doc__="""Alternative hypotheses (a.k.a. n-best list).
-  
-  
-  Attributes:
-      transcript:
-          Transcript text representing the words that the user spoke.
-      confidence:
-          The confidence estimate between 0.0 and 1.0. A higher number
-          indicates an estimated greater likelihood that the recognized
-          words are correct. This field is set only for the top
-          alternative of a non-streaming result or, of a streaming
-          result where ``is_final=true``. This field is not guaranteed
-          to be accurate and users should not rely on it to be always
-          provided. The default of 0.0 is a sentinel value indicating
-          ``confidence`` was not set.
-      words:
-          A list of word-specific information for each recognized word.
-          Note: When ``enable_speaker_diarization`` is true, you will
-          see all the words from the beginning of the audio.
-  """,
-        # @@protoc_insertion_point(class_scope:google.cloud.speech.v1p1beta1.SpeechRecognitionAlternative)
-    ),
-)
-_sym_db.RegisterMessage(SpeechRecognitionAlternative)
-
-WordInfo = _reflection.GeneratedProtocolMessageType(
-    "WordInfo",
-    (_message.Message,),
-    dict(
-        DESCRIPTOR=_WORDINFO,
-        __module__="google.cloud.speech_v1p1beta1.proto.cloud_speech_pb2",
-        __doc__="""Word-specific information for recognized words.
-  
-  
-  Attributes:
-      start_time:
-          Time offset relative to the beginning of the audio, and
-          corresponding to the start of the spoken word. This field is
-          only set if ``enable_word_time_offsets=true`` and only in the
-          top hypothesis. This is an experimental feature and the
-          accuracy of the time offset can vary.
-      end_time:
-          Time offset relative to the beginning of the audio, and
-          corresponding to the end of the spoken word. This field is
-          only set if ``enable_word_time_offsets=true`` and only in the
-          top hypothesis. This is an experimental feature and the
-          accuracy of the time offset can vary.
-      word:
-          The word corresponding to this set of information.
-      confidence:
-          The confidence estimate between 0.0 and 1.0. A higher number
-          indicates an estimated greater likelihood that the recognized
-          words are correct. This field is set only for the top
-          alternative of a non-streaming result or, of a streaming
-          result where ``is_final=true``. This field is not guaranteed
-          to be accurate and users should not rely on it to be always
-          provided. The default of 0.0 is a sentinel value indicating
-          ``confidence`` was not set.
-      speaker_tag:
-          A distinct integer value is assigned for every speaker within
-          the audio. This field specifies which one of those speakers
-          was detected to have spoken this word. Value ranges from '1'
-          to diarization\_speaker\_count. speaker\_tag is set if
-          enable\_speaker\_diarization = 'true' and only in the top
-          alternative.
-  """,
-        # @@protoc_insertion_point(class_scope:google.cloud.speech.v1p1beta1.WordInfo)
-    ),
-)
-_sym_db.RegisterMessage(WordInfo)
-
-
-DESCRIPTOR._options = None
-_RECOGNIZEREQUEST.fields_by_name["config"]._options = None
-_RECOGNIZEREQUEST.fields_by_name["audio"]._options = None
-_LONGRUNNINGRECOGNIZEREQUEST.fields_by_name["config"]._options = None
-_LONGRUNNINGRECOGNIZEREQUEST.fields_by_name["audio"]._options = None
-_STREAMINGRECOGNITIONCONFIG.fields_by_name["config"]._options = None
-_RECOGNITIONCONFIG.fields_by_name["language_code"]._options = None
-_RECOGNITIONCONFIG.fields_by_name["enable_speaker_diarization"]._options = None
-_RECOGNITIONCONFIG.fields_by_name["diarization_speaker_count"]._options = None
-_RECOGNITIONMETADATA.fields_by_name["obfuscated_id"]._options = None
-
-_SPEECH = _descriptor.ServiceDescriptor(
-    name="Speech",
-    full_name="google.cloud.speech.v1p1beta1.Speech",
-    file=DESCRIPTOR,
-    index=0,
-    serialized_options=_b(
-        "\312A\025speech.googleapis.com\322A.https://www.googleapis.com/auth/cloud-platform"
-    ),
-    serialized_start=4626,
-    serialized_end=5268,
-    methods=[
-        _descriptor.MethodDescriptor(
-            name="Recognize",
-            full_name="google.cloud.speech.v1p1beta1.Speech.Recognize",
-            index=0,
-            containing_service=None,
-            input_type=_RECOGNIZEREQUEST,
-            output_type=_RECOGNIZERESPONSE,
-            serialized_options=_b(
-                '\202\323\344\223\002 "\033/v1p1beta1/speech:recognize:\001*\332A\014config,audio'
-            ),
-        ),
-        _descriptor.MethodDescriptor(
-            name="LongRunningRecognize",
-            full_name="google.cloud.speech.v1p1beta1.Speech.LongRunningRecognize",
-            index=1,
-            containing_service=None,
-            input_type=_LONGRUNNINGRECOGNIZEREQUEST,
-            output_type=google_dot_longrunning_dot_operations__pb2._OPERATION,
-            serialized_options=_b(
-                '\202\323\344\223\002+"&/v1p1beta1/speech:longrunningrecognize:\001*\332A\014config,audio\312A<\n\034LongRunningRecognizeResponse\022\034LongRunningRecognizeMetadata'
-            ),
-        ),
-        _descriptor.MethodDescriptor(
-            name="StreamingRecognize",
-            full_name="google.cloud.speech.v1p1beta1.Speech.StreamingRecognize",
-            index=2,
-            containing_service=None,
-            input_type=_STREAMINGRECOGNIZEREQUEST,
-            output_type=_STREAMINGRECOGNIZERESPONSE,
-            serialized_options=None,
-        ),
-    ],
-)
-_sym_db.RegisterServiceDescriptor(_SPEECH)
-
-DESCRIPTOR.services_by_name["Speech"] = _SPEECH
-
-# @@protoc_insertion_point(module_scope)
diff --git a/google/cloud/speech_v1p1beta1/proto/cloud_speech_pb2_grpc.py b/google/cloud/speech_v1p1beta1/proto/cloud_speech_pb2_grpc.py
deleted file mode 100644
index ba4950fd..00000000
--- a/google/cloud/speech_v1p1beta1/proto/cloud_speech_pb2_grpc.py
+++ /dev/null
@@ -1,93 +0,0 @@
-# Generated by the gRPC Python protocol compiler plugin. DO NOT EDIT!
-import grpc
-
-from google.cloud.speech_v1p1beta1.proto import (
-    cloud_speech_pb2 as google_dot_cloud_dot_speech__v1p1beta1_dot_proto_dot_cloud__speech__pb2,
-)
-from google.longrunning import (
-    operations_pb2 as google_dot_longrunning_dot_operations__pb2,
-)
-
-
-class SpeechStub(object):
-    """Service that implements Google Cloud Speech API.
-  """
-
-    def __init__(self, channel):
-        """Constructor.
-
-    Args:
-      channel: A grpc.Channel.
-    """
-        self.Recognize = channel.unary_unary(
-            "/google.cloud.speech.v1p1beta1.Speech/Recognize",
-            request_serializer=google_dot_cloud_dot_speech__v1p1beta1_dot_proto_dot_cloud__speech__pb2.RecognizeRequest.SerializeToString,
-            response_deserializer=google_dot_cloud_dot_speech__v1p1beta1_dot_proto_dot_cloud__speech__pb2.RecognizeResponse.FromString,
-        )
-        self.LongRunningRecognize = channel.unary_unary(
-            "/google.cloud.speech.v1p1beta1.Speech/LongRunningRecognize",
-            request_serializer=google_dot_cloud_dot_speech__v1p1beta1_dot_proto_dot_cloud__speech__pb2.LongRunningRecognizeRequest.SerializeToString,
-            response_deserializer=google_dot_longrunning_dot_operations__pb2.Operation.FromString,
-        )
-        self.StreamingRecognize = channel.stream_stream(
-            "/google.cloud.speech.v1p1beta1.Speech/StreamingRecognize",
-            request_serializer=google_dot_cloud_dot_speech__v1p1beta1_dot_proto_dot_cloud__speech__pb2.StreamingRecognizeRequest.SerializeToString,
-            response_deserializer=google_dot_cloud_dot_speech__v1p1beta1_dot_proto_dot_cloud__speech__pb2.StreamingRecognizeResponse.FromString,
-        )
-
-
-class SpeechServicer(object):
-    """Service that implements Google Cloud Speech API.
-  """
-
-    def Recognize(self, request, context):
-        """Performs synchronous speech recognition: receive results after all audio
-    has been sent and processed.
-    """
-        context.set_code(grpc.StatusCode.UNIMPLEMENTED)
-        context.set_details("Method not implemented!")
-        raise NotImplementedError("Method not implemented!")
-
-    def LongRunningRecognize(self, request, context):
-        """Performs asynchronous speech recognition: receive results via the
-    google.longrunning.Operations interface. Returns either an
-    `Operation.error` or an `Operation.response` which contains
-    a `LongRunningRecognizeResponse` message.
-    For more information on asynchronous speech recognition, see the
-    [how-to](https://cloud.google.com/speech-to-text/docs/async-recognize).
-    """
-        context.set_code(grpc.StatusCode.UNIMPLEMENTED)
-        context.set_details("Method not implemented!")
-        raise NotImplementedError("Method not implemented!")
-
-    def StreamingRecognize(self, request_iterator, context):
-        """Performs bidirectional streaming speech recognition: receive results while
-    sending audio. This method is only available via the gRPC API (not REST).
-    """
-        context.set_code(grpc.StatusCode.UNIMPLEMENTED)
-        context.set_details("Method not implemented!")
-        raise NotImplementedError("Method not implemented!")
-
-
-def add_SpeechServicer_to_server(servicer, server):
-    rpc_method_handlers = {
-        "Recognize": grpc.unary_unary_rpc_method_handler(
-            servicer.Recognize,
-            request_deserializer=google_dot_cloud_dot_speech__v1p1beta1_dot_proto_dot_cloud__speech__pb2.RecognizeRequest.FromString,
-            response_serializer=google_dot_cloud_dot_speech__v1p1beta1_dot_proto_dot_cloud__speech__pb2.RecognizeResponse.SerializeToString,
-        ),
-        "LongRunningRecognize": grpc.unary_unary_rpc_method_handler(
-            servicer.LongRunningRecognize,
-            request_deserializer=google_dot_cloud_dot_speech__v1p1beta1_dot_proto_dot_cloud__speech__pb2.LongRunningRecognizeRequest.FromString,
-            response_serializer=google_dot_longrunning_dot_operations__pb2.Operation.SerializeToString,
-        ),
-        "StreamingRecognize": grpc.stream_stream_rpc_method_handler(
-            servicer.StreamingRecognize,
-            request_deserializer=google_dot_cloud_dot_speech__v1p1beta1_dot_proto_dot_cloud__speech__pb2.StreamingRecognizeRequest.FromString,
-            response_serializer=google_dot_cloud_dot_speech__v1p1beta1_dot_proto_dot_cloud__speech__pb2.StreamingRecognizeResponse.SerializeToString,
-        ),
-    }
-    generic_handler = grpc.method_handlers_generic_handler(
-        "google.cloud.speech.v1p1beta1.Speech", rpc_method_handlers
-    )
-    server.add_generic_rpc_handlers((generic_handler,))
diff --git a/google/cloud/speech_v1p1beta1/types.py b/google/cloud/speech_v1p1beta1/types.py
deleted file mode 100644
index 35c7a2c0..00000000
--- a/google/cloud/speech_v1p1beta1/types.py
+++ /dev/null
@@ -1,48 +0,0 @@
-# -*- coding: utf-8 -*-
-#
-# Copyright 2020 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-
-from __future__ import absolute_import
-import sys
-
-from google.api_core.protobuf_helpers import get_messages
-
-from google.cloud.speech_v1p1beta1.proto import cloud_speech_pb2
-from google.longrunning import operations_pb2
-from google.protobuf import any_pb2
-from google.protobuf import duration_pb2
-from google.protobuf import timestamp_pb2
-from google.rpc import status_pb2
-
-
-_shared_modules = [operations_pb2, any_pb2, duration_pb2, timestamp_pb2, status_pb2]
-
-_local_modules = [cloud_speech_pb2]
-
-names = []
-
-for module in _shared_modules:  # pragma: NO COVER
-    for name, message in get_messages(module).items():
-        setattr(sys.modules[__name__], name, message)
-        names.append(name)
-for module in _local_modules:
-    for name, message in get_messages(module).items():
-        message.__module__ = "google.cloud.speech_v1p1beta1.types"
-        setattr(sys.modules[__name__], name, message)
-        names.append(name)
-
-
-__all__ = tuple(sorted(names))
diff --git a/noxfile.py b/noxfile.py
deleted file mode 100644
index 4cbae3ca..00000000
--- a/noxfile.py
+++ /dev/null
@@ -1,173 +0,0 @@
-# -*- coding: utf-8 -*-
-#
-# Copyright 2018 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-# Generated by synthtool. DO NOT EDIT!
-
-from __future__ import absolute_import
-import os
-import shutil
-
-import nox
-
-
-BLACK_VERSION = "black==19.3b0"
-BLACK_PATHS = ["docs", "google", "tests", "noxfile.py", "setup.py"]
-
-if os.path.exists("samples"):
-    BLACK_PATHS.append("samples")
-
-
-@nox.session(python="3.7")
-def lint(session):
-    """Run linters.
-
-    Returns a failure if the linters find linting errors or sufficiently
-    serious code quality issues.
-    """
-    session.install("flake8", BLACK_VERSION)
-    session.run("black", "--check", *BLACK_PATHS)
-    session.run("flake8", "google", "tests")
-
-
-@nox.session(python="3.6")
-def blacken(session):
-    """Run black.
-
-    Format code to uniform standard.
-
-    This currently uses Python 3.6 due to the automated Kokoro run of synthtool.
-    That run uses an image that doesn't have 3.6 installed. Before updating this
-    check the state of the `gcp_ubuntu_config` we use for that Kokoro run.
-    """
-    session.install(BLACK_VERSION)
-    session.run("black", *BLACK_PATHS)
-
-
-@nox.session(python="3.7")
-def lint_setup_py(session):
-    """Verify that setup.py is valid (including RST check)."""
-    session.install("docutils", "pygments")
-    session.run("python", "setup.py", "check", "--restructuredtext", "--strict")
-
-
-def default(session):
-    # Install all test dependencies, then install this package in-place.
-    session.install("mock", "pytest", "pytest-cov")
-    session.install("-e", ".")
-
-    # Run py.test against the unit tests.
-    session.run(
-        "py.test",
-        "--quiet",
-        "--cov=google.cloud",
-        "--cov=tests.unit",
-        "--cov-append",
-        "--cov-config=.coveragerc",
-        "--cov-report=",
-        "--cov-fail-under=0",
-        os.path.join("tests", "unit"),
-        *session.posargs,
-    )
-
-
-@nox.session(python=["2.7", "3.5", "3.6", "3.7", "3.8"])
-def unit(session):
-    """Run the unit test suite."""
-    default(session)
-
-
-@nox.session(python=["2.7", "3.7"])
-def system(session):
-    """Run the system test suite."""
-    system_test_path = os.path.join("tests", "system.py")
-    system_test_folder_path = os.path.join("tests", "system")
-    # Sanity check: Only run tests if the environment variable is set.
-    if not os.environ.get("GOOGLE_APPLICATION_CREDENTIALS", ""):
-        session.skip("Credentials must be set via environment variable")
-
-    system_test_exists = os.path.exists(system_test_path)
-    system_test_folder_exists = os.path.exists(system_test_folder_path)
-    # Sanity check: only run tests if found.
-    if not system_test_exists and not system_test_folder_exists:
-        session.skip("System tests were not found")
-
-    # Use pre-release gRPC for system tests.
-    session.install("--pre", "grpcio")
-
-    # Install all test dependencies, then install this package into the
-    # virtualenv's dist-packages.
