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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>
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<h2>Instance Methods</h2>
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<p class="toc_element">
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<code><a href="#longrunningrecognize">longrunningrecognize(body, x__xgafv=None)</a></code></p>
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<p class="firstline">Performs asynchronous speech recognition: receive results via the</p>
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<p class="toc_element">
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<code><a href="#recognize">recognize(body, x__xgafv=None)</a></code></p>
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<p class="firstline">Performs synchronous speech recognition: receive results after all audio</p>
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<h3>Method Details</h3>
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<div class="method">
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<code class="details" id="longrunningrecognize">longrunningrecognize(body, x__xgafv=None)</code>
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<pre>Performs asynchronous speech recognition: receive results via the
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google.longrunning.Operations interface. Returns either an
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`Operation.error` or an `Operation.response` which contains
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a `LongRunningRecognizeResponse` message.
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For more information on asynchronous speech recognition, see the
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[how-to](https://cloud.google.com/speech-to-text/docs/async-recognize).
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Args:
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body: object, The request body. (required)
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The object takes the form of:
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{ # The top-level message sent by the client for the `LongRunningRecognize`
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# method.
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"audio": { # Contains audio data in the encoding specified in the `RecognitionConfig`. # *Required* The audio data to be recognized.
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# Either `content` or `uri` must be supplied. Supplying both or neither
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# returns google.rpc.Code.INVALID_ARGUMENT. See
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# [content limits](/speech-to-text/quotas#content).
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"content": "A String", # The audio data bytes encoded as specified in
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# `RecognitionConfig`. Note: as with all bytes fields, proto buffers use a
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# pure binary representation, whereas JSON representations use base64.
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"uri": "A String", # URI that points to a file that contains audio data bytes as specified in
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# `RecognitionConfig`. The file must not be compressed (for example, gzip).
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# Currently, only Google Cloud Storage URIs are
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# supported, which must be specified in the following format:
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# `gs://bucket_name/object_name` (other URI formats return
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# google.rpc.Code.INVALID_ARGUMENT). For more information, see
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# [Request URIs](https://cloud.google.com/storage/docs/reference-uris).
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},
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"config": { # Provides information to the recognizer that specifies how to process the # *Required* Provides information to the recognizer that specifies how to
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# process the request.
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# request.
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"languageCode": "A String", # *Required* The language of the supplied audio as a
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# [BCP-47](https://www.rfc-editor.org/rfc/bcp/bcp47.txt) language tag.
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# Example: "en-US".
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# See [Language Support](/speech-to-text/docs/languages)
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# for a list of the currently supported language codes.
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"audioChannelCount": 42, # *Optional* The number of channels in the input audio data.
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# ONLY set this for MULTI-CHANNEL recognition.
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# Valid values for LINEAR16 and FLAC are `1`-`8`.
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# Valid values for OGG_OPUS are '1'-'254'.
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# Valid value for MULAW, AMR, AMR_WB and SPEEX_WITH_HEADER_BYTE is only `1`.
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# If `0` or omitted, defaults to one channel (mono).
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# Note: We only recognize the first channel by default.
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# To perform independent recognition on each channel set
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# `enable_separate_recognition_per_channel` to 'true'.
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"encoding": "A String", # Encoding of audio data sent in all `RecognitionAudio` messages.
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# This field is optional for `FLAC` and `WAV` audio files and required
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# for all other audio formats. For details, see AudioEncoding.
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"enableAutomaticPunctuation": True or False, # *Optional* If 'true', adds punctuation to recognition result hypotheses.
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# This feature is only available in select languages. Setting this for
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# requests in other languages has no effect at all.
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# The default 'false' value does not add punctuation to result hypotheses.
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# Note: This is currently offered as an experimental service, complimentary
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# to all users. In the future this may be exclusively available as a
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# premium feature.
