chore: regens API reference docs (#889)
diff --git a/docs/dyn/speech_v1.speech.html b/docs/dyn/speech_v1.speech.html
index 18a6564..e743764 100644
--- a/docs/dyn/speech_v1.speech.html
+++ b/docs/dyn/speech_v1.speech.html
@@ -75,14 +75,14 @@
<h1><a href="speech_v1.html">Cloud Speech-to-Text API</a> . <a href="speech_v1.speech.html">speech</a></h1>
<h2>Instance Methods</h2>
<p class="toc_element">
- <code><a href="#longrunningrecognize">longrunningrecognize(body, x__xgafv=None)</a></code></p>
+ <code><a href="#longrunningrecognize">longrunningrecognize(body=None, x__xgafv=None)</a></code></p>
<p class="firstline">Performs asynchronous speech recognition: receive results via the</p>
<p class="toc_element">
- <code><a href="#recognize">recognize(body, x__xgafv=None)</a></code></p>
+ <code><a href="#recognize">recognize(body=None, x__xgafv=None)</a></code></p>
<p class="firstline">Performs synchronous speech recognition: receive results after all audio</p>
<h3>Method Details</h3>
<div class="method">
- <code class="details" id="longrunningrecognize">longrunningrecognize(body, x__xgafv=None)</code>
+ <code class="details" id="longrunningrecognize">longrunningrecognize(body=None, x__xgafv=None)</code>
<pre>Performs asynchronous speech recognition: receive results via the
google.longrunning.Operations interface. Returns either an
`Operation.error` or an `Operation.response` which contains
@@ -91,15 +91,15 @@
[how-to](https://cloud.google.com/speech-to-text/docs/async-recognize).
Args:
- body: object, The request body. (required)
+ body: object, The request body.
The object takes the form of:
{ # The top-level message sent by the client for the `LongRunningRecognize`
# method.
- "audio": { # Contains audio data in the encoding specified in the `RecognitionConfig`. # *Required* The audio data to be recognized.
+ "audio": { # Contains audio data in the encoding specified in the `RecognitionConfig`. # Required. The audio data to be recognized.
# Either `content` or `uri` must be supplied. Supplying both or neither
# returns google.rpc.Code.INVALID_ARGUMENT. See
- # [content limits](/speech-to-text/quotas#content).
+ # [content limits](https://cloud.google.com/speech-to-text/quotas#content).
"content": "A String", # 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.
@@ -111,15 +111,16 @@
# google.rpc.Code.INVALID_ARGUMENT). For more information, see
# [Request URIs](https://cloud.google.com/storage/docs/reference-uris).
},
- "config": { # Provides information to the recognizer that specifies how to process the # *Required* Provides information to the recognizer that specifies how to
+ "config": { # Provides information to the recognizer that specifies how to process the # Required. Provides information to the recognizer that specifies how to
# process the request.
# request.
- "languageCode": "A String", # *Required* The language of the supplied audio as a
+ "languageCode": "A String", # 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](/speech-to-text/docs/languages)
- # for a list of the currently supported language codes.
- "audioChannelCount": 42, # *Optional* The number of channels in the input audio data.
+ # See [Language
+ # Support](https://cloud.google.com/speech-to-text/docs/languages) for a list
+ # of the currently supported language codes.
+ "audioChannelCount": 42, # 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'.
@@ -131,44 +132,34 @@
"encoding": "A String", # 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.
- "enableAutomaticPunctuation": True or False, # *Optional* If 'true', adds punctuation to recognition result hypotheses.
+ "enableAutomaticPunctuation": True or False, # 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.
- "enableSeparateRecognitionPerChannel": True or False, # This needs to be set to `true` explicitly and `audio_channel_count` > 1
+ "enableSeparateRecognitionPerChannel": True or False, # 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.
- "enableWordTimeOffsets": True or False, # *Optional* If `true`, the top result includes a list of words and
+ "enableWordTimeOffsets": True or False, # 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`.
- "maxAlternatives": 42, # *Optional* Maximum number of recognition hypotheses to be returned.
+ "maxAlternatives": 42, # 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.
- "useEnhanced": True or False, # *Optional* Set to true to use an enhanced model for speech recognition.
+ "useEnhanced": True or False, # 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:
- # 1. project is eligible for requesting enhanced models
- # 2. an enhanced model exists for the audio
+ # 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.
- #
- # Enhanced speech models require that you opt-in to data logging using
- # instructions in the
- # [documentation](/speech-to-text/docs/enable-data-logging). If you set
- # `use_enhanced` to true and you have not enabled audio logging, then you
- # will receive an error.
"sampleRateHertz": 42, # 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
@@ -176,53 +167,74 @@
# 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.
- "profanityFilter": True or False, # *Optional* If set to `true`, the server will attempt to filter out
+ "profanityFilter": True or False, # 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.