-    session.install("mock", "pytest")
-
-    session.install("-e", ".")
-
-    # Run py.test against the system tests.
-    if system_test_exists:
-        session.run("py.test", "--quiet", system_test_path, *session.posargs)
-    if system_test_folder_exists:
-        session.run("py.test", "--quiet", system_test_folder_path, *session.posargs)
-
-
-@nox.session(python=["3.7"])
-def samples(session):
-    """Run the sample test suite."""
-    # Sanity check: Only run tests if the environment variable is set.
-    if not os.environ.get("GOOGLE_APPLICATION_CREDENTIALS", ""):
-        session.skip("Credentials must be set via environment variable")
-
-    samples_path = "samples"
-    if not os.path.exists(samples_path):
-        session.skip("Samples not found.")
-
-    session.install("pyyaml")
-    session.install("sample-tester")
-    session.install("-e", ".")
-
-    session.run("sample-tester", samples_path, *session.posargs)
-
-
-@nox.session(python="3.7")
-def cover(session):
-    """Run the final coverage report.
-
-    This outputs the coverage report aggregating coverage from the unit
-    test runs (not system test runs), and then erases coverage data.
-    """
-    session.install("coverage", "pytest-cov")
-    session.run("coverage", "report", "--show-missing", "--fail-under=87")
-
-    session.run("coverage", "erase")
-
-
-@nox.session(python="3.7")
-def docs(session):
-    """Build the docs for this library."""
-
-    session.install("-e", ".")
-    session.install("sphinx", "alabaster", "recommonmark")
-
-    shutil.rmtree(os.path.join("docs", "_build"), ignore_errors=True)
-    session.run(
-        "sphinx-build",
-        "-W",  # warnings as errors
-        "-T",  # show full traceback on exception
-        "-N",  # no colors
-        "-b",
-        "html",
-        "-d",
-        os.path.join("docs", "_build", "doctrees", ""),
-        os.path.join("docs", ""),
-        os.path.join("docs", "_build", "html", ""),
-    )
diff --git a/pylint.config.py b/pylint.config.py
deleted file mode 100644
index 5d64b9d2..00000000
--- a/pylint.config.py
+++ /dev/null
@@ -1,25 +0,0 @@
-# Copyright 2017 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     http://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-"""This module is used to configure gcp-devrel-py-tools run-pylint."""
-
-# Library configuration
-
-# library_additions = {}
-# library_replacements = {}
-
-# Test configuration
-
-# test_additions = copy.deepcopy(library_additions)
-# test_replacements = copy.deepcopy(library_replacements)
diff --git a/renovate.json b/renovate.json
deleted file mode 100644
index 4fa94931..00000000
--- a/renovate.json
+++ /dev/null
@@ -1,5 +0,0 @@
-{
-  "extends": [
-    "config:base",  ":preserveSemverRanges"
-  ]
-}
diff --git a/samples/resources/brooklyn_bridge.flac b/samples/resources/brooklyn_bridge.flac
deleted file mode 100644
index 0e101a55..00000000
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diff --git a/samples/resources/brooklyn_bridge.mp3 b/samples/resources/brooklyn_bridge.mp3
deleted file mode 100644
index 97f9955c..00000000
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diff --git a/samples/resources/brooklyn_bridge.raw b/samples/resources/brooklyn_bridge.raw
deleted file mode 100644
index 5ebf79d3..00000000
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diff --git a/samples/resources/brooklyn_bridge.wav b/samples/resources/brooklyn_bridge.wav
deleted file mode 100644
index 044086e9..00000000
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diff --git a/samples/resources/commercial_mono.wav b/samples/resources/commercial_mono.wav
deleted file mode 100644
index e6b9ed43..00000000
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diff --git a/samples/resources/hello.raw b/samples/resources/hello.raw
deleted file mode 100644
index b5b46450..00000000
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diff --git a/samples/resources/hello.wav b/samples/resources/hello.wav
deleted file mode 100644
index 69b50693..00000000
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diff --git a/samples/resources/multi.flac b/samples/resources/multi.flac
deleted file mode 100644
index c550e0f4..00000000
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diff --git a/samples/resources/multi.wav b/samples/resources/multi.wav
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diff --git a/samples/v1/speech_transcribe_async.py b/samples/v1/speech_transcribe_async.py
deleted file mode 100644
index 6e803983..00000000
--- a/samples/v1/speech_transcribe_async.py
+++ /dev/null
@@ -1,90 +0,0 @@
-# -*- coding: utf-8 -*-
-#
-# Copyright 2020 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-# DO NOT EDIT! This is a generated sample ("LongRunningPromise",  "speech_transcribe_async")
-
-# To install the latest published package dependency, execute the following:
-#   pip install google-cloud-speech
-
-# sample-metadata
-#   title: Transcribe Audio File using Long Running Operation (Local File) (LRO)
-#   description: Transcribe a long audio file using asynchronous speech recognition
-#   usage: python3 samples/v1/speech_transcribe_async.py [--local_file_path "resources/brooklyn_bridge.raw"]
-
-# [START speech_transcribe_async]
-from google.cloud import speech_v1
-from google.cloud.speech_v1 import enums
-import io
-
-
-def sample_long_running_recognize(local_file_path):
-    """
-    Transcribe a long audio file using asynchronous speech recognition
-
-    Args:
-      local_file_path Path to local audio file, e.g. /path/audio.wav
-    """
-
-    client = speech_v1.SpeechClient()
-
-    # local_file_path = 'resources/brooklyn_bridge.raw'
-
-    # The language of the supplied audio
-    language_code = "en-US"
-
-    # Sample rate in Hertz of the audio data sent
-    sample_rate_hertz = 16000
-
-    # Encoding of audio data sent. This sample sets this explicitly.
-    # This field is optional for FLAC and WAV audio formats.
-    encoding = enums.RecognitionConfig.AudioEncoding.LINEAR16
-    config = {
-        "language_code": language_code,
-        "sample_rate_hertz": sample_rate_hertz,
-        "encoding": encoding,
-    }
-    with io.open(local_file_path, "rb") as f:
-        content = f.read()
-    audio = {"content": content}
-
-    operation = client.long_running_recognize(config, audio)
-
-    print(u"Waiting for operation to complete...")
-    response = operation.result()
-
-    for result in response.results:
-        # First alternative is the most probable result
-        alternative = result.alternatives[0]
-        print(u"Transcript: {}".format(alternative.transcript))
-
-
-# [END speech_transcribe_async]
-
-
-def main():
-    import argparse
-
-    parser = argparse.ArgumentParser()
-    parser.add_argument(
-        "--local_file_path", type=str, default="resources/brooklyn_bridge.raw"
-    )
-    args = parser.parse_args()
-
-    sample_long_running_recognize(args.local_file_path)
-
-
-if __name__ == "__main__":
-    main()
diff --git a/samples/v1/speech_transcribe_async_gcs.py b/samples/v1/speech_transcribe_async_gcs.py
deleted file mode 100644
index 79dd2cf7..00000000
--- a/samples/v1/speech_transcribe_async_gcs.py
+++ /dev/null
@@ -1,91 +0,0 @@
-# -*- coding: utf-8 -*-
-#
-# Copyright 2020 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-# DO NOT EDIT! This is a generated sample ("LongRunningPromise",  "speech_transcribe_async_gcs")
-
-# To install the latest published package dependency, execute the following:
-#   pip install google-cloud-speech
-
-# sample-metadata
-#   title: Transcript Audio File using Long Running Operation (Cloud Storage) (LRO)
-#   description: Transcribe long audio file from Cloud Storage using asynchronous speech
-#     recognition
-#   usage: python3 samples/v1/speech_transcribe_async_gcs.py [--storage_uri "gs://cloud-samples-data/speech/brooklyn_bridge.raw"]
-
-# [START speech_transcribe_async_gcs]
-from google.cloud import speech_v1
-from google.cloud.speech_v1 import enums
-
-
-def sample_long_running_recognize(storage_uri):
-    """
-    Transcribe long audio file from Cloud Storage using asynchronous speech
-    recognition
-
-    Args:
-      storage_uri URI for audio file in Cloud Storage, e.g. gs://[BUCKET]/[FILE]
-    """
-
-    client = speech_v1.SpeechClient()
-
-    # storage_uri = 'gs://cloud-samples-data/speech/brooklyn_bridge.raw'
-
-    # Sample rate in Hertz of the audio data sent
-    sample_rate_hertz = 16000
-
-    # The language of the supplied audio
-    language_code = "en-US"
-
-    # Encoding of audio data sent. This sample sets this explicitly.
-    # This field is optional for FLAC and WAV audio formats.
-    encoding = enums.RecognitionConfig.AudioEncoding.LINEAR16
-    config = {
-        "sample_rate_hertz": sample_rate_hertz,
-        "language_code": language_code,
-        "encoding": encoding,
-    }
-    audio = {"uri": storage_uri}
-
-    operation = client.long_running_recognize(config, audio)
-
-    print(u"Waiting for operation to complete...")
-    response = operation.result()
-
-    for result in response.results:
-        # First alternative is the most probable result
-        alternative = result.alternatives[0]
-        print(u"Transcript: {}".format(alternative.transcript))
-
-
-# [END speech_transcribe_async_gcs]
-
-
-def main():
-    import argparse
-
-    parser = argparse.ArgumentParser()
-    parser.add_argument(
-        "--storage_uri",
-        type=str,
-        default="gs://cloud-samples-data/speech/brooklyn_bridge.raw",
-    )
-    args = parser.parse_args()
-
-    sample_long_running_recognize(args.storage_uri)
-
-
-if __name__ == "__main__":
-    main()
diff --git a/samples/v1/speech_transcribe_async_word_time_offsets_gcs.py b/samples/v1/speech_transcribe_async_word_time_offsets_gcs.py
deleted file mode 100644
index 2adcb7cb..00000000
--- a/samples/v1/speech_transcribe_async_word_time_offsets_gcs.py
+++ /dev/null
@@ -1,98 +0,0 @@
-# -*- coding: utf-8 -*-
-#
-# Copyright 2020 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-# DO NOT EDIT! This is a generated sample ("LongRunningPromise",  "speech_transcribe_async_word_time_offsets_gcs")
-
-# To install the latest published package dependency, execute the following:
-#   pip install google-cloud-speech
-
-# sample-metadata
-#   title: Getting word timestamps (Cloud Storage) (LRO)
-#   description: Print start and end time of each word spoken in audio file from Cloud Storage
-#   usage: python3 samples/v1/speech_transcribe_async_word_time_offsets_gcs.py [--storage_uri "gs://cloud-samples-data/speech/brooklyn_bridge.flac"]
-
-# [START speech_transcribe_async_word_time_offsets_gcs]
-from google.cloud import speech_v1
-
-
-def sample_long_running_recognize(storage_uri):
-    """
-    Print start and end time of each word spoken in audio file from Cloud Storage
-
-    Args:
-      storage_uri URI for audio file in Cloud Storage, e.g. gs://[BUCKET]/[FILE]
-    """
-
-    client = speech_v1.SpeechClient()
-
-    # storage_uri = 'gs://cloud-samples-data/speech/brooklyn_bridge.flac'
-
-    # When enabled, the first result returned by the API will include a list
-    # of words and the start and end time offsets (timestamps) for those words.
-    enable_word_time_offsets = True
-
-    # The language of the supplied audio
-    language_code = "en-US"
-    config = {
-        "enable_word_time_offsets": enable_word_time_offsets,
-        "language_code": language_code,
-    }
-    audio = {"uri": storage_uri}
-
-    operation = client.long_running_recognize(config, audio)
-
-    print(u"Waiting for operation to complete...")
-    response = operation.result()
-
-    # The first result includes start and end time word offsets
-    result = response.results[0]
-    # First alternative is the most probable result
-    alternative = result.alternatives[0]
-    print(u"Transcript: {}".format(alternative.transcript))
-    # Print the start and end time of each word
-    for word in alternative.words:
-        print(u"Word: {}".format(word.word))
-        print(
-            u"Start time: {} seconds {} nanos".format(
-                word.start_time.seconds, word.start_time.nanos
-            )
-        )
-        print(
-            u"End time: {} seconds {} nanos".format(
-                word.end_time.seconds, word.end_time.nanos
-            )
-        )
-
-
-# [END speech_transcribe_async_word_time_offsets_gcs]
-
-
-def main():
-    import argparse
-
-    parser = argparse.ArgumentParser()
-    parser.add_argument(
-        "--storage_uri",
-        type=str,
-        default="gs://cloud-samples-data/speech/brooklyn_bridge.flac",
-    )
-    args = parser.parse_args()
-
-    sample_long_running_recognize(args.storage_uri)
-
-
-if __name__ == "__main__":
-    main()
diff --git a/samples/v1/speech_transcribe_enhanced_model.py b/samples/v1/speech_transcribe_enhanced_model.py
deleted file mode 100644
index 72df9a93..00000000
--- a/samples/v1/speech_transcribe_enhanced_model.py
+++ /dev/null
@@ -1,84 +0,0 @@
-# -*- coding: utf-8 -*-
-#
-# Copyright 2020 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-# DO NOT EDIT! This is a generated sample ("Request",  "speech_transcribe_enhanced_model")
-
-# To install the latest published package dependency, execute the following:
-#   pip install google-cloud-speech
-
-# sample-metadata
-#   title: Using Enhanced Models (Local File)
-#   description: Transcribe a short audio file using an enhanced model
-#   usage: python3 samples/v1/speech_transcribe_enhanced_model.py [--local_file_path "resources/hello.wav"]
-
-# [START speech_transcribe_enhanced_model]
-from google.cloud import speech_v1
-import io
-
-
-def sample_recognize(local_file_path):
-    """
-    Transcribe a short audio file using an enhanced model
-
-    Args:
-      local_file_path Path to local audio file, e.g. /path/audio.wav
-    """
-
-    client = speech_v1.SpeechClient()
-
-    # local_file_path = 'resources/hello.wav'
-
-    # The enhanced model to use, e.g. phone_call
-    model = "phone_call"
-
-    # Use an enhanced model for speech recognition (when set to true).
-    # Project must be eligible for requesting enhanced models.
-    # Enhanced speech models require that you opt-in to data logging.