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"enableSeparateRecognitionPerChannel": True or False, # This needs to be set to `true` explicitly and `audio_channel_count` > 1
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# to get each channel recognized separately. The recognition result will
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# contain a `channel_tag` field to state which channel that result belongs
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# to. If this is not true, we will only recognize the first channel. The
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# request is billed cumulatively for all channels recognized:
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# `audio_channel_count` multiplied by the length of the audio.
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"enableWordTimeOffsets": True or False, # *Optional* If `true`, the top result includes a list of words and
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# the start and end time offsets (timestamps) for those words. If
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# `false`, no word-level time offset information is returned. The default is
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# `false`.
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"maxAlternatives": 42, # *Optional* Maximum number of recognition hypotheses to be returned.
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# Specifically, the maximum number of `SpeechRecognitionAlternative` messages
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# within each `SpeechRecognitionResult`.
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# The server may return fewer than `max_alternatives`.
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# Valid values are `0`-`30`. A value of `0` or `1` will return a maximum of
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# one. If omitted, will return a maximum of one.
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"useEnhanced": True or False, # *Optional* Set to true to use an enhanced model for speech recognition.
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# If `use_enhanced` is set to true and the `model` field is not set, then
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# an appropriate enhanced model is chosen if:
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# 1. project is eligible for requesting enhanced models
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# 2. an enhanced model exists for the audio
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#
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# If `use_enhanced` is true and an enhanced version of the specified model
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# does not exist, then the speech is recognized using the standard version
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# of the specified model.
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#
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# Enhanced speech models require that you opt-in to data logging using
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# instructions in the
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# [documentation](/speech-to-text/docs/enable-data-logging). If you set
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# `use_enhanced` to true and you have not enabled audio logging, then you
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# will receive an error.
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"sampleRateHertz": 42, # Sample rate in Hertz of the audio data sent in all
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# `RecognitionAudio` messages. Valid values are: 8000-48000.
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# 16000 is optimal. For best results, set the sampling rate of the audio
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# source to 16000 Hz. If that's not possible, use the native sample rate of
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# the audio source (instead of re-sampling).
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# This field is optional for FLAC and WAV audio files, but is
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# required for all other audio formats. For details, see AudioEncoding.
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"profanityFilter": True or False, # *Optional* If set to `true`, the server will attempt to filter out
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# profanities, replacing all but the initial character in each filtered word
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# with asterisks, e.g. "f***". If set to `false` or omitted, profanities
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# won't be filtered out.
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"model": "A String", # *Optional* Which model to select for the given request. Select the model
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# best suited to your domain to get best results. If a model is not
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# explicitly specified, then we auto-select a model based on the parameters
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# in the RecognitionConfig.
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# <table>
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# <tr>
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# <td><b>Model</b></td>
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# <td><b>Description</b></td>
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# </tr>
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# <tr>
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# <td><code>command_and_search</code></td>
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# <td>Best for short queries such as voice commands or voice search.</td>
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# </tr>
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# <tr>
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# <td><code>phone_call</code></td>
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# <td>Best for audio that originated from a phone call (typically
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# recorded at an 8khz sampling rate).</td>
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# </tr>
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# <tr>
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# <td><code>video</code></td>
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# <td>Best for audio that originated from from video or includes multiple
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# speakers. Ideally the audio is recorded at a 16khz or greater
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# sampling rate. This is a premium model that costs more than the
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# standard rate.</td>
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# </tr>
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# <tr>
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# <td><code>default</code></td>
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# <td>Best for audio that is not one of the specific audio models.
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# For example, long-form audio. Ideally the audio is high-fidelity,
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# recorded at a 16khz or greater sampling rate.</td>
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# </tr>
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# </table>
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"speechContexts": [ # *Optional* array of SpeechContext.
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# A means to provide context to assist the speech recognition. For more
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# information, see [Phrase Hints](/speech-to-text/docs/basics#phrase-hints).
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{ # Provides "hints" to the speech recognizer to favor specific words and phrases
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# in the results.
