Module types (1.2.0)

API documentation for speech_v1p1beta1.types module.

API documentation for speech_v1p1beta1.types.Any class.

API documentation for speech_v1p1beta1.types.CancelOperationRequest class.

API documentation for speech_v1p1beta1.types.DeleteOperationRequest class.

API documentation for speech_v1p1beta1.types.Duration class.

API documentation for speech_v1p1beta1.types.GetOperationRequest class.

API documentation for speech_v1p1beta1.types.ListOperationsRequest class.

API documentation for speech_v1p1beta1.types.ListOperationsResponse class.

Describes the progress of a long-running LongRunningRecognize call. It is included in the metadata field of the Operation returned by the GetOperation call of the google::longrunning::Operations service.

Time when the request was received.

The top-level message sent by the client for the LongRunningRecognize method.

Required The audio data to be recognized.

The only message returned to the client by the LongRunningRecognize method. It contains the result as zero or more sequential SpeechRecognitionResult messages. It is included in the result.response field of the Operation returned by the GetOperation call of the google::longrunning::Operations service.

API documentation for speech_v1p1beta1.types.Operation class.

API documentation for speech_v1p1beta1.types.OperationInfo class.

Contains audio data in the encoding specified in the RecognitionConfig. Either content or uri must be supplied. Supplying both or neither returns [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT]. See content limits </speech-to-text/quotas#content>__.

The audio data bytes encoded as specified in RecognitionConfig. Note: as with all bytes fields, protobuffers use a pure binary representation, whereas JSON representations use base64.

Provides information to the recognizer that specifies how to process the request.

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 and required for all other audio formats. For details, see [AudioEncoding][google.cloud.speech.v1p1beta1.Rec ognitionConfig.AudioEncoding].

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.

Optional A list of up to 3 additional BCP-47 <https://www.rfc-editor.org/rfc/bcp/bcp47.txt> language tags, listing possible alternative languages of the supplied audio. See Language Support </speech-to- text/docs/languages> for a list of the currently supported language codes. If alternative languages are listed, recognition result will contain recognition in the most likely language detected including the main language_code. The recognition result will include the language tag of the language detected in the audio. Note: This feature is only supported for Voice Command and Voice Search use cases and performance may vary for other use cases (e.g., phone call transcription).

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.

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.

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.

Optional If set, specifies the estimated number of speakers in the conversation. If not set, defaults to '2'. Ignored unless enable_speaker_diarization is set to true."

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. .. raw:: html

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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.

Description of audio data to be recognized.

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/.

The original media the speech was recorded on.

The device used to make the recording. Examples 'Nexus 5X' or 'Polycom SoundStation IP 6000' or 'POTS' or 'VoIP' or 'Cardioid Microphone'.

Obfuscated (privacy-protected) ID of the user, to identify number of unique users using the service.

The top-level message sent by the client for the Recognize method.

Required The audio data to be recognized.

The only message returned to the client by the Recognize method. It contains the result as zero or more sequential SpeechRecognitionResult messages.

Provides "hints" to the speech recognizer to favor specific words and phrases in the results.

Hint Boost. Positive value will increase the probability that a specific phrase will be recognized over other similar sounding phrases. The higher the boost, the higher the chance of false positive recognition as well. Negative boost values would correspond to anti-biasing. Anti-biasing is not enabled, so negative boost will simply be ignored. Though boost can accept a wide range of positive values, most use cases are best served with values between 0 and 20. We recommend using a binary search approach to finding the optimal value for your use case.

Alternative hypotheses (a.k.a. n-best list).

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.

A speech recognition result corresponding to a portion of the audio.

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'.

API documentation for speech_v1p1beta1.types.Status class.

Provides information to the recognizer that specifies how to process the request.

Optional If false or omitted, the recognizer will perform continuous recognition (continuing to wait for and process audio even if the user pauses speaking) until the client closes the input stream (gRPC API) or until the maximum time limit has been reached. May return multiple StreamingRecognitionResult\ s with the is_final flag set to true. If true, the recognizer will detect a single spoken utterance. When it detects that the user has paused or stopped speaking, it will return an END_OF_SINGLE_UTTERANCE event and cease recognition. It will return no more than one StreamingRecognitionResult with the is_final flag set to true.

A streaming speech recognition result corresponding to a portion of the audio that is currently being processed.

Output only. If false, this StreamingRecognitionResult represents an interim result that may change. If true, this is the final time the speech service will return this particular StreamingRecognitionResult, the recognizer will not return any further hypotheses for this portion of the transcript and corresponding audio.

Output only. Time offset of the end of this result relative to the beginning of the audio.

Output only. The BCP-47 <https://www.rfc- editor.org/rfc/bcp/bcp47.txt>__ language tag of the language in this result. This language code was detected to have the most likelihood of being spoken in the audio.

The top-level message sent by the client for the StreamingRecognize method. Multiple StreamingRecognizeRequest messages are sent. The first message must contain a streaming_config message and must not contain audio data. All subsequent messages must contain audio data and must not contain a streaming_config message.

Provides information to the recognizer that specifies how to process the request. The first StreamingRecognizeRequest message must contain a streaming_config message.

StreamingRecognizeResponse is the only message returned to the client by StreamingRecognize. A series of zero or more StreamingRecognizeResponse messages are streamed back to the client. If there is no recognizable audio, and single_utterance is set to false, then no messages are streamed back to the client.

Here's an example of a series of ten StreamingRecognizeResponse\ s that might be returned while processing audio:

1. results { alternatives { transcript: "tube" } stability: 0.01 }

2. results { alternatives { transcript: "to be a" } stability: 0.01 }

3. results { alternatives { transcript: "to be" } stability: 0.9 } results { alternatives { transcript: " or not to be" } stability: 0.01 }

4. results { alternatives { transcript: "to be or not to be" confidence: 0.92 } alternatives { transcript: "to bee or not to bee" } is_final: true }

5. results { alternatives { transcript: " that's" } stability: 0.01 }

6. results { alternatives { transcript: " that is" } stability: 0.9 } results { alternatives { transcript: " the question" } stability: 0.01 }

7. results { alternatives { transcript: " that is the question" confidence: 0.98 } alternatives { transcript: " that was the question" } is_final: true }

Notes:

• Only two of the above responses #4 and #7 contain final results; they are indicated by is_final: true. Concatenating these together generates the full transcript: "to be or not to be that is the question".

• The others contain interim results. #3 and #6 contain two interim results: the first portion has a high stability and is less likely to change; the second portion has a low stability and is very likely to change. A UI designer might choose to show only high stability results.

• The specific stability and confidence values shown above are only for illustrative purposes. Actual values may vary.

• In each response, only one of these fields will be set: error, speech_event_type, or one or more (repeated) results.

  Output only. This repeated list contains zero or more results that correspond to consecutive portions of the audio currently being processed. It contains zero or one is_final=true result (the newly settled portion), followed by zero or more is_final=false results (the interim results).

API documentation for speech_v1p1beta1.types.WaitOperationRequest class.

Word-specific information for recognized words.

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.

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.