-    use_enhanced = True
-
-    # The language of the supplied audio
-    language_code = "en-US"
-    config = {
-        "model": model,
-        "use_enhanced": use_enhanced,
-        "language_code": language_code,
-    }
-    with io.open(local_file_path, "rb") as f:
-        content = f.read()
-    audio = {"content": content}
-
-    response = client.recognize(config, audio)
-    for result in response.results:
-        # First alternative is the most probable result
-        alternative = result.alternatives[0]
-        print(u"Transcript: {}".format(alternative.transcript))
-
-
-# [END speech_transcribe_enhanced_model]
-
-
-def main():
-    import argparse
-
-    parser = argparse.ArgumentParser()
-    parser.add_argument("--local_file_path", type=str, default="resources/hello.wav")
-    args = parser.parse_args()
-
-    sample_recognize(args.local_file_path)
-
-
-if __name__ == "__main__":
-    main()
diff --git a/samples/v1/speech_transcribe_model_selection.py b/samples/v1/speech_transcribe_model_selection.py
deleted file mode 100644
index aed61cfc..00000000
--- a/samples/v1/speech_transcribe_model_selection.py
+++ /dev/null
@@ -1,77 +0,0 @@
-# -*- coding: utf-8 -*-
-#
-# Copyright 2020 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-# DO NOT EDIT! This is a generated sample ("Request",  "speech_transcribe_model_selection")
-
-# To install the latest published package dependency, execute the following:
-#   pip install google-cloud-speech
-
-# sample-metadata
-#   title: Selecting a Transcription Model (Local File)
-#   description: Transcribe a short audio file using a specified transcription model
-#   usage: python3 samples/v1/speech_transcribe_model_selection.py [--local_file_path "resources/hello.wav"] [--model "phone_call"]
-
-# [START speech_transcribe_model_selection]
-from google.cloud import speech_v1
-import io
-
-
-def sample_recognize(local_file_path, model):
-    """
-    Transcribe a short audio file using a specified transcription model
-
-    Args:
-      local_file_path Path to local audio file, e.g. /path/audio.wav
-      model The transcription model to use, e.g. video, phone_call, default
-      For a list of available transcription models, see:
-      https://cloud.google.com/speech-to-text/docs/transcription-model#transcription_models
-    """
-
-    client = speech_v1.SpeechClient()
-
-    # local_file_path = 'resources/hello.wav'
-    # model = 'phone_call'
-
-    # The language of the supplied audio
-    language_code = "en-US"
-    config = {"model": model, "language_code": language_code}
-    with io.open(local_file_path, "rb") as f:
-        content = f.read()
-    audio = {"content": content}
-
-    response = client.recognize(config, audio)
-    for result in response.results:
-        # First alternative is the most probable result
-        alternative = result.alternatives[0]
-        print(u"Transcript: {}".format(alternative.transcript))
-
-
-# [END speech_transcribe_model_selection]
-
-
-def main():
-    import argparse
-
-    parser = argparse.ArgumentParser()
-    parser.add_argument("--local_file_path", type=str, default="resources/hello.wav")
-    parser.add_argument("--model", type=str, default="phone_call")
-    args = parser.parse_args()
-
-    sample_recognize(args.local_file_path, args.model)
-
-
-if __name__ == "__main__":
-    main()
diff --git a/samples/v1/speech_transcribe_model_selection_gcs.py b/samples/v1/speech_transcribe_model_selection_gcs.py
deleted file mode 100644
index 813132dc..00000000
--- a/samples/v1/speech_transcribe_model_selection_gcs.py
+++ /dev/null
@@ -1,78 +0,0 @@
-# -*- coding: utf-8 -*-
-#
-# Copyright 2020 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-# DO NOT EDIT! This is a generated sample ("Request",  "speech_transcribe_model_selection_gcs")
-
-# To install the latest published package dependency, execute the following:
-#   pip install google-cloud-speech
-
-# sample-metadata
-#   title: Selecting a Transcription Model (Cloud Storage)
-#   description: Transcribe a short audio file from Cloud Storage using a specified
-#     transcription model
-#   usage: python3 samples/v1/speech_transcribe_model_selection_gcs.py [--storage_uri "gs://cloud-samples-data/speech/hello.wav"] [--model "phone_call"]
-
-# [START speech_transcribe_model_selection_gcs]
-from google.cloud import speech_v1
-
-
-def sample_recognize(storage_uri, model):
-    """
-    Transcribe a short audio file from Cloud Storage using a specified
-    transcription model
-
-    Args:
-      storage_uri URI for audio file in Cloud Storage, e.g. gs://[BUCKET]/[FILE]
-      model The transcription model to use, e.g. video, phone_call, default
-      For a list of available transcription models, see:
-      https://cloud.google.com/speech-to-text/docs/transcription-model#transcription_models
-    """
-
-    client = speech_v1.SpeechClient()
-
-    # storage_uri = 'gs://cloud-samples-data/speech/hello.wav'
-    # model = 'phone_call'
-
-    # The language of the supplied audio
-    language_code = "en-US"
-    config = {"model": model, "language_code": language_code}
-    audio = {"uri": storage_uri}
-
-    response = client.recognize(config, audio)
-    for result in response.results:
-        # First alternative is the most probable result
-        alternative = result.alternatives[0]
-        print(u"Transcript: {}".format(alternative.transcript))
-
-
-# [END speech_transcribe_model_selection_gcs]
-
-
-def main():
-    import argparse
-
-    parser = argparse.ArgumentParser()
-    parser.add_argument(
-        "--storage_uri", type=str, default="gs://cloud-samples-data/speech/hello.wav"
-    )
-    parser.add_argument("--model", type=str, default="phone_call")
-    args = parser.parse_args()
-
-    sample_recognize(args.storage_uri, args.model)
-
-
-if __name__ == "__main__":
-    main()
diff --git a/samples/v1/speech_transcribe_multichannel.py b/samples/v1/speech_transcribe_multichannel.py
deleted file mode 100644
index 68864888..00000000
--- a/samples/v1/speech_transcribe_multichannel.py
+++ /dev/null
@@ -1,86 +0,0 @@
-# -*- coding: utf-8 -*-
-#
-# Copyright 2020 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-# DO NOT EDIT! This is a generated sample ("Request",  "speech_transcribe_multichannel")
-
-# To install the latest published package dependency, execute the following:
-#   pip install google-cloud-speech
-
-# sample-metadata
-#   title: Multi-Channel Audio Transcription (Local File)
-#   description: Transcribe a short audio file with multiple channels
-#   usage: python3 samples/v1/speech_transcribe_multichannel.py [--local_file_path "resources/multi.wav"]
-
-# [START speech_transcribe_multichannel]
-from google.cloud import speech_v1
-import io
-
-
-def sample_recognize(local_file_path):
-    """
-    Transcribe a short audio file with multiple channels
-
-    Args:
-      local_file_path Path to local audio file, e.g. /path/audio.wav
-    """
-
-    client = speech_v1.SpeechClient()
-
-    # local_file_path = 'resources/multi.wav'
-
-    # The number of channels in the input audio file (optional)
-    audio_channel_count = 2
-
-    # When set to true, each audio channel will be recognized separately.
-    # The recognition result will contain a channel_tag field to state which
-    # channel that result belongs to
-    enable_separate_recognition_per_channel = True
-
-    # The language of the supplied audio
-    language_code = "en-US"
-    config = {
-        "audio_channel_count": audio_channel_count,
-        "enable_separate_recognition_per_channel": enable_separate_recognition_per_channel,
-        "language_code": language_code,
-    }
-    with io.open(local_file_path, "rb") as f:
-        content = f.read()
-    audio = {"content": content}
-
-    response = client.recognize(config, audio)
-    for result in response.results:
-        # channel_tag to recognize which audio channel this result is for
-        print(u"Channel tag: {}".format(result.channel_tag))
-        # First alternative is the most probable result
-        alternative = result.alternatives[0]
-        print(u"Transcript: {}".format(alternative.transcript))
-
-
-# [END speech_transcribe_multichannel]
-
-
-def main():
-    import argparse
-
-    parser = argparse.ArgumentParser()
-    parser.add_argument("--local_file_path", type=str, default="resources/multi.wav")
-    args = parser.parse_args()
-
-    sample_recognize(args.local_file_path)
-
-
-if __name__ == "__main__":
-    main()
diff --git a/samples/v1/speech_transcribe_multichannel_gcs.py b/samples/v1/speech_transcribe_multichannel_gcs.py
deleted file mode 100644
index 9285a5e1..00000000
--- a/samples/v1/speech_transcribe_multichannel_gcs.py
+++ /dev/null
@@ -1,85 +0,0 @@
-# -*- coding: utf-8 -*-
-#
-# Copyright 2020 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-# DO NOT EDIT! This is a generated sample ("Request",  "speech_transcribe_multichannel_gcs")
-
-# To install the latest published package dependency, execute the following:
-#   pip install google-cloud-speech
-
-# sample-metadata
-#   title: Multi-Channel Audio Transcription (Cloud Storage)
-#   description: Transcribe a short audio file from Cloud Storage with multiple channels
-#   usage: python3 samples/v1/speech_transcribe_multichannel_gcs.py [--storage_uri "gs://cloud-samples-data/speech/multi.wav"]
-
-# [START speech_transcribe_multichannel_gcs]
-from google.cloud import speech_v1
-
-
-def sample_recognize(storage_uri):
-    """
-    Transcribe a short audio file from Cloud Storage with multiple channels
-
-    Args:
-      storage_uri URI for audio file in Cloud Storage, e.g. gs://[BUCKET]/[FILE]
-    """
-
-    client = speech_v1.SpeechClient()
-
-    # storage_uri = 'gs://cloud-samples-data/speech/multi.wav'
-
-    # The number of channels in the input audio file (optional)
-    audio_channel_count = 2
-
-    # When set to true, each audio channel will be recognized separately.
-    # The recognition result will contain a channel_tag field to state which
-    # channel that result belongs to
-    enable_separate_recognition_per_channel = True
-
-    # The language of the supplied audio
-    language_code = "en-US"
-    config = {
-        "audio_channel_count": audio_channel_count,
-        "enable_separate_recognition_per_channel": enable_separate_recognition_per_channel,
-        "language_code": language_code,
-    }
-    audio = {"uri": storage_uri}
-
-    response = client.recognize(config, audio)
-    for result in response.results:
-        # channel_tag to recognize which audio channel this result is for
-        print(u"Channel tag: {}".format(result.channel_tag))
-        # First alternative is the most probable result
-        alternative = result.alternatives[0]
-        print(u"Transcript: {}".format(alternative.transcript))
-
-
-# [END speech_transcribe_multichannel_gcs]
-
-
-def main():
-    import argparse
-
-    parser = argparse.ArgumentParser()
-    parser.add_argument(
-        "--storage_uri", type=str, default="gs://cloud-samples-data/speech/multi.wav"
-    )
-    args = parser.parse_args()
-
-    sample_recognize(args.storage_uri)
-
-
-if __name__ == "__main__":
-    main()
diff --git a/samples/v1/speech_transcribe_sync.py b/samples/v1/speech_transcribe_sync.py
deleted file mode 100644
index 8ba8f4a9..00000000
--- a/samples/v1/speech_transcribe_sync.py
+++ /dev/null
@@ -1,86 +0,0 @@
-# -*- coding: utf-8 -*-
-#
-# Copyright 2020 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-# DO NOT EDIT! This is a generated sample ("Request",  "speech_transcribe_sync")
-
-# To install the latest published package dependency, execute the following:
-#   pip install google-cloud-speech
-
-# sample-metadata
-#   title: Transcribe Audio File (Local File)
-#   description: Transcribe a short audio file using synchronous speech recognition
-#   usage: python3 samples/v1/speech_transcribe_sync.py [--local_file_path "resources/brooklyn_bridge.raw"]
-
-# [START speech_transcribe_sync]
-from google.cloud import speech_v1
-from google.cloud.speech_v1 import enums
-import io
-
-
-def sample_recognize(local_file_path):
-    """
-    Transcribe a short audio file using synchronous speech recognition
-
-    Args:
-      local_file_path Path to local audio file, e.g. /path/audio.wav
-    """
-
-    client = speech_v1.SpeechClient()
-
-    # local_file_path = 'resources/brooklyn_bridge.raw'
-
-    # The language of the supplied audio
-    language_code = "en-US"
-
-    # Sample rate in Hertz of the audio data sent
-    sample_rate_hertz = 16000
-
-    # Encoding of audio data sent. This sample sets this explicitly.
-    # This field is optional for FLAC and WAV audio formats.
-    encoding = enums.RecognitionConfig.AudioEncoding.LINEAR16
-    config = {
-        "language_code": language_code,
-        "sample_rate_hertz": sample_rate_hertz,
-        "encoding": encoding,
-    }
-    with io.open(local_file_path, "rb") as f:
-        content = f.read()
-    audio = {"content": content}
-
-    response = client.recognize(config, audio)
-    for result in response.results:
-        # First alternative is the most probable result
-        alternative = result.alternatives[0]
-        print(u"Transcript: {}".format(alternative.transcript))
-
-
-# [END speech_transcribe_sync]
-
-
-def main():
-    import argparse
-
-    parser = argparse.ArgumentParser()
-    parser.add_argument(
-        "--local_file_path", type=str, default="resources/brooklyn_bridge.raw"
-    )
-    args = parser.parse_args()
-
-    sample_recognize(args.local_file_path)
-
-
-if __name__ == "__main__":
-    main()
diff --git a/samples/v1/speech_transcribe_sync_gcs.py b/samples/v1/speech_transcribe_sync_gcs.py
deleted file mode 100644
index 5afe26ff..00000000
--- a/samples/v1/speech_transcribe_sync_gcs.py
+++ /dev/null
@@ -1,87 +0,0 @@
-# -*- coding: utf-8 -*-
-#
-# Copyright 2020 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-# DO NOT EDIT! This is a generated sample ("Request",  "speech_transcribe_sync_gcs")
-
-# To install the latest published package dependency, execute the following:
-#   pip install google-cloud-speech
-
-# sample-metadata
-#   title: Transcript Audio File (Cloud Storage)
-#   description: Transcribe short audio file from Cloud Storage using synchronous speech
-#     recognition
-#   usage: python3 samples/v1/speech_transcribe_sync_gcs.py [--storage_uri "gs://cloud-samples-data/speech/brooklyn_bridge.raw"]
-
-# [START speech_transcribe_sync_gcs]
-from google.cloud import speech_v1
-from google.cloud.speech_v1 import enums
-
-
-def sample_recognize(storage_uri):
-    """
-    Transcribe short audio file from Cloud Storage using synchronous speech
-    recognition
-
-    Args:
-      storage_uri URI for audio file in Cloud Storage, e.g. gs://[BUCKET]/[FILE]
-    """
-
-    client = speech_v1.SpeechClient()
-
-    # storage_uri = 'gs://cloud-samples-data/speech/brooklyn_bridge.raw'
-
-    # Sample rate in Hertz of the audio data sent
-    sample_rate_hertz = 16000
-
-    # The language of the supplied audio
-    language_code = "en-US"
-
-    # Encoding of audio data sent. This sample sets this explicitly.
-    # This field is optional for FLAC and WAV audio formats.
-    encoding = enums.RecognitionConfig.AudioEncoding.LINEAR16
-    config = {
-        "sample_rate_hertz": sample_rate_hertz,
-        "language_code": language_code,
-        "encoding": encoding,
-    }
-    audio = {"uri": storage_uri}
-
-    response = client.recognize(config, audio)
-    for result in response.results:
-        # First alternative is the most probable result
-        alternative = result.alternatives[0]
-        print(u"Transcript: {}".format(alternative.transcript))
-
-
-# [END speech_transcribe_sync_gcs]
-
-
-def main():
-    import argparse
-
-    parser = argparse.ArgumentParser()
-    parser.add_argument(
-        "--storage_uri",
-        type=str,
-        default="gs://cloud-samples-data/speech/brooklyn_bridge.raw",
-    )
-    args = parser.parse_args()
-
-    sample_recognize(args.storage_uri)
-
-
-if __name__ == "__main__":
-    main()
diff --git a/samples/v1/test/samples.manifest.yaml b/samples/v1/test/samples.manifest.yaml
deleted file mode 100644
index f989660c..00000000
--- a/samples/v1/test/samples.manifest.yaml
+++ /dev/null
@@ -1,38 +0,0 @@
-type: manifest/samples
-schema_version: 3
-base: &common
-  env: 'python'
-  bin: 'python3'
-  chdir: '{@manifest_dir}/../..'
-  basepath: '.'