- "model": "A String", # *Optional* Which model to select for the given request. Select the model
+ "model": "A String", # 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
+ # <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.
+ # 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>
- "speechContexts": [ # *Optional* array of SpeechContext.
+ # recorded at a 16khz or greater sampling rate.</td>
+ # </tr>
+ # </table>
+ "diarizationConfig": { # Config to enable speaker diarization. # 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.
+ "minSpeakerCount": 42, # 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.
+ "enableSpeakerDiarization": True or False, # 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.
+ "maxSpeakerCount": 42, # 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.
+ "speakerTag": 42, # Output only. Unused.
+ },
+ "speechContexts": [ # Array of SpeechContext.
# A means to provide context to assist the speech recognition. For more
- # information, see [Phrase Hints](/speech-to-text/docs/basics#phrase-hints).
+ # information, see
+ # [speech
+ # adaptation](https://cloud.google.com/speech-to-text/docs/context-strength).
{ # Provides "hints" to the speech recognizer to favor specific words and phrases
# in the results.
- "phrases": [ # *Optional* A list of strings containing words and phrases "hints" so that
+ "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](/speech-to-text/quotas#content).
+ # [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
@@ -233,12 +245,10 @@
],
},
],
- "metadata": { # Description of audio data to be recognized. # *Optional* Metadata regarding this request.
+ "metadata": { # Description of audio data to be recognized. # Metadata regarding this request.
"recordingDeviceType": "A String", # The type of device the speech was recorded with.
"originalMediaType": "A String", # The original media the speech was recorded on.
"microphoneDistance": "A String", # The audio type that most closely describes the audio being recognized.
- "obfuscatedId": "A String", # Obfuscated (privacy-protected) ID of the user, to identify number of
- # unique users using the service.
"originalMimeType": "A String", # 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
@@ -267,6 +277,12 @@
{ # This resource represents a long-running operation that is the result of a
# network API call.
+ "metadata": { # Service-specific metadata associated with the operation. It typically
+ # contains progress information and common metadata such as create time.
+ # Some services might not provide such metadata. Any method that returns a
+ # long-running operation should document the metadata type, if any.
+ "a_key": "", # Properties of the object. Contains field @type with type URL.
+ },
"error": { # The `Status` type defines a logical error model that is suitable for # The error result of the operation in case of failure or cancellation.
# different programming environments, including REST APIs and RPC APIs. It is
# used by [gRPC](https://github.com/grpc). Each `Status` message contains
@@ -301,29 +317,23 @@
"name": "A String", # The server-assigned name, which is only unique within the same service that
# originally returns it. If you use the default HTTP mapping, the
# `name` should be a resource name ending with `operations/{unique_id}`.
- "metadata": { # Service-specific metadata associated with the operation. It typically
- # contains progress information and common metadata such as create time.
- # Some services might not provide such metadata. Any method that returns a
- # long-running operation should document the metadata type, if any.
- "a_key": "", # Properties of the object. Contains field @type with type URL.
- },
}</pre>
</div>
<div class="method">
- <code class="details" id="recognize">recognize(body, x__xgafv=None)</code>
+ <code class="details" id="recognize">recognize(body=None, x__xgafv=None)</code>
<pre>Performs synchronous speech recognition: receive results after all audio
has been sent and processed.
Args:
- body: object, The request body. (required)
+ body: object, The request body.
The object takes the form of:
{ # The top-level message sent by the client for the `Recognize` method.
- "audio": { # Contains audio data in the encoding specified in the `RecognitionConfig`. # *Required* The audio data to be recognized.
+ "audio": { # Contains audio data in the encoding specified in the `RecognitionConfig`. # Required. The audio data to be recognized.
# Either `content` or `uri` must be supplied. Supplying both or neither
# returns google.rpc.Code.INVALID_ARGUMENT. See
- # [content limits](/speech-to-text/quotas#content).
+ # [content limits](https://cloud.google.com/speech-to-text/quotas#content).
"content": "A String", # 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.
@@ -335,15 +345,16 @@
# google.rpc.Code.INVALID_ARGUMENT). For more information, see
# [Request URIs](https://cloud.google.com/storage/docs/reference-uris).
},
- "config": { # Provides information to the recognizer that specifies how to process the # *Required* Provides information to the recognizer that specifies how to
+ "config": { # Provides information to the recognizer that specifies how to process the # Required. Provides information to the recognizer that specifies how to
# process the request.
# request.
- "languageCode": "A String", # *Required* The language of the supplied audio as a
+ "languageCode": "A String", # 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](/speech-to-text/docs/languages)
- # for a list of the currently supported language codes.
- "audioChannelCount": 42, # *Optional* The number of channels in the input audio data.
+ # See [Language
+ # Support](https://cloud.google.com/speech-to-text/docs/languages) for a list
+ # of the currently supported language codes.
