# Cloud Speech-to-Text API . speech

## Instance Methods

Performs asynchronous speech recognition: receive results via the

Performs synchronous speech recognition: receive results after all audio

### Method Details

`longrunningrecognize(body, x__xgafv=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).

    Args:
      body: object, The request body. (required)
        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.
            # 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": "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.
          "uri": "A String", # 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).
        },
        "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
              # [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.
              # 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'.
          "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.
              # 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
              # 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
              # 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.
              # 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.
              # 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
              #
              # 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
              # 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.
          "profanityFilter": True or False, # *Optional* 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
              # 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.
              #
              #
              #     Model
              #     Description
              #
              #
              #     command_and_search
              #     Best for short queries such as voice commands or voice search.
              #
              #
              #     phone_call
              #     Best for audio that originated from a phone call (typically
              #     recorded at an 8khz sampling rate).
              #
              #
              #     video
              #     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.
              #
              #
              #     default
              #     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.
              #
              #
          "speechContexts": [ # *Optional* 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).
            { # 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
                  # 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).
                  #
                  # 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.
                "A String",
              ],
            },
          ],
          "metadata": { # Description of audio data to be recognized. # *Optional* 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
                # http://www.iana.org/assignments/media-types/media-types.xhtml#audio
            "industryNaicsCodeOfAudio": 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/.
            "audioTopic": "A String", # Description of the content. Eg. "Recordings of federal supreme court
                # hearings from 2012".
            "recordingDeviceName": "A String", # The device used to make the recording.  Examples 'Nexus 5X' or
                # 'Polycom SoundStation IP 6000' or 'POTS' or 'VoIP' or
                # 'Cardioid Microphone'.
            "interactionType": "A String", # The use case most closely describing the audio content to be recognized.
          },
        },
      }

      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.
        "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
            # 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).
          "message": "A String", # 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.
          "code": 42, # The status code, which should be an enum value of google.rpc.Code.
          "details": [ # A list of messages that carry the error details.  There is a common set of
              # message types for APIs to use.
            {
              "a_key": "", # Properties of the object. Contains field @type with type URL.
            },
          ],
        },
        "done": 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.
        "response": { # 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`.
          "a_key": "", # Properties of the object. Contains field @type with type URL.
        },
        "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.
        },
      }

`recognize(body, x__xgafv=None)`

    Performs synchronous speech recognition: receive results after all audio
    has been sent and processed.

    Args:
      body: object, The request body. (required)
        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.
            # 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": "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.
          "uri": "A String", # 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).
        },
        "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
              # [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.
              # 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'.
          "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.
              # 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
              # 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
              # 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.
              # 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.
              # 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
              #
              # 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
              # 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.
          "profanityFilter": True or False, # *Optional* 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
              # 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.
              #
              #
              #     Model
              #     Description
              #
              #
              #     command_and_search
              #     Best for short queries such as voice commands or voice search.
              #
              #
              #     phone_call
              #     Best for audio that originated from a phone call (typically
              #     recorded at an 8khz sampling rate).
              #
              #
              #     video
              #     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.
              #
              #
              #     default
              #     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.
              #
              #
          "speechContexts": [ # *Optional* 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).
            { # 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
                  # 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).
                  #
                  # 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.
                "A String",
              ],
            },
          ],
          "metadata": { # Description of audio data to be recognized. # *Optional* 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
                # http://www.iana.org/assignments/media-types/media-types.xhtml#audio
            "industryNaicsCodeOfAudio": 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/.
            "audioTopic": "A String", # Description of the content. Eg. "Recordings of federal supreme court
                # hearings from 2012".
            "recordingDeviceName": "A String", # The device used to make the recording.  Examples 'Nexus 5X' or
                # 'Polycom SoundStation IP 6000' or 'POTS' or 'VoIP' or
                # 'Cardioid Microphone'.
            "interactionType": "A String", # The use case most closely describing the audio content to be recognized.
          },
        },
      }

      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.
        "results": [ # Output only. 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
                # 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
                    # 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.
                    # 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,
                        # 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,
                        # 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.
                  },
                ],
              },
            ],
          },
        ],
      }