-samples:
-- <<: *common
-  path: '{basepath}/v1/speech_transcribe_async.py'
-  sample: 'speech_transcribe_async'
-- <<: *common
-  path: '{basepath}/v1/speech_transcribe_async_gcs.py'
-  sample: 'speech_transcribe_async_gcs'
-- <<: *common
-  path: '{basepath}/v1/speech_transcribe_async_word_time_offsets_gcs.py'
-  sample: 'speech_transcribe_async_word_time_offsets_gcs'
-- <<: *common
-  path: '{basepath}/v1/speech_transcribe_enhanced_model.py'
-  sample: 'speech_transcribe_enhanced_model'
-- <<: *common
-  path: '{basepath}/v1/speech_transcribe_model_selection.py'
-  sample: 'speech_transcribe_model_selection'
-- <<: *common
-  path: '{basepath}/v1/speech_transcribe_model_selection_gcs.py'
-  sample: 'speech_transcribe_model_selection_gcs'
-- <<: *common
-  path: '{basepath}/v1/speech_transcribe_multichannel.py'
-  sample: 'speech_transcribe_multichannel'
-- <<: *common
-  path: '{basepath}/v1/speech_transcribe_multichannel_gcs.py'
-  sample: 'speech_transcribe_multichannel_gcs'
-- <<: *common
-  path: '{basepath}/v1/speech_transcribe_sync.py'
-  sample: 'speech_transcribe_sync'
-- <<: *common
-  path: '{basepath}/v1/speech_transcribe_sync_gcs.py'
-  sample: 'speech_transcribe_sync_gcs'
diff --git a/samples/v1/test/speech_transcribe_async.test.yaml b/samples/v1/test/speech_transcribe_async.test.yaml
deleted file mode 100644
index f26cfbab..00000000
--- a/samples/v1/test/speech_transcribe_async.test.yaml
+++ /dev/null
@@ -1,28 +0,0 @@
-type: test/samples
-schema_version: 1
-test:
-  suites:
-  - name: Transcribe Audio File using Long Running Operation (Local File) (LRO)
-    cases:
-
-    # This sample should default to using brooklyn_bridge.raw
-    # with explicitly configured sample_rate_hertz and encoding
-    - name: speech_transcribe_async (no arguments)
-      spec:
-      - call:
-          sample: speech_transcribe_async
-      - assert_contains:
-        - literal: "how old is the Brooklyn Bridge"
-
-    # Confirm that another file can be transcribed (use another .raw PCM file)
-    - name: speech_transcribe_async (--local_file_path)
-      spec:
-      - call:
-          sample: speech_transcribe_async
-          params:
-            local_file_path:
-              literal: "resources/hello.raw"
-      - assert_contains:
-        - literal: "hello"
-      - assert_not_contains:
-        - literal: "how old is the Brooklyn Bridge"
diff --git a/samples/v1/test/speech_transcribe_async_gcs.test.yaml b/samples/v1/test/speech_transcribe_async_gcs.test.yaml
deleted file mode 100644
index d3d83133..00000000
--- a/samples/v1/test/speech_transcribe_async_gcs.test.yaml
+++ /dev/null
@@ -1,28 +0,0 @@
-type: test/samples
-schema_version: 1
-test:
-  suites:
-  - name: Transcript Audio File using Long Running Operation (Cloud Storage) (LRO)
-    cases:
-
-    # This sample should default to using gs://cloud-samples-data/brooklyn_bridge.raw
-    # with explicitly configured sample_rate_hertz and encoding
-    - name: speech_transcribe_async_gcs (no arguments)
-      spec:
-      - call:
-          sample: speech_transcribe_async_gcs
-      - assert_contains:
-        - literal: "how old is the Brooklyn Bridge"
-
-    # Confirm that another file can be transcribed (use another .raw PCM file)
-    - name: speech_transcribe_async_gcs (--storage_uri)
-      spec:
-      - call:
-          sample: speech_transcribe_async_gcs
-          params:
-            storage_uri:
-              literal: "gs://cloud-samples-data/speech/hello.raw"
-      - assert_contains:
-        - literal: "hello"
-      - assert_not_contains:
-        - literal: "how old is the Brooklyn Bridge"
diff --git a/samples/v1/test/speech_transcribe_async_word_time_offsets_gcs.test.yaml b/samples/v1/test/speech_transcribe_async_word_time_offsets_gcs.test.yaml
deleted file mode 100644
index 11784726..00000000
--- a/samples/v1/test/speech_transcribe_async_word_time_offsets_gcs.test.yaml
+++ /dev/null
@@ -1,37 +0,0 @@
-type: test/samples
-schema_version: 1
-test:
-  suites:
-  - name: Getting word timestamps (Cloud Storage) (LRO)
-    cases:
-
-    # This sample should default to using gs://cloud-samples-data/speech/brooklyn_bridge.flac
-    - name: speech_transcribe_async_word_time_offsets_gcs (no arguments)
-      spec:
-      - call:
-          sample: speech_transcribe_async_word_time_offsets_gcs
-      - assert_contains:
-        - literal: "how old is the Brooklyn Bridge"
-        - literal: "Word: how"
-        - literal: "Word: old"
-        - literal: "Word: is"
-        - literal: "Start time: 0 seconds"
-        - literal: "End time: 1 seconds"
-
-    # Confirm that another file can be transcribed (use another file)
-    - name: speech_transcribe_async_word_time_offsets_gcs (--storage_uri)
-      spec:
-      - call:
-          sample: speech_transcribe_async_word_time_offsets_gcs
-          params:
-            storage_uri:
-              literal: "gs://cloud-samples-data/speech/multi.flac"
-      - assert_contains:
-        - literal: "how are you doing"
-        - literal: "Word: how"
-        - literal: "Word: are"
-        - literal: "Word: you"
-        - literal: "Start time: 0 seconds"
-        - literal: "End time: 1 seconds"
-      - assert_not_contains:
-        - literal: "how old is the Brooklyn Bridge"
diff --git a/samples/v1/test/speech_transcribe_enhanced_model.test.yaml b/samples/v1/test/speech_transcribe_enhanced_model.test.yaml
deleted file mode 100644
index 6eab33b5..00000000
--- a/samples/v1/test/speech_transcribe_enhanced_model.test.yaml
+++ /dev/null
@@ -1,29 +0,0 @@
-type: test/samples
-schema_version: 1
-test:
-  suites:
-  - name: Using Enhanced Models (Local File)
-    cases:
-
-    # This sample should default to using hello.wav
-    # and the phone_call model (only currently available enhanced model)
-    #
-    # Note: if the project used to run these tests isn't eligible for
-    #       enhanced models, you will receive an error.
-    - name: speech_transcribe_enhanced_model (no arguments)
-      spec:
-      - call:
-          sample: speech_transcribe_enhanced_model
-      - assert_contains:
-        - literal: "hello"
-
-    # Confirm that another file can be transcribed (use another .wav file)
-    - name: speech_transcribe_enhanced_model (--local_file_path)
-      spec:
-      - call:
-          sample: speech_transcribe_enhanced_model
-          params:
-            local_file_path:
-              literal: "resources/commercial_mono.wav"
-      - assert_contains:
-        - literal: "Chrome"
diff --git a/samples/v1/test/speech_transcribe_model_selection.test.yaml b/samples/v1/test/speech_transcribe_model_selection.test.yaml
deleted file mode 100644
index b5ec2d90..00000000
--- a/samples/v1/test/speech_transcribe_model_selection.test.yaml
+++ /dev/null
@@ -1,52 +0,0 @@
-type: test/samples
-schema_version: 1
-test:
-  suites:
-  - name: Selecting a Transcription Model (Local File)
-    cases:
-
-    # This sample should default to using hello.wav
-    # and the phone_call model
-    - name: speech_transcribe_model_selection (no arguments)
-      spec:
-      - call:
-          sample: speech_transcribe_model_selection
-      - assert_contains:
-        - literal: "Hello"
-
-    # Confirm that another file can be transcribed (use another .wav file)
-    - name: speech_transcribe_model_selection (--local_file_path)
-      spec:
-      - call:
-          sample: speech_transcribe_model_selection
-          params:
-            local_file_path:
-              literal: "resources/commercial_mono.wav"
-      - assert_contains:
-        - literal: "Chrome"
-
-    # Confirm that --model can be specified and the sample does not blow up
-    #
-    # Note: we are not using example audio files which result in deterministically
-    #       different results when using different models. so we simply test
-    #       that regular transcription continues to work.
-    - name: speech_transcribe_model_selection (--model)
-      spec:
-      - call:
-          sample: speech_transcribe_model_selection
-          params:
-            model:
-              literal: video
-      - assert_contains:
-        - literal: "hello"
-
-    # Confirm that --model is being passed through by providing an invalid model
-    - name: speech_transcribe_model_selection (invalid --model)
-      spec:
-      - call_may_fail:
-          sample: speech_transcribe_model_selection
-          params:
-            model:
-              literal: I_DONT_EXIST
-      - assert_contains:
-        - literal: "Incorrect model specified"
diff --git a/samples/v1/test/speech_transcribe_model_selection_gcs.test.yaml b/samples/v1/test/speech_transcribe_model_selection_gcs.test.yaml
deleted file mode 100644
index 60c45c97..00000000
--- a/samples/v1/test/speech_transcribe_model_selection_gcs.test.yaml
+++ /dev/null
@@ -1,52 +0,0 @@
-type: test/samples
-schema_version: 1
-test:
-  suites:
-  - name: Selecting a Transcription Model (Cloud Storage)
-    cases:
-
-    # This sample should default to using gs://cloud-samples-data/speech/hello.wav
-    # and the phone_call model
-    - name: speech_transcribe_model_selection_gcs (no arguments)
-      spec:
-      - call:
-          sample: speech_transcribe_model_selection_gcs
-      - assert_contains:
-        - literal: "Hello"
-
-    # Confirm that another file can be transcribed (use another .wav file)
-    - name: speech_transcribe_model_selection_gcs (--local_file_path)
-      spec:
-      - call:
-          sample: speech_transcribe_model_selection_gcs
-          params:
-            storage_uri:
-              literal: "gs://cloud-samples-data/speech/commercial_mono.wav"
-      - assert_contains:
-        - literal: "Chrome"
-
-    # Confirm that --model can be specified and the sample does not blow up
-    #
-    # Note: we are not using example audio files which result in deterministically
-    #       different results when using different models. so we simply test
-    #       that regular transcription continues to work.
-    - name: speech_transcribe_model_selection_gcs (--model)
-      spec:
-      - call:
-          sample: speech_transcribe_model_selection_gcs
-          params:
-            model:
-              literal: video
-      - assert_contains:
-        - literal: "hello"
-
-    # Confirm that --model is being passed through by providing an invalid model
-    - name: speech_transcribe_model_selection_gcs (invalid --model)
-      spec:
-      - call_may_fail:
-          sample: speech_transcribe_model_selection_gcs
-          params:
-            model:
-              literal: I_DONT_EXIST
-      - assert_contains:
-        - literal: "Incorrect model specified"
diff --git a/samples/v1/test/speech_transcribe_multichannel.test.yaml b/samples/v1/test/speech_transcribe_multichannel.test.yaml
deleted file mode 100644
index 9d5379dc..00000000
--- a/samples/v1/test/speech_transcribe_multichannel.test.yaml
+++ /dev/null
@@ -1,31 +0,0 @@
-type: test/samples
-schema_version: 1
-test:
-  suites:
-  - name: Multi-Channel Audio Transcription (Local File)
-    cases:
-
-    # This sample should default to using multi.wav (2 channels)
-    - name: speech_transcribe_multichannel (no arguments)
-      spec:
-      - call:
-          sample: speech_transcribe_multichannel
-      - assert_contains:
-        - literal: "Channel tag: 1"
-        - literal: "Channel tag: 2"
-        - literal: "how are you doing"
-
-    # Confirm that another file can be transcribed (use another 2 channel .wav file)
-    - name: speech_transcribe_multichannel (--local_file_path)
-      spec:
-      - call:
-          sample: speech_transcribe_multichannel
-          params:
-            local_file_path:
-              literal: "resources/brooklyn_bridge.wav"
-      - assert_contains:
-        # Only one channel of data is present in brooklyn_bridge.wav
-        - literal: "Channel tag:"
-        - literal: "how old is the Brooklyn Bridge"
-      - assert_not_contains:
-        - literal: "how are you doing"
diff --git a/samples/v1/test/speech_transcribe_multichannel_gcs.test.yaml b/samples/v1/test/speech_transcribe_multichannel_gcs.test.yaml
deleted file mode 100644
index 64c9340c..00000000
--- a/samples/v1/test/speech_transcribe_multichannel_gcs.test.yaml
+++ /dev/null
@@ -1,32 +0,0 @@
-type: test/samples
-schema_version: 1
-test:
-  suites:
-  - name: Multi-Channel Audio Transcription (Cloud Storage)
-    cases:
-
-    # This sample should default to using gs://cloud-samples-data/speech/multi.wav
-    # with 2 audio channels of data
-    - name: speech_transcribe_multichannel_gcs (no arguments)
-      spec:
-      - call:
-          sample: speech_transcribe_multichannel_gcs
-      - assert_contains:
-        - literal: "Channel tag: 1"
-        - literal: "Channel tag: 2"
-        - literal: "how are you doing"
-
-    # Confirm that another file can be transcribed (use another 2 channel .wav file)
-    - name: speech_transcribe_multichannel_gcs (--storage_uri)
-      spec:
-      - call:
-          sample: speech_transcribe_multichannel_gcs
-          params:
-            storage_uri:
-              literal: "gs://cloud-samples-data/speech/brooklyn_bridge.wav"
-      - assert_contains:
-        # Only one channel of data is present in brooklyn_bridge.wav
-        - literal: "Channel tag:"
-        - literal: "how old is the Brooklyn Bridge"
-      - assert_not_contains:
-        - literal: "how are you doing"
diff --git a/samples/v1/test/speech_transcribe_sync.test.yaml b/samples/v1/test/speech_transcribe_sync.test.yaml
deleted file mode 100644
index 47cc8c1a..00000000
--- a/samples/v1/test/speech_transcribe_sync.test.yaml
+++ /dev/null
@@ -1,28 +0,0 @@
-type: test/samples
-schema_version: 1
-test:
-  suites:
-  - name: Transcribe Audio File (Local File)
-    cases:
-
-    # This sample should default to using brooklyn_bridge.raw
-    # with explicitly configured sample_rate_hertz and encoding
-    - name: speech_transcribe_sync (no arguments)
-      spec:
-      - call:
-          sample: speech_transcribe_sync
-      - assert_contains:
-        - literal: "how old is the Brooklyn Bridge"
-
-    # Confirm that another file can be transcribed (use another .raw PCM file)
-    - name: speech_transcribe_sync (--local_file_path)
-      spec:
-      - call:
-          sample: speech_transcribe_sync
-          params:
-            local_file_path:
-              literal: "resources/hello.raw"
-      - assert_contains:
-        - literal: "hello"
-      - assert_not_contains:
-        - literal: "how old is the Brooklyn Bridge"
diff --git a/samples/v1/test/speech_transcribe_sync_gcs.test.yaml b/samples/v1/test/speech_transcribe_sync_gcs.test.yaml
deleted file mode 100644
index 3defdf28..00000000
--- a/samples/v1/test/speech_transcribe_sync_gcs.test.yaml
+++ /dev/null
@@ -1,28 +0,0 @@
-type: test/samples
-schema_version: 1
-test:
-  suites:
-  - name: Transcript Audio File (Cloud Storage)
-    cases:
-
-    # This sample should default to using gs://cloud-samples-data/speech/brooklyn_bridge.raw
-    # with explicitly configured sample_rate_hertz and encoding
-    - name: speech_transcribe_sync_gcs (no arguments)
-      spec:
-      - call:
-          sample: speech_transcribe_sync_gcs
-      - assert_contains:
-        - literal: "how old is the Brooklyn Bridge"
-
-    # Confirm that another file can be transcribed (use another .raw PCM file)
-    - name: speech_transcribe_sync_gcs (--storage_uri)
-      spec:
-      - call:
-          sample: speech_transcribe_sync_gcs
-          params:
-            storage_uri:
-              literal: "gs://cloud-samples-data/speech/hello.raw"
-      - assert_contains:
-        - literal: "hello"
-      - assert_not_contains:
-        - literal: "how old is the Brooklyn Bridge"
diff --git a/samples/v1p1beta1/speech_adaptation_beta.py b/samples/v1p1beta1/speech_adaptation_beta.py
deleted file mode 100644
index 364bddf2..00000000
--- a/samples/v1p1beta1/speech_adaptation_beta.py
+++ /dev/null
@@ -1,100 +0,0 @@
-# -*- coding: utf-8 -*-
-#
-# Copyright 2020 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-# DO NOT EDIT! This is a generated sample ("Request",  "speech_adaptation_beta")
-
-# To install the latest published package dependency, execute the following:
-#   pip install google-cloud-speech
-
-# sample-metadata
-#   title: Speech Adaptation (Cloud Storage)
-#   description: Transcribe a short audio file with speech adaptation.
