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<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=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=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=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
a `LongRunningRecognizeResponse` message.
For more information on asynchronous speech recognition, see the
[how-to](https://cloud.google.com/speech-to-text/docs/async-recognize).

Args:
  body: object, The request body.
    The object takes the form of:

{ # The top-level message sent by the client for the `LongRunningRecognize`
      # method.
    &quot;config&quot;: { # 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.
      &quot;enableWordTimeOffsets&quot;: 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`.
      &quot;diarizationConfig&quot;: { # 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.
        &quot;minSpeakerCount&quot;: 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.
        &quot;maxSpeakerCount&quot;: 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.
        &quot;speakerTag&quot;: 42, # Output only. Unused.
        &quot;enableSpeakerDiarization&quot;: True or False, # If &#x27;true&#x27;, enables speaker detection for each recognized word in
            # the top alternative of the recognition result using a speaker_tag provided
            # in the WordInfo.
      },
      &quot;languageCode&quot;: &quot;A String&quot;, # Required. The language of the supplied audio as a
          # [BCP-47](https://www.rfc-editor.org/rfc/bcp/bcp47.txt) language tag.
          # Example: &quot;en-US&quot;.
          # See [Language
          # Support](https://cloud.google.com/speech-to-text/docs/languages) for a list
          # of the currently supported language codes.
      &quot;profanityFilter&quot;: 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. &quot;f***&quot;. If set to `false` or omitted, profanities
          # won&#x27;t be filtered out.
      &quot;useEnhanced&quot;: 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 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.
      &quot;metadata&quot;: { # Description of audio data to be recognized. # Metadata regarding this request.
        &quot;originalMediaType&quot;: &quot;A String&quot;, # The original media the speech was recorded on.
        &quot;recordingDeviceType&quot;: &quot;A String&quot;, # The type of device the speech was recorded with.
        &quot;interactionType&quot;: &quot;A String&quot;, # The use case most closely describing the audio content to be recognized.
        &quot;audioTopic&quot;: &quot;A String&quot;, # Description of the content. Eg. &quot;Recordings of federal supreme court
            # hearings from 2012&quot;.
        &quot;originalMimeType&quot;: &quot;A String&quot;, # 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
        &quot;recordingDeviceName&quot;: &quot;A String&quot;, # The device used to make the recording.  Examples &#x27;Nexus 5X&#x27; or
            # &#x27;Polycom SoundStation IP 6000&#x27; or &#x27;POTS&#x27; or &#x27;VoIP&#x27; or
            # &#x27;Cardioid Microphone&#x27;.
        &quot;industryNaicsCodeOfAudio&quot;: 42, # 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/.
        &quot;microphoneDistance&quot;: &quot;A String&quot;, # The audio type that most closely describes the audio being recognized.
      },
      &quot;sampleRateHertz&quot;: 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
          # source to 16000 Hz. If that&#x27;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.
      &quot;enableSeparateRecognitionPerChannel&quot;: True or False, # This needs to be set to `true` explicitly and `audio_channel_count` &gt; 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.
      &quot;enableAutomaticPunctuation&quot;: True or False, # If &#x27;true&#x27;, 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 &#x27;false&#x27; value does not add punctuation to result hypotheses.
      &quot;maxAlternatives&quot;: 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.
      &quot;speechContexts&quot;: [ # Array of 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).
        { # Provides &quot;hints&quot; to the speech recognizer to favor specific words and phrases
            # in the results.
          &quot;phrases&quot;: [ # A list of strings containing words and phrases &quot;hints&quot; 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.
            &quot;A String&quot;,
          ],
        },
      ],
      &quot;encoding&quot;: &quot;A String&quot;, # 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.
      &quot;model&quot;: &quot;A String&quot;, # 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.
          # &lt;table&gt;
          #   &lt;tr&gt;
          #     &lt;td&gt;&lt;b&gt;Model&lt;/b&gt;&lt;/td&gt;
          #     &lt;td&gt;&lt;b&gt;Description&lt;/b&gt;&lt;/td&gt;
          #   &lt;/tr&gt;
          #   &lt;tr&gt;
          #     &lt;td&gt;&lt;code&gt;command_and_search&lt;/code&gt;&lt;/td&gt;
          #     &lt;td&gt;Best for short queries such as voice commands or voice search.&lt;/td&gt;
          #   &lt;/tr&gt;
          #   &lt;tr&gt;
          #     &lt;td&gt;&lt;code&gt;phone_call&lt;/code&gt;&lt;/td&gt;
          #     &lt;td&gt;Best for audio that originated from a phone call (typically
          #     recorded at an 8khz sampling rate).&lt;/td&gt;
          #   &lt;/tr&gt;
          #   &lt;tr&gt;
          #     &lt;td&gt;&lt;code&gt;video&lt;/code&gt;&lt;/td&gt;
          #     &lt;td&gt;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.&lt;/td&gt;
          #   &lt;/tr&gt;
          #   &lt;tr&gt;
          #     &lt;td&gt;&lt;code&gt;default&lt;/code&gt;&lt;/td&gt;
          #     &lt;td&gt;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.&lt;/td&gt;
          #   &lt;/tr&gt;
          # &lt;/table&gt;
      &quot;audioChannelCount&quot;: 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 &#x27;1&#x27;-&#x27;254&#x27;.
          # 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 &#x27;true&#x27;.
    },
    &quot;audio&quot;: { # 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](https://cloud.google.com/speech-to-text/quotas#content).
      &quot;content&quot;: &quot;A String&quot;, # 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.
      &quot;uri&quot;: &quot;A String&quot;, # 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). For more information, see
          # [Request URIs](https://cloud.google.com/storage/docs/reference-uris).
    },
  }