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"phrases": [ # *Optional* A list of strings containing words and phrases "hints" so that
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# the speech recognition is more likely to recognize them. This can be used
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# to improve the accuracy for specific words and phrases, for example, if
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# specific commands are typically spoken by the user. This can also be used
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# to add additional words to the vocabulary of the recognizer. See
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# [usage limits](/speech-to-text/quotas#content).
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#
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# List items can also be set to classes for groups of words that represent
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# common concepts that occur in natural language. For example, rather than
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# providing phrase hints for every month of the year, using the $MONTH class
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# improves the likelihood of correctly transcribing audio that includes
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# months.
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"A String",
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],
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},
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],
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"metadata": { # Description of audio data to be recognized. # *Optional* Metadata regarding this request.
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"recordingDeviceType": "A String", # The type of device the speech was recorded with.
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"originalMediaType": "A String", # The original media the speech was recorded on.
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"microphoneDistance": "A String", # The audio type that most closely describes the audio being recognized.
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"obfuscatedId": "A String", # Obfuscated (privacy-protected) ID of the user, to identify number of
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# unique users using the service.
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"originalMimeType": "A String", # Mime type of the original audio file. For example `audio/m4a`,
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# `audio/x-alaw-basic`, `audio/mp3`, `audio/3gpp`.
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# A list of possible audio mime types is maintained at
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# http://www.iana.org/assignments/media-types/media-types.xhtml#audio
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"industryNaicsCodeOfAudio": 42, # The industry vertical to which this speech recognition request most
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# closely applies. This is most indicative of the topics contained
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# in the audio. Use the 6-digit NAICS code to identify the industry
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# vertical - see https://www.naics.com/search/.
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"audioTopic": "A String", # Description of the content. Eg. "Recordings of federal supreme court
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# hearings from 2012".
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"recordingDeviceName": "A String", # The device used to make the recording. Examples 'Nexus 5X' or
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# 'Polycom SoundStation IP 6000' or 'POTS' or 'VoIP' or
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# 'Cardioid Microphone'.
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"interactionType": "A String", # The use case most closely describing the audio content to be recognized.
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},
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},
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}
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x__xgafv: string, V1 error format.
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Allowed values
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1 - v1 error format
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2 - v2 error format
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Returns:
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An object of the form:
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{ # This resource represents a long-running operation that is the result of a
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# network API call.
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"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.
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# different programming environments, including REST APIs and RPC APIs. It is
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# used by [gRPC](https://github.com/grpc). Each `Status` message contains
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# three pieces of data: error code, error message, and error details.
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#
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# You can find out more about this error model and how to work with it in the
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# [API Design Guide](https://cloud.google.com/apis/design/errors).
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"message": "A String", # A developer-facing error message, which should be in English. Any
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# user-facing error message should be localized and sent in the
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# google.rpc.Status.details field, or localized by the client.
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"code": 42, # The status code, which should be an enum value of google.rpc.Code.
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"details": [ # A list of messages that carry the error details. There is a common set of
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# message types for APIs to use.
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{
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"a_key": "", # Properties of the object. Contains field @type with type URL.
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},
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],
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},
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"done": True or False, # If the value is `false`, it means the operation is still in progress.
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# If `true`, the operation is completed, and either `error` or `response` is
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# available.
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"response": { # The normal response of the operation in case of success. If the original
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# method returns no data on success, such as `Delete`, the response is
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# `google.protobuf.Empty`. If the original method is standard
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# `Get`/`Create`/`Update`, the response should be the resource. For other
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# methods, the response should have the type `XxxResponse`, where `Xxx`
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# is the original method name. For example, if the original method name
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# is `TakeSnapshot()`, the inferred response type is
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# `TakeSnapshotResponse`.
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"a_key": "", # Properties of the object. Contains field @type with type URL.