-#   usage: python3 samples/v1p1beta1/speech_adaptation_beta.py [--storage_uri "gs://cloud-samples-data/speech/brooklyn_bridge.mp3"] [--phrase "Brooklyn Bridge"]
-
-# [START speech_adaptation_beta]
-from google.cloud import speech_v1p1beta1
-from google.cloud.speech_v1p1beta1 import enums
-
-
-def sample_recognize(storage_uri, phrase):
-    """
-    Transcribe a short audio file with speech adaptation.
-
-    Args:
-      storage_uri URI for audio file in Cloud Storage, e.g. gs://[BUCKET]/[FILE]
-      phrase Phrase "hints" help recognize the specified phrases from your audio.
-    """
-
-    client = speech_v1p1beta1.SpeechClient()
-
-    # storage_uri = 'gs://cloud-samples-data/speech/brooklyn_bridge.mp3'
-    # phrase = 'Brooklyn Bridge'
-    phrases = [phrase]
-
-    # Hint Boost. This value increases the probability that a specific
-    # phrase will be recognized over other similar sounding phrases.
-    # The higher the boost, the higher the chance of false positive
-    # recognition as well. Can accept wide range of positive values.
-    # Most use cases are best served with values between 0 and 20.
-    # Using a binary search happroach may help you find the optimal value.
-    boost = 20.0
-    speech_contexts_element = {"phrases": phrases, "boost": boost}
-    speech_contexts = [speech_contexts_element]
-
-    # Sample rate in Hertz of the audio data sent
-    sample_rate_hertz = 44100
-
-    # The language of the supplied audio
-    language_code = "en-US"
-
-    # Encoding of audio data sent. This sample sets this explicitly.
-    # This field is optional for FLAC and WAV audio formats.
-    encoding = enums.RecognitionConfig.AudioEncoding.MP3
-    config = {
-        "speech_contexts": speech_contexts,
-        "sample_rate_hertz": sample_rate_hertz,
-        "language_code": language_code,
-        "encoding": encoding,
-    }
-    audio = {"uri": storage_uri}
-
-    response = client.recognize(config, audio)
-    for result in response.results:
-        # First alternative is the most probable result
-        alternative = result.alternatives[0]
-        print(u"Transcript: {}".format(alternative.transcript))
-
-
-# [END speech_adaptation_beta]
-
-
-def main():
-    import argparse
-
-    parser = argparse.ArgumentParser()
-    parser.add_argument(
-        "--storage_uri",
-        type=str,
-        default="gs://cloud-samples-data/speech/brooklyn_bridge.mp3",
-    )
-    parser.add_argument("--phrase", type=str, default="Brooklyn Bridge")
-    args = parser.parse_args()
-
-    sample_recognize(args.storage_uri, args.phrase)
-
-
-if __name__ == "__main__":
-    main()
diff --git a/samples/v1p1beta1/speech_contexts_classes_beta.py b/samples/v1p1beta1/speech_contexts_classes_beta.py
deleted file mode 100644
index 04632d80..00000000
--- a/samples/v1p1beta1/speech_contexts_classes_beta.py
+++ /dev/null
@@ -1,93 +0,0 @@
-# -*- coding: utf-8 -*-
-#
-# Copyright 2020 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-# DO NOT EDIT! This is a generated sample ("Request",  "speech_contexts_classes_beta")
-
-# To install the latest published package dependency, execute the following:
-#   pip install google-cloud-speech
-
-# sample-metadata
-#   title: Using Context Classes (Cloud Storage)
-#   description: Transcribe a short audio file with static context classes.
-#   usage: python3 samples/v1p1beta1/speech_contexts_classes_beta.py [--storage_uri "gs://cloud-samples-data/speech/time.mp3"] [--phrase "$TIME"]
-
-# [START speech_contexts_classes_beta]
-from google.cloud import speech_v1p1beta1
-from google.cloud.speech_v1p1beta1 import enums
-
-
-def sample_recognize(storage_uri, phrase):
-    """
-    Transcribe a short audio file with static context classes.
-
-    Args:
-      storage_uri URI for audio file in Cloud Storage, e.g. gs://[BUCKET]/[FILE]
-      phrase Phrase "hints" help recognize the specified phrases from your audio.
-      In this sample we are using a static class phrase ($TIME).
-      Classes represent groups of words that represent common concepts
-      that occur in natural language.
-    """
-
-    client = speech_v1p1beta1.SpeechClient()
-
-    # storage_uri = 'gs://cloud-samples-data/speech/time.mp3'
-    # phrase = '$TIME'
-    phrases = [phrase]
-    speech_contexts_element = {"phrases": phrases}
-    speech_contexts = [speech_contexts_element]
-
-    # The language of the supplied audio
-    language_code = "en-US"
-
-    # Sample rate in Hertz of the audio data sent
-    sample_rate_hertz = 24000
-
-    # Encoding of audio data sent. This sample sets this explicitly.
-    # This field is optional for FLAC and WAV audio formats.
-    encoding = enums.RecognitionConfig.AudioEncoding.MP3
-    config = {
-        "speech_contexts": speech_contexts,
-        "language_code": language_code,
-        "sample_rate_hertz": sample_rate_hertz,
-        "encoding": encoding,
-    }
-    audio = {"uri": storage_uri}
-
-    response = client.recognize(config, audio)
-    for result in response.results:
-        # First alternative is the most probable result
-        alternative = result.alternatives[0]
-        print(u"Transcript: {}".format(alternative.transcript))
-
-
-# [END speech_contexts_classes_beta]
-
-
-def main():
-    import argparse
-
-    parser = argparse.ArgumentParser()
-    parser.add_argument(
-        "--storage_uri", type=str, default="gs://cloud-samples-data/speech/time.mp3"
-    )
-    parser.add_argument("--phrase", type=str, default="$TIME")
-    args = parser.parse_args()
-
-    sample_recognize(args.storage_uri, args.phrase)
-
-
-if __name__ == "__main__":
-    main()
diff --git a/samples/v1p1beta1/speech_quickstart_beta.py b/samples/v1p1beta1/speech_quickstart_beta.py
deleted file mode 100644
index 76b13a5e..00000000
--- a/samples/v1p1beta1/speech_quickstart_beta.py
+++ /dev/null
@@ -1,85 +0,0 @@
-# -*- coding: utf-8 -*-
-#
-# Copyright 2020 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-# DO NOT EDIT! This is a generated sample ("Request",  "speech_quickstart_beta")
-
-# To install the latest published package dependency, execute the following:
-#   pip install google-cloud-speech
-
-# sample-metadata
-#   title: Quickstart Beta
-#   description: Performs synchronous speech recognition on an audio file
-#   usage: python3 samples/v1p1beta1/speech_quickstart_beta.py [--storage_uri "gs://cloud-samples-data/speech/brooklyn_bridge.mp3"]
-
-# [START speech_quickstart_beta]
-from google.cloud import speech_v1p1beta1
-from google.cloud.speech_v1p1beta1 import enums
-
-
-def sample_recognize(storage_uri):
-    """
-    Performs synchronous speech recognition on an audio file
-
-    Args:
-      storage_uri URI for audio file in Cloud Storage, e.g. gs://[BUCKET]/[FILE]
-    """
-
-    client = speech_v1p1beta1.SpeechClient()
-
-    # storage_uri = 'gs://cloud-samples-data/speech/brooklyn_bridge.mp3'
-
-    # The language of the supplied audio
-    language_code = "en-US"
-
-    # Sample rate in Hertz of the audio data sent
-    sample_rate_hertz = 44100
-
-    # Encoding of audio data sent. This sample sets this explicitly.
-    # This field is optional for FLAC and WAV audio formats.
-    encoding = enums.RecognitionConfig.AudioEncoding.MP3
-    config = {
-        "language_code": language_code,
-        "sample_rate_hertz": sample_rate_hertz,
-        "encoding": encoding,
-    }
-    audio = {"uri": storage_uri}
-
-    response = client.recognize(config, audio)
-    for result in response.results:
-        # First alternative is the most probable result
-        alternative = result.alternatives[0]
-        print(u"Transcript: {}".format(alternative.transcript))
-
-
-# [END speech_quickstart_beta]
-
-
-def main():
-    import argparse
-
-    parser = argparse.ArgumentParser()
-    parser.add_argument(
-        "--storage_uri",
-        type=str,
-        default="gs://cloud-samples-data/speech/brooklyn_bridge.mp3",
-    )
-    args = parser.parse_args()
-
-    sample_recognize(args.storage_uri)
-
-
-if __name__ == "__main__":
-    main()
diff --git a/samples/v1p1beta1/speech_transcribe_auto_punctuation_beta.py b/samples/v1p1beta1/speech_transcribe_auto_punctuation_beta.py
deleted file mode 100644
index b012d42a..00000000
--- a/samples/v1p1beta1/speech_transcribe_auto_punctuation_beta.py
+++ /dev/null
@@ -1,82 +0,0 @@
-# -*- coding: utf-8 -*-
-#
-# Copyright 2020 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-# DO NOT EDIT! This is a generated sample ("Request",  "speech_transcribe_auto_punctuation_beta")
-
-# To install the latest published package dependency, execute the following:
-#   pip install google-cloud-speech
-
-# sample-metadata
-#   title: Getting punctuation in results (Local File) (Beta)
-#   description: Transcribe a short audio file with punctuation
-#   usage: python3 samples/v1p1beta1/speech_transcribe_auto_punctuation_beta.py [--local_file_path "resources/commercial_mono.wav"]
-
-# [START speech_transcribe_auto_punctuation_beta]
-from google.cloud import speech_v1p1beta1
-import io
-
-
-def sample_recognize(local_file_path):
-    """
-    Transcribe a short audio file with punctuation
-
-    Args:
-      local_file_path Path to local audio file, e.g. /path/audio.wav
-    """
-
-    client = speech_v1p1beta1.SpeechClient()
-
-    # local_file_path = 'resources/commercial_mono.wav'
-
-    # When enabled, trascription results may include punctuation
-    # (available for select languages).
-    enable_automatic_punctuation = True
-
-    # The language of the supplied audio. Even though additional languages are
-    # provided by alternative_language_codes, a primary language is still required.
-    language_code = "en-US"
-    config = {
-        "enable_automatic_punctuation": enable_automatic_punctuation,
-        "language_code": language_code,
-    }
-    with io.open(local_file_path, "rb") as f:
-        content = f.read()
-    audio = {"content": content}
-
-    response = client.recognize(config, audio)
-    for result in response.results:
-        # First alternative is the most probable result
-        alternative = result.alternatives[0]
-        print(u"Transcript: {}".format(alternative.transcript))
-
-
-# [END speech_transcribe_auto_punctuation_beta]
-
-
-def main():
-    import argparse
-
-    parser = argparse.ArgumentParser()
-    parser.add_argument(
-        "--local_file_path", type=str, default="resources/commercial_mono.wav"
-    )
-    args = parser.parse_args()
-
-    sample_recognize(args.local_file_path)
-
-
-if __name__ == "__main__":
-    main()
diff --git a/samples/v1p1beta1/speech_transcribe_diarization_beta.py b/samples/v1p1beta1/speech_transcribe_diarization_beta.py
deleted file mode 100644
index bdc8e3e1..00000000
--- a/samples/v1p1beta1/speech_transcribe_diarization_beta.py
+++ /dev/null
@@ -1,97 +0,0 @@
-# -*- coding: utf-8 -*-
-#
-# Copyright 2020 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-# DO NOT EDIT! This is a generated sample ("LongRunningPromise",  "speech_transcribe_diarization_beta")
-
-# To install the latest published package dependency, execute the following:
-#   pip install google-cloud-speech
-
-# sample-metadata
-#   title: Separating different speakers (Local File) (LRO) (Beta)
-#   description: Print confidence level for individual words in a transcription of a short audio
-#     file
-#     Separating different speakers in an audio file recording
-#   usage: python3 samples/v1p1beta1/speech_transcribe_diarization_beta.py [--local_file_path "resources/commercial_mono.wav"]
-
-# [START speech_transcribe_diarization_beta]
-from google.cloud import speech_v1p1beta1
-import io
-
-
-def sample_long_running_recognize(local_file_path):
-    """
-    Print confidence level for individual words in a transcription of a short audio
-    file
-    Separating different speakers in an audio file recording
-
-    Args:
-      local_file_path Path to local audio file, e.g. /path/audio.wav
-    """
-
-    client = speech_v1p1beta1.SpeechClient()
-
-    # local_file_path = 'resources/commercial_mono.wav'
-
-    # If enabled, each word in the first alternative of each result will be
-    # tagged with a speaker tag to identify the speaker.
-    enable_speaker_diarization = True
-
-    # Optional. Specifies the estimated number of speakers in the conversation.
-    diarization_speaker_count = 2
-
-    # The language of the supplied audio
-    language_code = "en-US"
-    config = {
-        "enable_speaker_diarization": enable_speaker_diarization,
-        "diarization_speaker_count": diarization_speaker_count,
-        "language_code": language_code,
-    }
-    with io.open(local_file_path, "rb") as f:
-        content = f.read()
-    audio = {"content": content}
-
-    operation = client.long_running_recognize(config, audio)
-
-    print(u"Waiting for operation to complete...")
-    response = operation.result()
-
-    for result in response.results:
-        # First alternative has words tagged with speakers
-        alternative = result.alternatives[0]
-        print(u"Transcript: {}".format(alternative.transcript))
-        # Print the speaker_tag of each word
-        for word in alternative.words:
-            print(u"Word: {}".format(word.word))
-            print(u"Speaker tag: {}".format(word.speaker_tag))
-
-
-# [END speech_transcribe_diarization_beta]
-
-
-def main():
-    import argparse
-
-    parser = argparse.ArgumentParser()
-    parser.add_argument(
-        "--local_file_path", type=str, default="resources/commercial_mono.wav"
-    )
-    args = parser.parse_args()
-
-    sample_long_running_recognize(args.local_file_path)
-
-
-if __name__ == "__main__":
-    main()
diff --git a/samples/v1p1beta1/speech_transcribe_multilanguage_beta.py b/samples/v1p1beta1/speech_transcribe_multilanguage_beta.py
deleted file mode 100644
index dde461fd..00000000
--- a/samples/v1p1beta1/speech_transcribe_multilanguage_beta.py
+++ /dev/null
@@ -1,91 +0,0 @@
-# -*- coding: utf-8 -*-
-#
-# Copyright 2020 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-# DO NOT EDIT! This is a generated sample ("Request",  "speech_transcribe_multilanguage_beta")
-
-# To install the latest published package dependency, execute the following:
-#   pip install google-cloud-speech
-
-# sample-metadata
-#   title: Detecting language spoken automatically (Local File) (Beta)
-#   description: Transcribe a short audio file with language detected from a list of possible
-#     languages
-#   usage: python3 samples/v1p1beta1/speech_transcribe_multilanguage_beta.py [--local_file_path "resources/brooklyn_bridge.flac"]
-
-# [START speech_transcribe_multilanguage_beta]
-from google.cloud import speech_v1p1beta1
-import io
-
-
-def sample_recognize(local_file_path):
-    """
-    Transcribe a short audio file with language detected from a list of possible
-    languages
-
-    Args:
-      local_file_path Path to local audio file, e.g. /path/audio.wav
-    """
-
-    client = speech_v1p1beta1.SpeechClient()
-
-    # local_file_path = 'resources/brooklyn_bridge.flac'
-
-    # The language of the supplied audio. Even though additional languages are
-    # provided by alternative_language_codes, a primary language is still required.
-    language_code = "fr"
-
-    # Specify up to 3 additional languages as possible alternative languages
-    # of the supplied audio.