  x__xgafv: string, V1 error format.
    Allowed values
      1 - v1 error format
      2 - v2 error format

Returns:
  An object of the form:

    { # This resource represents a long-running operation that is the result of a
      # network API call.
    &quot;response&quot;: { # The normal response of the operation in case of success.  If the original
        # method returns no data on success, such as `Delete`, the response is
        # `google.protobuf.Empty`.  If the original method is standard
        # `Get`/`Create`/`Update`, the response should be the resource.  For other
        # methods, the response should have the type `XxxResponse`, where `Xxx`
        # is the original method name.  For example, if the original method name
        # is `TakeSnapshot()`, the inferred response type is
        # `TakeSnapshotResponse`.
      &quot;a_key&quot;: &quot;&quot;, # Properties of the object. Contains field @type with type URL.
    },
    &quot;name&quot;: &quot;A String&quot;, # 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}`.
    &quot;error&quot;: { # 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
        # three pieces of data: error code, error message, and error details.
        #
        # You can find out more about this error model and how to work with it in the
        # [API Design Guide](https://cloud.google.com/apis/design/errors).
      &quot;details&quot;: [ # A list of messages that carry the error details.  There is a common set of
          # message types for APIs to use.
        {
          &quot;a_key&quot;: &quot;&quot;, # Properties of the object. Contains field @type with type URL.
        },
      ],
      &quot;code&quot;: 42, # The status code, which should be an enum value of google.rpc.Code.
      &quot;message&quot;: &quot;A String&quot;, # A developer-facing error message, which should be in English. Any
          # user-facing error message should be localized and sent in the
          # google.rpc.Status.details field, or localized by the client.
    },
    &quot;metadata&quot;: { # 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.
      &quot;a_key&quot;: &quot;&quot;, # Properties of the object. Contains field @type with type URL.
    },
    &quot;done&quot;: True or False, # If the value is `false`, it means the operation is still in progress.
        # If `true`, the operation is completed, and either `error` or `response` is
        # available.
  }</pre>
</div>

<div class="method">
    <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.
    The object takes the form of:

{ # The top-level message sent by the client for the `Recognize` method.
    &quot;config&quot;: { # 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.
      &quot;enableWordTimeOffsets&quot;: 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`.
      &quot;diarizationConfig&quot;: { # 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.
        &quot;minSpeakerCount&quot;: 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.
        &quot;maxSpeakerCount&quot;: 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.
        &quot;speakerTag&quot;: 42, # Output only. Unused.
        &quot;enableSpeakerDiarization&quot;: True or False, # If &#x27;true&#x27;, enables speaker detection for each recognized word in
            # the top alternative of the recognition result using a speaker_tag provided
            # in the WordInfo.
      },
      &quot;languageCode&quot;: &quot;A String&quot;, # Required. The language of the supplied audio as a
          # [BCP-47](https://www.rfc-editor.org/rfc/bcp/bcp47.txt) language tag.
          # Example: &quot;en-US&quot;.
          # See [Language
          # Support](https://cloud.google.com/speech-to-text/docs/languages) for a list
          # of the currently supported language codes.
      &quot;profanityFilter&quot;: 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. &quot;f***&quot;. If set to `false` or omitted, profanities
          # won&#x27;t be filtered out.
      &quot;useEnhanced&quot;: 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 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.
      &quot;metadata&quot;: { # Description of audio data to be recognized. # Metadata regarding this request.
        &quot;originalMediaType&quot;: &quot;A String&quot;, # The original media the speech was recorded on.
        &quot;recordingDeviceType&quot;: &quot;A String&quot;, # The type of device the speech was recorded with.
        &quot;interactionType&quot;: &quot;A String&quot;, # The use case most closely describing the audio content to be recognized.
        &quot;audioTopic&quot;: &quot;A String&quot;, # Description of the content. Eg. &quot;Recordings of federal supreme court
            # hearings from 2012&quot;.
        &quot;originalMimeType&quot;: &quot;A String&quot;, # 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
        &quot;recordingDeviceName&quot;: &quot;A String&quot;, # The device used to make the recording.  Examples &#x27;Nexus 5X&#x27; or
            # &#x27;Polycom SoundStation IP 6000&#x27; or &#x27;POTS&#x27; or &#x27;VoIP&#x27; or
            # &#x27;Cardioid Microphone&#x27;.
        &quot;industryNaicsCodeOfAudio&quot;: 42, # 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/.
        &quot;microphoneDistance&quot;: &quot;A String&quot;, # The audio type that most closely describes the audio being recognized.
      },
      &quot;sampleRateHertz&quot;: 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
          # source to 16000 Hz. If that&#x27;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.
      &quot;enableSeparateRecognitionPerChannel&quot;: True or False, # This needs to be set to `true` explicitly and `audio_channel_count` &gt; 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.
      &quot;enableAutomaticPunctuation&quot;: True or False, # If &#x27;true&#x27;, 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 &#x27;false&#x27; value does not add punctuation to result hypotheses.
      &quot;maxAlternatives&quot;: 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.
      &quot;speechContexts&quot;: [ # Array of 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).
        { # Provides &quot;hints&quot; to the speech recognizer to favor specific words and phrases
            # in the results.
          &quot;phrases&quot;: [ # A list of strings containing words and phrases &quot;hints&quot; 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.
            &quot;A String&quot;,
          ],
        },
      ],
      &quot;encoding&quot;: &quot;A String&quot;, # 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.
      &quot;model&quot;: &quot;A String&quot;, # 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.
          # &lt;table&gt;
          #   &lt;tr&gt;
          #     &lt;td&gt;&lt;b&gt;Model&lt;/b&gt;&lt;/td&gt;
          #     &lt;td&gt;&lt;b&gt;Description&lt;/b&gt;&lt;/td&gt;
          #   &lt;/tr&gt;
          #   &lt;tr&gt;
          #     &lt;td&gt;&lt;code&gt;command_and_search&lt;/code&gt;&lt;/td&gt;
          #     &lt;td&gt;Best for short queries such as voice commands or voice search.&lt;/td&gt;
          #   &lt;/tr&gt;
          #   &lt;tr&gt;
          #     &lt;td&gt;&lt;code&gt;phone_call&lt;/code&gt;&lt;/td&gt;
          #     &lt;td&gt;Best for audio that originated from a phone call (typically
          #     recorded at an 8khz sampling rate).&lt;/td&gt;
          #   &lt;/tr&gt;
          #   &lt;tr&gt;
          #     &lt;td&gt;&lt;code&gt;video&lt;/code&gt;&lt;/td&gt;
          #     &lt;td&gt;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.&lt;/td&gt;
          #   &lt;/tr&gt;
          #   &lt;tr&gt;
          #     &lt;td&gt;&lt;code&gt;default&lt;/code&gt;&lt;/td&gt;
          #     &lt;td&gt;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.&lt;/td&gt;
          #   &lt;/tr&gt;
          # &lt;/table&gt;
      &quot;audioChannelCount&quot;: 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 &#x27;1&#x27;-&#x27;254&#x27;.
          # 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 &#x27;true&#x27;.
    },
    &quot;audio&quot;: { # 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](https://cloud.google.com/speech-to-text/quotas#content).
      &quot;content&quot;: &quot;A String&quot;, # 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.
      &quot;uri&quot;: &quot;A String&quot;, # 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). For more information, see
          # [Request URIs](https://cloud.google.com/storage/docs/reference-uris).
    },
  }

  x__xgafv: string, V1 error format.
    Allowed values
      1 - v1 error format
      2 - v2 error format

Returns:
  An object of the form:

    { # The only message returned to the client by the `Recognize` method. It
      # contains the result as zero or more sequential `SpeechRecognitionResult`
      # messages.
    &quot;results&quot;: [ # Sequential list of transcription results corresponding to
        # sequential portions of audio.
      { # A speech recognition result corresponding to a portion of the audio.
        &quot;channelTag&quot;: 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 &#x27;1&#x27; to &#x27;N&#x27;.
        &quot;alternatives&quot;: [ # 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).
            &quot;confidence&quot;: 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.
            &quot;transcript&quot;: &quot;A String&quot;, # Transcript text representing the words that the user spoke.
            &quot;words&quot;: [ # 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.
                &quot;endTime&quot;: &quot;A String&quot;, # 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.
                &quot;startTime&quot;: &quot;A String&quot;, # 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.
                &quot;word&quot;: &quot;A String&quot;, # The word corresponding to this set of information.
                &quot;speakerTag&quot;: 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 &#x27;1&#x27; to diarization_speaker_count.
                    # speaker_tag is set if enable_speaker_diarization = &#x27;true&#x27; and only in the
                    # top alternative.
              },
            ],
          },
        ],
      },
    ],
  }</pre>
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