+ "audioChannelCount": 42, # 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'.
@@ -355,44 +366,34 @@
"encoding": "A String", # 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.
- "enableAutomaticPunctuation": True or False, # *Optional* If 'true', adds punctuation to recognition result hypotheses.
+ "enableAutomaticPunctuation": True or False, # 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.
- "enableSeparateRecognitionPerChannel": True or False, # This needs to be set to `true` explicitly and `audio_channel_count` > 1
+ "enableSeparateRecognitionPerChannel": True or False, # 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.
- "enableWordTimeOffsets": True or False, # *Optional* If `true`, the top result includes a list of words and
+ "enableWordTimeOffsets": True or False, # 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`.
- "maxAlternatives": 42, # *Optional* Maximum number of recognition hypotheses to be returned.
+ "maxAlternatives": 42, # 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.
- "useEnhanced": True or False, # *Optional* Set to true to use an enhanced model for speech recognition.
+ "useEnhanced": True or False, # 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:
- # 1. project is eligible for requesting enhanced models
- # 2. an enhanced model exists for the audio
+ # 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.
- #
- # Enhanced speech models require that you opt-in to data logging using
- # instructions in the
- # [documentation](/speech-to-text/docs/enable-data-logging). If you set
- # `use_enhanced` to true and you have not enabled audio logging, then you
- # will receive an error.
"sampleRateHertz": 42, # 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
@@ -400,53 +401,74 @@
# 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.
- "profanityFilter": True or False, # *Optional* If set to `true`, the server will attempt to filter out
+ "profanityFilter": True or False, # 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.
- "model": "A String", # *Optional* Which model to select for the given request. Select the model
+ "model": "A String", # 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
+ # <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.
+ # 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>
- "speechContexts": [ # *Optional* array of SpeechContext.
+ # recorded at a 16khz or greater sampling rate.</td>
+ # </tr>
+ # </table>
+ "diarizationConfig": { # Config to enable speaker diarization. # 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.
+ "minSpeakerCount": 42, # 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.
+ "enableSpeakerDiarization": True or False, # 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.
+ "maxSpeakerCount": 42, # 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.
+ "speakerTag": 42, # Output only. Unused.
+ },
+ "speechContexts": [ # Array of SpeechContext.
# A means to provide context to assist the speech recognition. For more
- # information, see [Phrase Hints](/speech-to-text/docs/basics#phrase-hints).
+ # information, see
+ # [speech
+ # adaptation](https://cloud.google.com/speech-to-text/docs/context-strength).
{ # Provides "hints" to the speech recognizer to favor specific words and phrases
# in the results.
- "phrases": [ # *Optional* A list of strings containing words and phrases "hints" so that
+ "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](/speech-to-text/quotas#content).
+ # [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
@@ -457,12 +479,10 @@
],
},
],
- "metadata": { # Description of audio data to be recognized. # *Optional* Metadata regarding this request.
+ "metadata": { # Description of audio data to be recognized. # Metadata regarding this request.
"recordingDeviceType": "A String", # The type of device the speech was recorded with.
"originalMediaType": "A String", # The original media the speech was recorded on.
"microphoneDistance": "A String", # The audio type that most closely describes the audio being recognized.
- "obfuscatedId": "A String", # Obfuscated (privacy-protected) ID of the user, to identify number of
- # unique users using the service.
"originalMimeType": "A String", # 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
@@ -492,42 +512,47 @@
{ # The only message returned to the client by the `Recognize` method. It
# contains the result as zero or more sequential `SpeechRecognitionResult`
# messages.
- "results": [ # Output only. Sequential list of transcription results corresponding to
+ "results": [ # Sequential list of transcription results corresponding to
# sequential portions of audio.
{ # A speech recognition result corresponding to a portion of the audio.
"channelTag": 42, # 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'.
- "alternatives": [ # Output only. May contain one or more recognition hypotheses (up to the
+ "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.
{ # Alternative hypotheses (a.k.a. n-best list).
- "confidence": 3.14, # Output only. The confidence estimate between 0.0 and 1.0. A higher number
+ "confidence": 3.14, # 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.
- "transcript": "A String", # Output only. Transcript text representing the words that the user spoke.
- "words": [ # Output only. A list of word-specific information for each recognized word.
+ "transcript": "A String", # Transcript text representing the words that the user spoke.
+ "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.
{ # Word-specific information for recognized words.
- "endTime": "A String", # Output only. Time offset relative to the beginning of the audio,
+ "endTime": "A String", # 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": "A String", # Output only. The word corresponding to this set of information.
- "startTime": "A String", # Output only. Time offset relative to the beginning of the audio,
+ "word": "A String", # The word corresponding to this set of information.
+ "startTime": "A String", # 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.
+ "speakerTag": 42, # Output only. 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.
},
],
},