-    alternative_language_codes_element = "es"
-    alternative_language_codes_element_2 = "en"
-    alternative_language_codes = [
-        alternative_language_codes_element,
-        alternative_language_codes_element_2,
-    ]
-    config = {
-        "language_code": language_code,
-        "alternative_language_codes": alternative_language_codes,
-    }
-    with io.open(local_file_path, "rb") as f:
-        content = f.read()
-    audio = {"content": content}
-
-    response = client.recognize(config, audio)
-    for result in response.results:
-        # The language_code which was detected as the most likely being spoken in the audio
-        print(u"Detected language: {}".format(result.language_code))
-        # First alternative is the most probable result
-        alternative = result.alternatives[0]
-        print(u"Transcript: {}".format(alternative.transcript))
-
-
-# [END speech_transcribe_multilanguage_beta]
-
-
-def main():
-    import argparse
-
-    parser = argparse.ArgumentParser()
-    parser.add_argument(
-        "--local_file_path", type=str, default="resources/brooklyn_bridge.flac"
-    )
-    args = parser.parse_args()
-
-    sample_recognize(args.local_file_path)
-
-
-if __name__ == "__main__":
-    main()
diff --git a/samples/v1p1beta1/speech_transcribe_recognition_metadata_beta.py b/samples/v1p1beta1/speech_transcribe_recognition_metadata_beta.py
deleted file mode 100644
index c11feb88..00000000
--- a/samples/v1p1beta1/speech_transcribe_recognition_metadata_beta.py
+++ /dev/null
@@ -1,92 +0,0 @@
-# -*- coding: utf-8 -*-
-#
-# Copyright 2020 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-# DO NOT EDIT! This is a generated sample ("Request",  "speech_transcribe_recognition_metadata_beta")
-
-# To install the latest published package dependency, execute the following:
-#   pip install google-cloud-speech
-
-# sample-metadata
-#   title: Adding recognition metadata (Local File) (Beta)
-#   description: Adds additional details short audio file included in this recognition request
-#   usage: python3 samples/v1p1beta1/speech_transcribe_recognition_metadata_beta.py [--local_file_path "resources/commercial_mono.wav"]
-
-# [START speech_transcribe_recognition_metadata_beta]
-from google.cloud import speech_v1p1beta1
-from google.cloud.speech_v1p1beta1 import enums
-import io
-
-
-def sample_recognize(local_file_path):
-    """
-    Adds additional details short audio file included in this recognition request
-
-    Args:
-      local_file_path Path to local audio file, e.g. /path/audio.wav
-    """
-
-    client = speech_v1p1beta1.SpeechClient()
-
-    # local_file_path = 'resources/commercial_mono.wav'
-
-    # The use case of the audio, e.g. PHONE_CALL, DISCUSSION, PRESENTATION, et al.
-    interaction_type = enums.RecognitionMetadata.InteractionType.VOICE_SEARCH
-
-    # The kind of device used to capture the audio
-    recording_device_type = enums.RecognitionMetadata.RecordingDeviceType.SMARTPHONE
-
-    # The device used to make the recording.
-    # Arbitrary string, e.g. 'Pixel XL', 'VoIP', 'Cardioid Microphone', or other
-    # value.
-    recording_device_name = "Pixel 3"
-    metadata = {
-        "interaction_type": interaction_type,
-        "recording_device_type": recording_device_type,
-        "recording_device_name": recording_device_name,
-    }
-
-    # The language of the supplied audio. Even though additional languages are
-    # provided by alternative_language_codes, a primary language is still required.
-    language_code = "en-US"
-    config = {"metadata": metadata, "language_code": language_code}
-    with io.open(local_file_path, "rb") as f:
-        content = f.read()
-    audio = {"content": content}
-
-    response = client.recognize(config, audio)
-    for result in response.results:
-        # First alternative is the most probable result
-        alternative = result.alternatives[0]
-        print(u"Transcript: {}".format(alternative.transcript))
-
-
-# [END speech_transcribe_recognition_metadata_beta]
-
-
-def main():
-    import argparse
-
-    parser = argparse.ArgumentParser()
-    parser.add_argument(
-        "--local_file_path", type=str, default="resources/commercial_mono.wav"
-    )
-    args = parser.parse_args()
-
-    sample_recognize(args.local_file_path)
-
-
-if __name__ == "__main__":
-    main()
diff --git a/samples/v1p1beta1/speech_transcribe_word_level_confidence_beta.py b/samples/v1p1beta1/speech_transcribe_word_level_confidence_beta.py
deleted file mode 100644
index 321f964a..00000000
--- a/samples/v1p1beta1/speech_transcribe_word_level_confidence_beta.py
+++ /dev/null
@@ -1,88 +0,0 @@
-# -*- coding: utf-8 -*-
-#
-# Copyright 2020 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-# DO NOT EDIT! This is a generated sample ("Request",  "speech_transcribe_word_level_confidence_beta")
-
-# To install the latest published package dependency, execute the following:
-#   pip install google-cloud-speech
-
-# sample-metadata
-#   title: Enabling word-level confidence (Local File) (Beta)
-#   description: Print confidence level for individual words in a transcription of a short audio
-#     file.
-#   usage: python3 samples/v1p1beta1/speech_transcribe_word_level_confidence_beta.py [--local_file_path "resources/brooklyn_bridge.flac"]
-
-# [START speech_transcribe_word_level_confidence_beta]
-from google.cloud import speech_v1p1beta1
-import io
-
-
-def sample_recognize(local_file_path):
-    """
-    Print confidence level for individual words in a transcription of a short audio
-    file.
-
-    Args:
-      local_file_path Path to local audio file, e.g. /path/audio.wav
-    """
-
-    client = speech_v1p1beta1.SpeechClient()
-
-    # local_file_path = 'resources/brooklyn_bridge.flac'
-
-    # When enabled, the first result returned by the API will include a list
-    # of words and the confidence level for each of those words.
-    enable_word_confidence = True
-
-    # The language of the supplied audio
-    language_code = "en-US"
-    config = {
-        "enable_word_confidence": enable_word_confidence,
-        "language_code": language_code,
-    }
-    with io.open(local_file_path, "rb") as f:
-        content = f.read()
-    audio = {"content": content}
-
-    response = client.recognize(config, audio)
-    # The first result includes confidence levels per word
-    result = response.results[0]
-    # First alternative is the most probable result
-    alternative = result.alternatives[0]
-    print(u"Transcript: {}".format(alternative.transcript))
-    # Print the confidence level of each word
-    for word in alternative.words:
-        print(u"Word: {}".format(word.word))
-        print(u"Confidence: {}".format(word.confidence))
-
-
-# [END speech_transcribe_word_level_confidence_beta]
-
-
-def main():
-    import argparse
-
-    parser = argparse.ArgumentParser()
-    parser.add_argument(
-        "--local_file_path", type=str, default="resources/brooklyn_bridge.flac"
-    )
-    args = parser.parse_args()
-
-    sample_recognize(args.local_file_path)
-
-
-if __name__ == "__main__":
-    main()
diff --git a/samples/v1p1beta1/test/samples.manifest.yaml b/samples/v1p1beta1/test/samples.manifest.yaml
deleted file mode 100644
index 787e335e..00000000
--- a/samples/v1p1beta1/test/samples.manifest.yaml
+++ /dev/null
@@ -1,32 +0,0 @@
-type: manifest/samples
-schema_version: 3
-base: &common
-  env: 'python'
-  bin: 'python3'
-  chdir: '{@manifest_dir}/../..'
-  basepath: '.'
-samples:
-- <<: *common
-  path: '{basepath}/v1p1beta1/speech_adaptation_beta.py'
-  sample: 'speech_adaptation_beta'
-- <<: *common
-  path: '{basepath}/v1p1beta1/speech_contexts_classes_beta.py'
-  sample: 'speech_contexts_classes_beta'
-- <<: *common
-  path: '{basepath}/v1p1beta1/speech_quickstart_beta.py'
-  sample: 'speech_quickstart_beta'
-- <<: *common
-  path: '{basepath}/v1p1beta1/speech_transcribe_auto_punctuation_beta.py'
-  sample: 'speech_transcribe_auto_punctuation_beta'
-- <<: *common
-  path: '{basepath}/v1p1beta1/speech_transcribe_diarization_beta.py'
-  sample: 'speech_transcribe_diarization_beta'
-- <<: *common
-  path: '{basepath}/v1p1beta1/speech_transcribe_multilanguage_beta.py'
-  sample: 'speech_transcribe_multilanguage_beta'
-- <<: *common
-  path: '{basepath}/v1p1beta1/speech_transcribe_recognition_metadata_beta.py'
-  sample: 'speech_transcribe_recognition_metadata_beta'
-- <<: *common
-  path: '{basepath}/v1p1beta1/speech_transcribe_word_level_confidence_beta.py'
-  sample: 'speech_transcribe_word_level_confidence_beta'
diff --git a/samples/v1p1beta1/test/speech_adaptation_beta.test.yaml b/samples/v1p1beta1/test/speech_adaptation_beta.test.yaml
deleted file mode 100644
index 4efe8e83..00000000
--- a/samples/v1p1beta1/test/speech_adaptation_beta.test.yaml
+++ /dev/null
@@ -1,11 +0,0 @@
-type: test/samples
-schema_version: 1
-test:
-  suites:
-  - name: Speech-to-Text Sample Tests For Speech Adaptation
-    cases:
-    - name: speech_adaptation_beta
-      spec:
-      - call: {sample: speech_adaptation_beta}
-      - assert_contains:
-        - literal: "how old is the Brooklyn Bridge"
diff --git a/samples/v1p1beta1/test/speech_contexts_classes_beta.test.yaml b/samples/v1p1beta1/test/speech_contexts_classes_beta.test.yaml
deleted file mode 100644
index b6dccfc7..00000000
--- a/samples/v1p1beta1/test/speech_contexts_classes_beta.test.yaml
+++ /dev/null
@@ -1,11 +0,0 @@
-type: test/samples
-schema_version: 1
-test:
-  suites:
-  - name: Speech-to-Text Sample Tests For Speech Contexts Static Classes
-    cases:
-    - name: speech_contexts_classes_beta
-      spec:
-      - call: {sample: speech_contexts_classes_beta}
-      - assert_contains:
-        - literal: "the time is 5:45 p.m."
diff --git a/samples/v1p1beta1/test/speech_quickstart_beta.test.yaml b/samples/v1p1beta1/test/speech_quickstart_beta.test.yaml
deleted file mode 100644
index bd5bf670..00000000
--- a/samples/v1p1beta1/test/speech_quickstart_beta.test.yaml
+++ /dev/null
@@ -1,11 +0,0 @@
-type: test/samples
-schema_version: 1
-test:
-  suites:
-  - name: Speech-to-Text Sample Tests For Quickstart
-    cases:
-    - name: speech_quickstart_beta
-      spec:
-      - call: {sample: speech_quickstart_beta}
-      - assert_contains:
-        - literal: "how old is the Brooklyn Bridge"
diff --git a/samples/v1p1beta1/test/speech_transcribe_auto_punctuation_beta.test.yaml b/samples/v1p1beta1/test/speech_transcribe_auto_punctuation_beta.test.yaml
deleted file mode 100644
index 1ab5f79a..00000000
--- a/samples/v1p1beta1/test/speech_transcribe_auto_punctuation_beta.test.yaml
+++ /dev/null
@@ -1,28 +0,0 @@
-type: test/samples
-schema_version: 1
-test:
-  suites:
-  - name: Getting punctuation in results (Local File) (Beta)
-    cases:
-
-    # This sample should default to using commercial_mono.wav
-    - name: speech_transcribe_auto_punctuation_beta (no arguments)
-      spec:
-      - call:
-          sample: speech_transcribe_auto_punctuation_beta
-      - assert_contains:
-        # Simply assert that actual punctuation is present from commercial_mono.wav
-        - literal: "?"
-        - literal: ","
-        - literal: ""
-
-    # Confirm that another file can be transcribed (use another file)
-    - name: speech_transcribe_auto_punctuation_beta (--local_file_path)
-      spec:
-      - call:
-          sample: speech_transcribe_auto_punctuation_beta
-          params:
-            local_file_path:
-              literal: "resources/brooklyn_bridge.flac"
-      - assert_contains:
-        - literal: "How old is the Brooklyn Bridge?"
diff --git a/samples/v1p1beta1/test/speech_transcribe_diarization_beta.test.yaml b/samples/v1p1beta1/test/speech_transcribe_diarization_beta.test.yaml
deleted file mode 100644
index 409e4b54..00000000
--- a/samples/v1p1beta1/test/speech_transcribe_diarization_beta.test.yaml
+++ /dev/null
@@ -1,40 +0,0 @@
-type: test/samples
-schema_version: 1
-test:
-  suites:
-  - name: Separating different speakers (Local File) (LRO) (Beta)
-    cases:
-
-    # This sample should default to using commercial_mono.wav
-    - name: speech_transcribe_diarization_beta (no arguments)
-      spec:
-      - call:
-          sample: speech_transcribe_diarization_beta
-      - assert_contains:
-        - literal: "Word: Chrome"
-        # The identifier given to each speaker is non-deterministic.
-        # For two speakers, it can be 0 and 1, 0 and 2, or other variations.
-        #
-        # The example audio file has two speakers, but this test is
-        # not verifying that greater than one speaker is detected.
-        - literal: "Speaker tag:"
-
-    # Confirm that another file can be transcribed (use another .flac file)
-    - name: speech_transcribe_diarization_beta (--local_file_path)
-      spec:
-      - call:
-          sample: speech_transcribe_diarization_beta
-          params:
-            local_file_path:
-              literal: "resources/multi.flac"
-      - assert_contains:
-        - literal: "how are you doing"
-        - literal: "Word: doing"
-        # The identifier given to each speaker is non-deterministic.
-        # For two speakers, it can be 0 and 1, 0 and 2, or other variations.
-        #
-        # The example audio file has two speakers, but this test is
-        # not verifying that greater than one speaker is detected.
-        - literal: "Speaker tag:"
-      - assert_not_contains:
-        - literal: "Chrome"
diff --git a/samples/v1p1beta1/test/speech_transcribe_multilanguage_beta.test.yaml b/samples/v1p1beta1/test/speech_transcribe_multilanguage_beta.test.yaml
deleted file mode 100644
index d9f2d710..00000000
--- a/samples/v1p1beta1/test/speech_transcribe_multilanguage_beta.test.yaml
+++ /dev/null
@@ -1,33 +0,0 @@
-type: test/samples
-schema_version: 1
-test:
-  suites:
-  - name: Detecting language spoken automatically (Local File) (Beta)
-    cases:
-
-    # This sample should default to using brooklyn_bridge.flac
-    - name: speech_transcribe_multilanguage_beta (no arguments)
-      spec:
-      - call:
-          sample: speech_transcribe_multilanguage_beta
-      - assert_contains:
-        - literal: "how old is the Brooklyn Bridge"
-        # Note: the primary language_code given was not English, but
-        #       English was provided in the list of alternative_language_codes
-        - literal: "Detected language: en-us"
-
-    # Confirm that another file can be transcribed (use another .flac file)
-    - name: speech_transcribe_multilanguage_beta (--local_file_path)
-      spec:
-      - call:
-          sample: speech_transcribe_multilanguage_beta
-          params:
-            local_file_path:
-              literal: "resources/multi.flac"
-      - assert_contains:
-        - literal: "how are you doing"
-        # Note: the primary language_code given was not English, but
-        #       English was provided in the list of alternative_language_codes
-        - literal: "Detected language: en-us"
-      - assert_not_contains:
-        - literal: "how old is the Brooklyn Bridge"
diff --git a/samples/v1p1beta1/test/speech_transcribe_recognition_metadata_beta.test.yaml b/samples/v1p1beta1/test/speech_transcribe_recognition_metadata_beta.test.yaml
deleted file mode 100644
index 57cf24a1..00000000
--- a/samples/v1p1beta1/test/speech_transcribe_recognition_metadata_beta.test.yaml
+++ /dev/null
@@ -1,27 +0,0 @@
-type: test/samples
-schema_version: 1
-test:
-  suites:
-  - name: Adding recognition metadata (Local File) (Beta)
-    cases:
-
-    # This sample should default to using commercial_mono.wav
-    - name: speech_transcribe_recognition_metadata_beta (no arguments)
-      spec:
-      - call:
-          sample: speech_transcribe_recognition_metadata_beta
-      - assert_contains:
-        - literal: "Chrome"
-
-    # Confirm that another file can be transcribed (use another file)
-    - name: speech_transcribe_recognition_metadata_beta (--local_file_path)
-      spec:
-      - call:
-          sample: speech_transcribe_recognition_metadata_beta
-          params:
-            local_file_path:
-              literal: "resources/brooklyn_bridge.flac"
-      - assert_contains:
-        - literal: "how old is the Brooklyn Bridge"
-      - assert_not_contains:
-        - literal: "Chrome"
diff --git a/samples/v1p1beta1/test/speech_transcribe_word_level_confidence_beta.test.yaml b/samples/v1p1beta1/test/speech_transcribe_word_level_confidence_beta.test.yaml
deleted file mode 100644
index 0d7d6057..00000000
--- a/samples/v1p1beta1/test/speech_transcribe_word_level_confidence_beta.test.yaml
+++ /dev/null
@@ -1,35 +0,0 @@
-type: test/samples
-schema_version: 1
-test:
-  suites:
-  - name: Enabling word-level confidence (Local File) (Beta)
-    cases:
-
-    # This sample should default to using brooklyn_bridge.flac
-    - name: speech_transcribe_word_level_confidence_beta (no arguments)
-      spec:
-      - call:
-          sample: speech_transcribe_word_level_confidence_beta
-      - assert_contains:
-        - literal: "how old is the Brooklyn Bridge"
-        - literal: "Word: how"
-        - literal: "Word: old"
-        - literal: "Word: is"
-        - literal: "Confidence: 0."