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},
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"name": "A String", # The server-assigned name, which is only unique within the same service that
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# originally returns it. If you use the default HTTP mapping, the
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# `name` should be a resource name ending with `operations/{unique_id}`.
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"metadata": { # Service-specific metadata associated with the operation. It typically
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# contains progress information and common metadata such as create time.
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# Some services might not provide such metadata. Any method that returns a
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# long-running operation should document the metadata type, if any.
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"a_key": "", # Properties of the object. Contains field @type with type URL.
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},
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}</pre>
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</div>
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<div class="method">
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<code class="details" id="recognize">recognize(body, x__xgafv=None)</code>
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<pre>Performs synchronous speech recognition: receive results after all audio
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has been sent and processed.
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Args:
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body: object, The request body. (required)
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The object takes the form of:
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{ # The top-level message sent by the client for the `Recognize` method.
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"audio": { # Contains audio data in the encoding specified in the `RecognitionConfig`. # *Required* The audio data to be recognized.
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# Either `content` or `uri` must be supplied. Supplying both or neither
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# returns google.rpc.Code.INVALID_ARGUMENT. See
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# [content limits](/speech-to-text/quotas#content).
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"content": "A String", # The audio data bytes encoded as specified in
|
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# `RecognitionConfig`. Note: as with all bytes fields, proto buffers use a
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# pure binary representation, whereas JSON representations use base64.
|
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"uri": "A String", # URI that points to a file that contains audio data bytes as specified in
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# `RecognitionConfig`. The file must not be compressed (for example, gzip).
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# Currently, only Google Cloud Storage URIs are
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# supported, which must be specified in the following format:
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# `gs://bucket_name/object_name` (other URI formats return
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# google.rpc.Code.INVALID_ARGUMENT). For more information, see
|
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# [Request URIs](https://cloud.google.com/storage/docs/reference-uris).
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},
|
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"config": { # Provides information to the recognizer that specifies how to process the # *Required* Provides information to the recognizer that specifies how to
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# process the request.
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# request.
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"languageCode": "A String", # *Required* The language of the supplied audio as a
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# [BCP-47](https://www.rfc-editor.org/rfc/bcp/bcp47.txt) language tag.
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|
# Example: "en-US".
|
|
# See [Language Support](/speech-to-text/docs/languages)
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# for a list of the currently supported language codes.
|
|
"audioChannelCount": 42, # *Optional* The number of channels in the input audio data.
|
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# 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`.
|
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# 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
|
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# `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
|
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# 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
|
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# premium feature.
|
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"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`.
|
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# 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
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# `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.
|
|
# <table>
|
|
# <tr>
|
|
# <td><b>Model</b></td>
|
|
# <td><b>Description</b></td>
|
|
# </tr>
|
|
# <tr>
|
|
# <td><code>command_and_search</code></td>
|
|
# <td>Best for short queries such as voice commands or voice search.</td>
|
|
# </tr>
|
|
# <tr>
|
|
# <td><code>phone_call</code></td>
|
|
# <td>Best for audio that originated from a phone call (typically
|
|
# recorded at an 8khz sampling rate).</td>
|
|
# </tr>
|
|
# <tr>
|
|
# <td><code>video</code></td>
|
|
# <td>Best for audio that originated from from video or includes multiple
|
|
# speakers. Ideally the audio is recorded at a 16khz or greater
|
|
# sampling rate. This is a premium model that costs more than the
|
|
# standard rate.</td>
|
|
# </tr>
|
|
# <tr>
|
|
# <td><code>default</code></td>
|
|
# <td>Best for audio that is not one of the specific audio models.
|
|
# For example, long-form audio. Ideally the audio is high-fidelity,
|
|
# recorded at a 16khz or greater sampling rate.</td>
|
|
# </tr>
|
|
# </table>
|
|
"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.
|
|
},
|
|
],
|
|
},
|
|
],
|
|
},
|
|
],
|
|
}</pre>
|
|
</div>
|
|
|
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</body></html> |