-
-    # Confirm that another file can be transcribed (use another .flac file)
-    - name: speech_transcribe_word_level_confidence_beta (--local_file_path)
-      spec:
-      - call:
-          sample: speech_transcribe_word_level_confidence_beta
-          params:
-            local_file_path:
-              literal: "resources/multi.flac"
-      - assert_contains:
-        - literal: "how are you doing"
-        - literal: "Word: how"
-        - literal: "Word: are"
-        - literal: "Word: you"
-        - literal: "Confidence: 0."
-      - assert_not_contains:
-        - literal: "how old is the Brooklyn Bridge"
diff --git a/setup.cfg b/setup.cfg
deleted file mode 100644
index 3bd55550..00000000
--- a/setup.cfg
+++ /dev/null
@@ -1,3 +0,0 @@
-# Generated by synthtool. DO NOT EDIT!
-[bdist_wheel]
-universal = 1
diff --git a/setup.py b/setup.py
deleted file mode 100644
index 49d6395d..00000000
--- a/setup.py
+++ /dev/null
@@ -1,86 +0,0 @@
-# Copyright 2018 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     http://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-import io
-import os
-
-import setuptools
-
-
-# Package metadata.
-
-name = "google-cloud-speech"
-description = "Google Cloud Speech API client library"
-version = "1.3.2"
-# Should be one of:
-# 'Development Status :: 3 - Alpha'
-# 'Development Status :: 4 - Beta'
-# 'Development Status :: 5 - Production/Stable'
-release_status = "Development Status :: 5 - Production/Stable"
-dependencies = ["google-api-core[grpc] >= 1.14.0, < 2.0.0dev"]
-extras = {}
-
-
-# Setup boilerplate below this line.
-
-package_root = os.path.abspath(os.path.dirname(__file__))
-
-readme_filename = os.path.join(package_root, "README.rst")
-with io.open(readme_filename, encoding="utf-8") as readme_file:
-    readme = readme_file.read()
-
-# Only include packages under the 'google' namespace. Do not include tests,
-# benchmarks, etc.
-packages = [
-    package for package in setuptools.find_packages() if package.startswith("google")
-]
-
-# Determine which namespaces are needed.
-namespaces = ["google"]
-if "google.cloud" in packages:
-    namespaces.append("google.cloud")
-
-
-setuptools.setup(
-    name=name,
-    version=version,
-    description=description,
-    long_description=readme,
-    author="Google LLC",
-    author_email="googleapis-packages@google.com",
-    license="Apache 2.0",
-    url="https://github.com/googleapis/python-speech",
-    classifiers=[
-        release_status,
-        "Intended Audience :: Developers",
-        "License :: OSI Approved :: Apache Software License",
-        "Programming Language :: Python",
-        "Programming Language :: Python :: 2",
-        "Programming Language :: Python :: 2.7",
-        "Programming Language :: Python :: 3",
-        "Programming Language :: Python :: 3.5",
-        "Programming Language :: Python :: 3.6",
-        "Programming Language :: Python :: 3.7",
-        "Operating System :: OS Independent",
-        "Topic :: Internet",
-    ],
-    platforms="Posix; MacOS X; Windows",
-    packages=packages,
-    namespace_packages=namespaces,
-    install_requires=dependencies,
-    extras_require=extras,
-    python_requires=">=2.7,!=3.0.*,!=3.1.*,!=3.2.*,!=3.3.*",
-    include_package_data=True,
-    zip_safe=False,
-)
diff --git a/synth.metadata b/synth.metadata
deleted file mode 100644
index 183eecb3..00000000
--- a/synth.metadata
+++ /dev/null
@@ -1,49 +0,0 @@
-{
-  "updateTime": "2020-02-03T22:57:22.365987Z",
-  "sources": [
-    {
-      "generator": {
-        "name": "artman",
-        "version": "0.44.4",
-        "dockerImage": "googleapis/artman@sha256:19e945954fc960a4bdfee6cb34695898ab21a8cf0bac063ee39b91f00a1faec8"
-      }
-    },
-    {
-      "git": {
-        "name": "googleapis",
-        "remote": "https://github.com/googleapis/googleapis.git",
-        "sha": "29d40b78e3dc1579b0b209463fbcb76e5767f72a",
-        "internalRef": "292979741"
-      }
-    },
-    {
-      "template": {
-        "name": "python_split_library",
-        "origin": "synthtool.gcp",
-        "version": "2019.10.17"
-      }
-    }
-  ],
-  "destinations": [
-    {
-      "client": {
-        "source": "googleapis",
-        "apiName": "speech",
-        "apiVersion": "v1p1beta1",
-        "language": "python",
-        "generator": "gapic",
-        "config": "google/cloud/speech/artman_speech_v1p1beta1.yaml"
-      }
-    },
-    {
-      "client": {
-        "source": "googleapis",
-        "apiName": "speech",
-        "apiVersion": "v1",
-        "language": "python",
-        "generator": "gapic",
-        "config": "google/cloud/speech/artman_speech_v1.yaml"
-      }
-    }
-  ]
-}
\ No newline at end of file
diff --git a/synth.py b/synth.py
deleted file mode 100644
index 51a14104..00000000
--- a/synth.py
+++ /dev/null
@@ -1,64 +0,0 @@
-# Copyright 2019 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     http://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-"""This script is used to synthesize generated parts of this library."""
-
-import synthtool as s
-from synthtool import gcp
-
-gapic = gcp.GAPICBazel()
-common = gcp.CommonTemplates()
-versions = ["v1p1beta1", "v1"]
-
-
-# ----------------------------------------------------------------------------
-# Generate speech GAPIC layer
-# ----------------------------------------------------------------------------
-for version in versions:
-    library = gapic.py_library(
-        service="speech",
-        version=version,
-        bazel_target=f"//google/cloud/speech/{version}:speech-{version}-py",
-        include_protos=True,
-    )
-
-    # Don't move over __init__.py, as we modify it to make the generated client
-    # use helpers.py.
-    s.move(library / f"google/cloud/speech_{version}/types.py")
-    s.move(library / f"google/cloud/speech_{version}/gapic")
-    s.move(library / f"google/cloud/speech_{version}/proto")
-    s.move(library / f"tests/unit/gapic/{version}")
-    s.move(library / f"docs/gapic/{version}")
-    s.move(library / f"samples")
-
-
-# Use the highest version library to generate documentation import alias.
-s.move(library / "google/cloud/speech.py")
-
-
-# Fix tests to use the direct gapic client instead of the wrapped helper
-# client.
-s.replace(
-    "tests/unit/**/test*client*.py",
-    r"from google\.cloud import speech_(.+?)$",
-    r"from google.cloud.speech_\1.gapic import speech_client as speech_\1",
-)
-
-# ----------------------------------------------------------------------------
-# Add templated files
-# ----------------------------------------------------------------------------
-templated_files = common.py_library(cov_level=87, include_samples=True)
-s.move(templated_files)
-
-s.shell.run(["nox", "-s", "blacken"], hide_output=False)
diff --git a/tests/system/gapic/v1/test_system_speech_v1.py b/tests/system/gapic/v1/test_system_speech_v1.py
deleted file mode 100644
index 1c30e3fa..00000000
--- a/tests/system/gapic/v1/test_system_speech_v1.py
+++ /dev/null
@@ -1,94 +0,0 @@
-# Copyright 2018 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-import os
-import io
-import requests
-
-from google.cloud import speech_v1
-
-
-class TestSystemSpeech(object):
-    def test_recognize(self):
-
-        try:
-            BUCKET = os.environ["GOOGLE_CLOUD_TESTS_SPEECH_BUCKET"]
-        except KeyError:
-            BUCKET = "cloud-samples-tests"
-
-        client = speech_v1.SpeechClient()
-
-        config = {
-            "encoding": speech_v1.enums.RecognitionConfig.AudioEncoding.FLAC,
-            "language_code": "en-US",
-            "sample_rate_hertz": 16000,
-        }
-
-        uri = "gs://{}/speech/brooklyn.flac".format(BUCKET)
-        audio = {"uri": uri}
-
-        response = client.recognize(config, audio)
-
-        assert response.results[0].alternatives[0].transcript is not None
-
-    def test_long_running_recognize(self):
-
-        try:
-            BUCKET = os.environ["GOOGLE_CLOUD_TESTS_SPEECH_BUCKET"]
-        except KeyError:
-            BUCKET = "cloud-samples-tests"
-
-        client = speech_v1.SpeechClient()
-
-        config = speech_v1.types.RecognitionConfig(
-            encoding=speech_v1.enums.RecognitionConfig.AudioEncoding.FLAC,
-            language_code="en-US",
-            sample_rate_hertz=16000,
-        )
-
-        uri = "gs://{}/speech/brooklyn.flac".format(BUCKET)
-        audio = {"uri": uri}
-
-        response = client.long_running_recognize(config, audio)
-
-        assert response.result() is not None
-
-    def test_streaming_recognize(self):
-
-        try:
-            BUCKET = os.environ["GOOGLE_CLOUD_TESTS_SPEECH_BUCKET"]
-        except KeyError:
-            BUCKET = "cloud-samples-tests"
-
-        client = speech_v1.SpeechClient()
-
-        config = speech_v1.types.RecognitionConfig(
-            encoding=speech_v1.enums.RecognitionConfig.AudioEncoding.FLAC,
-            language_code="en-US",
-            sample_rate_hertz=16000,
-        )
-        streamingConfig = speech_v1.types.StreamingRecognitionConfig(config=config)
-
-        uri = "https://storage.googleapis.com/{}/speech/brooklyn.flac".format(BUCKET)
-        streaming_requests = [
-            speech_v1.types.StreamingRecognizeRequest(
-                audio_content=requests.get(uri).content
-            )
-        ]
-
-        responses = client.streaming_recognize(streamingConfig, streaming_requests)
-
-        for response in responses:
-            for result in response.results:
-                assert result.alternatives[0].transcript is not None
diff --git a/tests/system/gapic/v1p1beta1/test_system_speech_v1p1beta1.py b/tests/system/gapic/v1p1beta1/test_system_speech_v1p1beta1.py
deleted file mode 100644
index 36514089..00000000
--- a/tests/system/gapic/v1p1beta1/test_system_speech_v1p1beta1.py
+++ /dev/null
@@ -1,96 +0,0 @@
-# Copyright 2018 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-import os
-import io
-import requests
-
-from google.cloud import speech_v1p1beta1
-
-
-class TestSystemSpeech(object):
-    def test_recognize(self):
-
-        try:
-            BUCKET = os.environ["GOOGLE_CLOUD_TESTS_SPEECH_BUCKET"]
-        except KeyError:
-            BUCKET = "cloud-samples-tests"
-
-        client = speech_v1p1beta1.SpeechClient()
-
-        config = {
-            "encoding": speech_v1p1beta1.enums.RecognitionConfig.AudioEncoding.FLAC,
-            "language_code": "en-US",
-            "sample_rate_hertz": 16000,
-        }
-
-        uri = "gs://{}/speech/brooklyn.flac".format(BUCKET)
-        audio = {"uri": uri}
-
-        response = client.recognize(config, audio)
-
-        assert response.results[0].alternatives[0].transcript is not None
-
-    def test_long_running_recognize(self):
-
-        try:
-            BUCKET = os.environ["GOOGLE_CLOUD_TESTS_SPEECH_BUCKET"]
-        except KeyError:
-            BUCKET = "cloud-samples-tests"
-
-        client = speech_v1p1beta1.SpeechClient()
-
-        config = speech_v1p1beta1.types.RecognitionConfig(
-            encoding=speech_v1p1beta1.enums.RecognitionConfig.AudioEncoding.FLAC,
-            language_code="en-US",
-            sample_rate_hertz=16000,
-        )
-
-        uri = "gs://{}/speech/brooklyn.flac".format(BUCKET)
-        audio = {"uri": uri}
-
-        response = client.long_running_recognize(config, audio)
-
-        assert response.result() is not None
-
-    def test_streaming_recognize(self):
-
-        try:
-            BUCKET = os.environ["GOOGLE_CLOUD_TESTS_SPEECH_BUCKET"]
-        except KeyError:
-            BUCKET = "cloud-samples-tests"
-
-        client = speech_v1p1beta1.SpeechClient()
-
-        config = speech_v1p1beta1.types.RecognitionConfig(
-            encoding=speech_v1p1beta1.enums.RecognitionConfig.AudioEncoding.FLAC,
-            language_code="en-US",
-            sample_rate_hertz=16000,
-        )
-        streamingConfig = speech_v1p1beta1.types.StreamingRecognitionConfig(
-            config=config
-        )
-
-        uri = "https://storage.googleapis.com/{}/speech/brooklyn.flac".format(BUCKET)
-        streaming_requests = [
-            speech_v1p1beta1.types.StreamingRecognizeRequest(
-                audio_content=requests.get(uri).content
-            )
-        ]
-
-        responses = client.streaming_recognize(streamingConfig, streaming_requests)
-
-        for response in responses:
-            for result in response.results:
-                assert result.alternatives[0].transcript is not None
diff --git a/tests/unit/gapic/v1/test_speech_client_v1.py b/tests/unit/gapic/v1/test_speech_client_v1.py
deleted file mode 100644
index 29e0e029..00000000
--- a/tests/unit/gapic/v1/test_speech_client_v1.py
+++ /dev/null
@@ -1,244 +0,0 @@
-# -*- coding: utf-8 -*-
-#
-# Copyright 2020 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-"""Unit tests."""
-
-import mock
-import pytest
-
-from google.rpc import status_pb2
-
-from google.cloud.speech_v1.gapic import speech_client as speech_v1
-from google.cloud.speech_v1 import enums
-from google.cloud.speech_v1.proto import cloud_speech_pb2
-from google.longrunning import operations_pb2
-
-
-class MultiCallableStub(object):
-    """Stub for the grpc.UnaryUnaryMultiCallable interface."""
-
-    def __init__(self, method, channel_stub):
-        self.method = method
-        self.channel_stub = channel_stub
-
-    def __call__(self, request, timeout=None, metadata=None, credentials=None):
-        self.channel_stub.requests.append((self.method, request))
-
-        response = None
-        if self.channel_stub.responses:
-            response = self.channel_stub.responses.pop()
-
-        if isinstance(response, Exception):
-            raise response
-
-        if response:
-            return response
-
-
-class ChannelStub(object):
-    """Stub for the grpc.Channel interface."""
-
-    def __init__(self, responses=[]):
-        self.responses = responses
-        self.requests = []
-
-    def unary_unary(self, method, request_serializer=None, response_deserializer=None):
-        return MultiCallableStub(method, self)
-
-    def stream_stream(
-        self, method, request_serializer=None, response_deserializer=None
-    ):
-        return MultiCallableStub(method, self)
-
-
-class CustomException(Exception):
-    pass
-
-
-class TestSpeechClient(object):
-    def test_recognize(self):
-        # Setup Expected Response
-        expected_response = {}
-        expected_response = cloud_speech_pb2.RecognizeResponse(**expected_response)
-
-        # Mock the API response
-        channel = ChannelStub(responses=[expected_response])
-        patch = mock.patch("google.api_core.grpc_helpers.create_channel")
-        with patch as create_channel:
-            create_channel.return_value = channel
-            client = speech_v1.SpeechClient()
-
-        # Setup Request
-        encoding = enums.RecognitionConfig.AudioEncoding.FLAC
-        sample_rate_hertz = 44100
-        language_code = "en-US"
-        config = {
-            "encoding": encoding,
-            "sample_rate_hertz": sample_rate_hertz,
-            "language_code": language_code,
-        }
-        uri = "gs://bucket_name/file_name.flac"
-        audio = {"uri": uri}
-
-        response = client.recognize(config, audio)
-        assert expected_response == response
-
-        assert len(channel.requests) == 1
-        expected_request = cloud_speech_pb2.RecognizeRequest(config=config, audio=audio)
-        actual_request = channel.requests[0][1]
-        assert expected_request == actual_request
-
-    def test_recognize_exception(self):
-        # Mock the API response
-        channel = ChannelStub(responses=[CustomException()])
-        patch = mock.patch("google.api_core.grpc_helpers.create_channel")
-        with patch as create_channel:
-            create_channel.return_value = channel
-            client = speech_v1.SpeechClient()
-
-        # Setup request
-        encoding = enums.RecognitionConfig.AudioEncoding.FLAC
-        sample_rate_hertz = 44100
-        language_code = "en-US"
-        config = {
-            "encoding": encoding,
-            "sample_rate_hertz": sample_rate_hertz,
-            "language_code": language_code,
-        }
-        uri = "gs://bucket_name/file_name.flac"
-        audio = {"uri": uri}
-
-        with pytest.raises(CustomException):
-            client.recognize(config, audio)
-
-    def test_long_running_recognize(self):
-        # Setup Expected Response
-        expected_response = {}
-        expected_response = cloud_speech_pb2.LongRunningRecognizeResponse(
-            **expected_response
-        )
-        operation = operations_pb2.Operation(
-            name="operations/test_long_running_recognize", done=True
-        )
-        operation.response.Pack(expected_response)
-
-        # Mock the API response
-        channel = ChannelStub(responses=[operation])
-        patch = mock.patch("google.api_core.grpc_helpers.create_channel")
-        with patch as create_channel:
-            create_channel.return_value = channel
-            client = speech_v1.SpeechClient()
-
-        # Setup Request
-        encoding = enums.RecognitionConfig.AudioEncoding.FLAC
-        sample_rate_hertz = 44100
-        language_code = "en-US"
-        config = {
-            "encoding": encoding,
-            "sample_rate_hertz": sample_rate_hertz,
-            "language_code": language_code,
-        }
-        uri = "gs://bucket_name/file_name.flac"
-        audio = {"uri": uri}
-
-        response = client.long_running_recognize(config, audio)
-        result = response.result()
-        assert expected_response == result
-
-        assert len(channel.requests) == 1
-        expected_request = cloud_speech_pb2.LongRunningRecognizeRequest(
-            config=config, audio=audio
-        )
-        actual_request = channel.requests[0][1]
-        assert expected_request == actual_request
-
-    def test_long_running_recognize_exception(self):
-        # Setup Response
-        error = status_pb2.Status()
-        operation = operations_pb2.Operation(
-            name="operations/test_long_running_recognize_exception", done=True
-        )
-        operation.error.CopyFrom(error)
-
-        # Mock the API response
-        channel = ChannelStub(responses=[operation])
-        patch = mock.patch("google.api_core.grpc_helpers.create_channel")
-        with patch as create_channel:
-            create_channel.return_value = channel
-            client = speech_v1.SpeechClient()
-
-        # Setup Request
-        encoding = enums.RecognitionConfig.AudioEncoding.FLAC
-        sample_rate_hertz = 44100
-        language_code = "en-US"
-        config = {
-            "encoding": encoding,
-            "sample_rate_hertz": sample_rate_hertz,
-            "language_code": language_code,
-        }
-        uri = "gs://bucket_name/file_name.flac"
-        audio = {"uri": uri}
-
-        response = client.long_running_recognize(config, audio)
-        exception = response.exception()
-        assert exception.errors[0] == error
-
-    def test_streaming_recognize(self):
-        # Setup Expected Response
-        expected_response = {}
-        expected_response = cloud_speech_pb2.StreamingRecognizeResponse(
-            **expected_response
-        )
-
-        # Mock the API response
-        channel = ChannelStub(responses=[iter([expected_response])])
-        patch = mock.patch("google.api_core.grpc_helpers.create_channel")
-        with patch as create_channel:
-            create_channel.return_value = channel
-            client = speech_v1.SpeechClient()
-
-        # Setup Request
-        request = {}
-        request = cloud_speech_pb2.StreamingRecognizeRequest(**request)
-        requests = [request]
-
-        response = client.streaming_recognize(requests)
-        resources = list(response)
-        assert len(resources) == 1
-        assert expected_response == resources[0]
-
-        assert len(channel.requests) == 1
-        actual_requests = channel.requests[0][1]
-        assert len(actual_requests) == 1
-        actual_request = list(actual_requests)[0]
-        assert request == actual_request
-
-    def test_streaming_recognize_exception(self):
-        # Mock the API response
-        channel = ChannelStub(responses=[CustomException()])
-        patch = mock.patch("google.api_core.grpc_helpers.create_channel")
-        with patch as create_channel:
-            create_channel.return_value = channel
-            client = speech_v1.SpeechClient()
-
-        # Setup request
-        request = {}
-
-        request = cloud_speech_pb2.StreamingRecognizeRequest(**request)
-        requests = [request]
-
-        with pytest.raises(CustomException):
-            client.streaming_recognize(requests)
diff --git a/tests/unit/gapic/v1p1beta1/test_speech_client_v1p1beta1.py b/tests/unit/gapic/v1p1beta1/test_speech_client_v1p1beta1.py
deleted file mode 100644
index c4588b52..00000000
--- a/tests/unit/gapic/v1p1beta1/test_speech_client_v1p1beta1.py
+++ /dev/null
@@ -1,244 +0,0 @@
-# -*- coding: utf-8 -*-
-#
-# Copyright 2020 Google LLC
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     https://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-"""Unit tests."""
-
-import mock
-import pytest
-
-from google.rpc import status_pb2
-
-from google.cloud.speech_v1p1beta1.gapic import speech_client as speech_v1p1beta1
-from google.cloud.speech_v1p1beta1 import enums
-from google.cloud.speech_v1p1beta1.proto import cloud_speech_pb2
-from google.longrunning import operations_pb2
-
-
-class MultiCallableStub(object):
-    """Stub for the grpc.UnaryUnaryMultiCallable interface."""
-
-    def __init__(self, method, channel_stub):
-        self.method = method
-        self.channel_stub = channel_stub
-
-    def __call__(self, request, timeout=None, metadata=None, credentials=None):
-        self.channel_stub.requests.append((self.method, request))
-
-        response = None
-        if self.channel_stub.responses:
-            response = self.channel_stub.responses.pop()
-
-        if isinstance(response, Exception):
-            raise response
-
-        if response:
-            return response
-
-
-class ChannelStub(object):
-    """Stub for the grpc.Channel interface."""
-
-    def __init__(self, responses=[]):
-        self.responses = responses
-        self.requests = []
-
-    def unary_unary(self, method, request_serializer=None, response_deserializer=None):
-        return MultiCallableStub(method, self)
-
-    def stream_stream(
-        self, method, request_serializer=None, response_deserializer=None
-    ):
-        return MultiCallableStub(method, self)
-
-
-class CustomException(Exception):
-    pass
-
-
-class TestSpeechClient(object):
-    def test_recognize(self):
-        # Setup Expected Response
-        expected_response = {}
-        expected_response = cloud_speech_pb2.RecognizeResponse(**expected_response)
-
-        # Mock the API response
-        channel = ChannelStub(responses=[expected_response])
-        patch = mock.patch("google.api_core.grpc_helpers.create_channel")
-        with patch as create_channel:
-            create_channel.return_value = channel
-            client = speech_v1p1beta1.SpeechClient()
-
-        # Setup Request
-        encoding = enums.RecognitionConfig.AudioEncoding.FLAC
-        sample_rate_hertz = 44100
-        language_code = "en-US"
-        config = {
-            "encoding": encoding,
-            "sample_rate_hertz": sample_rate_hertz,
-            "language_code": language_code,
-        }
-        uri = "gs://bucket_name/file_name.flac"
-        audio = {"uri": uri}
-
-        response = client.recognize(config, audio)
-        assert expected_response == response
-
-        assert len(channel.requests) == 1
-        expected_request = cloud_speech_pb2.RecognizeRequest(config=config, audio=audio)
-        actual_request = channel.requests[0][1]
-        assert expected_request == actual_request
-
-    def test_recognize_exception(self):
-        # Mock the API response
-        channel = ChannelStub(responses=[CustomException()])
-        patch = mock.patch("google.api_core.grpc_helpers.create_channel")
-        with patch as create_channel:
-            create_channel.return_value = channel
-            client = speech_v1p1beta1.SpeechClient()
-
-        # Setup request
-        encoding = enums.RecognitionConfig.AudioEncoding.FLAC
-        sample_rate_hertz = 44100
-        language_code = "en-US"
-        config = {
-            "encoding": encoding,
-            "sample_rate_hertz": sample_rate_hertz,
-            "language_code": language_code,
-        }
-        uri = "gs://bucket_name/file_name.flac"
-        audio = {"uri": uri}
-
-        with pytest.raises(CustomException):
-            client.recognize(config, audio)
-
-    def test_long_running_recognize(self):
-        # Setup Expected Response
-        expected_response = {}
-        expected_response = cloud_speech_pb2.LongRunningRecognizeResponse(
-            **expected_response
-        )
-        operation = operations_pb2.Operation(
-            name="operations/test_long_running_recognize", done=True
-        )
-        operation.response.Pack(expected_response)
-
-        # Mock the API response
-        channel = ChannelStub(responses=[operation])
-        patch = mock.patch("google.api_core.grpc_helpers.create_channel")
-        with patch as create_channel:
-            create_channel.return_value = channel
-            client = speech_v1p1beta1.SpeechClient()
-
-        # Setup Request
-        encoding = enums.RecognitionConfig.AudioEncoding.FLAC
-        sample_rate_hertz = 44100
-        language_code = "en-US"
-        config = {
-            "encoding": encoding,
-            "sample_rate_hertz": sample_rate_hertz,
-            "language_code": language_code,
-        }
-        uri = "gs://bucket_name/file_name.flac"
-        audio = {"uri": uri}
-
-        response = client.long_running_recognize(config, audio)
-        result = response.result()
-        assert expected_response == result
-
-        assert len(channel.requests) == 1
-        expected_request = cloud_speech_pb2.LongRunningRecognizeRequest(
-            config=config, audio=audio
-        )
-        actual_request = channel.requests[0][1]
-        assert expected_request == actual_request
-
-    def test_long_running_recognize_exception(self):
-        # Setup Response
-        error = status_pb2.Status()
-        operation = operations_pb2.Operation(
-            name="operations/test_long_running_recognize_exception", done=True
-        )
-        operation.error.CopyFrom(error)
-
-        # Mock the API response
-        channel = ChannelStub(responses=[operation])
-        patch = mock.patch("google.api_core.grpc_helpers.create_channel")
-        with patch as create_channel:
-            create_channel.return_value = channel
-            client = speech_v1p1beta1.SpeechClient()
-
-        # Setup Request
-        encoding = enums.RecognitionConfig.AudioEncoding.FLAC
-        sample_rate_hertz = 44100
-        language_code = "en-US"
-        config = {
-            "encoding": encoding,
-            "sample_rate_hertz": sample_rate_hertz,
-            "language_code": language_code,
-        }
-        uri = "gs://bucket_name/file_name.flac"
-        audio = {"uri": uri}
-
-        response = client.long_running_recognize(config, audio)
-        exception = response.exception()
-        assert exception.errors[0] == error
-
-    def test_streaming_recognize(self):
-        # Setup Expected Response
-        expected_response = {}
-        expected_response = cloud_speech_pb2.StreamingRecognizeResponse(
-            **expected_response
-        )
-
-        # Mock the API response
-        channel = ChannelStub(responses=[iter([expected_response])])
-        patch = mock.patch("google.api_core.grpc_helpers.create_channel")
-        with patch as create_channel:
-            create_channel.return_value = channel
-            client = speech_v1p1beta1.SpeechClient()
-
-        # Setup Request
-        request = {}
-        request = cloud_speech_pb2.StreamingRecognizeRequest(**request)
-        requests = [request]
-
-        response = client.streaming_recognize(requests)
-        resources = list(response)
-        assert len(resources) == 1
-        assert expected_response == resources[0]
-
-        assert len(channel.requests) == 1
-        actual_requests = channel.requests[0][1]
-        assert len(actual_requests) == 1
-        actual_request = list(actual_requests)[0]
-        assert request == actual_request
-
-    def test_streaming_recognize_exception(self):
-        # Mock the API response
-        channel = ChannelStub(responses=[CustomException()])
-        patch = mock.patch("google.api_core.grpc_helpers.create_channel")
-        with patch as create_channel:
-            create_channel.return_value = channel
-            client = speech_v1p1beta1.SpeechClient()
-
-        # Setup request
-        request = {}
-
-        request = cloud_speech_pb2.StreamingRecognizeRequest(**request)
-        requests = [request]
-
-        with pytest.raises(CustomException):
-            client.streaming_recognize(requests)
diff --git a/tests/unit/test_helpers.py b/tests/unit/test_helpers.py
deleted file mode 100644
index 366b1821..00000000
--- a/tests/unit/test_helpers.py
+++ /dev/null
@@ -1,53 +0,0 @@
-# Copyright 2017, Google LLC All rights reserved.
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#     http://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-from __future__ import absolute_import
-from types import GeneratorType
-
-import mock
-
-import google.auth.credentials
-from google.cloud.speech_v1 import SpeechClient
-from google.cloud.speech_v1 import types
-
-
-def make_speech_client():
-    credentials = mock.Mock(spec=google.auth.credentials.Credentials)
-    return SpeechClient(credentials=credentials)
-
-
-def test_streaming_recognize():
-    client = make_speech_client()
-
-    config = types.StreamingRecognitionConfig()
-    requests = [types.StreamingRecognizeRequest(audio_content=b"...")]
-    super_patch = mock.patch(
-        "google.cloud.speech_v1.speech_client.SpeechClient." "streaming_recognize",
-        autospec=True,
-    )
-
-    with super_patch as streaming_recognize:
-        client.streaming_recognize(config, requests)
-
-    # Assert that we called streaming recognize with an iterable
-    # that evaluates to the correct format.
-    _, args, kwargs = streaming_recognize.mock_calls[0]
-    api_requests = args[1]
-    assert isinstance(api_requests, GeneratorType)
-    assert list(api_requests) == [
-        types.StreamingRecognizeRequest(streaming_config=config),
-        requests[0],
-    ]
-    assert "retry" in kwargs
-    assert "timeout" in kwargs