On January 1, 2020 this library will no longer support Python 2 on the latest released version. Previously released library versions will continue to be available. For more information please visit Python 2 support on Google Cloud.

Types for Cloud Speech-to-Text API Client

class google.cloud.speech_v1.types.Any
type_url

Field google.protobuf.Any.type_url

value

Field google.protobuf.Any.value

class google.cloud.speech_v1.types.CancelOperationRequest
name

Field google.longrunning.CancelOperationRequest.name

class google.cloud.speech_v1.types.DeleteOperationRequest
name

Field google.longrunning.DeleteOperationRequest.name

class google.cloud.speech_v1.types.Duration
nanos

Field google.protobuf.Duration.nanos

seconds

Field google.protobuf.Duration.seconds

class google.cloud.speech_v1.types.GetOperationRequest
name

Field google.longrunning.GetOperationRequest.name

class google.cloud.speech_v1.types.ListOperationsRequest
filter

Field google.longrunning.ListOperationsRequest.filter

name

Field google.longrunning.ListOperationsRequest.name

page_size

Field google.longrunning.ListOperationsRequest.page_size

page_token

Field google.longrunning.ListOperationsRequest.page_token

class google.cloud.speech_v1.types.ListOperationsResponse
next_page_token

Field google.longrunning.ListOperationsResponse.next_page_token

operations

Field google.longrunning.ListOperationsResponse.operations

class google.cloud.speech_v1.types.LongRunningRecognizeMetadata

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.

progress_percent

Approximate percentage of audio processed thus far. Guaranteed to be 100 when the audio is fully processed and the results are available.

start_time

Time when the request was received.

last_update_time

Time of the most recent processing update.

last_update_time

Field google.cloud.speech.v1.LongRunningRecognizeMetadata.last_update_time

progress_percent

Field google.cloud.speech.v1.LongRunningRecognizeMetadata.progress_percent

start_time

Field google.cloud.speech.v1.LongRunningRecognizeMetadata.start_time

class google.cloud.speech_v1.types.LongRunningRecognizeRequest

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

config

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

audio

Required. The audio data to be recognized.

audio

Field google.cloud.speech.v1.LongRunningRecognizeRequest.audio

config

Field google.cloud.speech.v1.LongRunningRecognizeRequest.config

class google.cloud.speech_v1.types.LongRunningRecognizeResponse

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.

results

Sequential list of transcription results corresponding to sequential portions of audio.

results

Field google.cloud.speech.v1.LongRunningRecognizeResponse.results

class google.cloud.speech_v1.types.Operation
deserialize()

Creates new method instance from given serialized data.

done

Field google.longrunning.Operation.done

error

Field google.longrunning.Operation.error

metadata

Field google.longrunning.Operation.metadata

name

Field google.longrunning.Operation.name

response

Field google.longrunning.Operation.response

class google.cloud.speech_v1.types.OperationInfo
metadata_type

Field google.longrunning.OperationInfo.metadata_type

response_type

Field google.longrunning.OperationInfo.response_type

class google.cloud.speech_v1.types.RecognitionAudio

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.

audio_source

The audio source, which is either inline content or a Google Cloud Storage uri.

content

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

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][google.rpc. Code.INVALID_ARGUMENT]). For more information, see Request URIs.

content

Field google.cloud.speech.v1.RecognitionAudio.content

uri

Field google.cloud.speech.v1.RecognitionAudio.uri

class google.cloud.speech_v1.types.RecognitionConfig

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

encoding

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][google.cloud.speech.v1.Recognitio nConfig.AudioEncoding].

sample_rate_hertz

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][google.cloud.speech.v1.Recognitio nConfig.AudioEncoding].

audio_channel_count

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’.

enable_separate_recognition_per_channel

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.

language_code

Required. The language of the supplied audio as a BCP-47 language tag. Example: “en-US”. See Language Support for a list of the currently supported language codes.

max_alternatives

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.

profanity_filter

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.

speech_contexts

Array of [SpeechContext][google.cloud.speech.v1.SpeechContext]. A means to provide context to assist the speech recognition. For more information, see speech adaptation.

enable_word_time_offsets

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.

enable_automatic_punctuation

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.

diarization_config

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.

metadata

Metadata regarding this request.

model

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 <table> .. raw:: html <tr> :: <td><b>Model</b></td> <td><b>Description</b></td> .. raw:: html </tr> .. raw:: html <tr> :: <td><code>command_and_search</code></td> <td>Best for short queries such as voice commands or voice search.</td> .. raw:: html </tr> .. raw:: html <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> .. raw:: html </tr> .. raw:: html <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> .. raw:: html </tr> .. raw:: html <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> .. raw:: html </tr> .. raw:: html </table>

use_enhanced

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.

audio_channel_count

Field google.cloud.speech.v1.RecognitionConfig.audio_channel_count

diarization_config

Field google.cloud.speech.v1.RecognitionConfig.diarization_config

enable_automatic_punctuation

Field google.cloud.speech.v1.RecognitionConfig.enable_automatic_punctuation

enable_separate_recognition_per_channel

Field google.cloud.speech.v1.RecognitionConfig.enable_separate_recognition_per_channel

enable_word_time_offsets

Field google.cloud.speech.v1.RecognitionConfig.enable_word_time_offsets

encoding

Field google.cloud.speech.v1.RecognitionConfig.encoding

language_code

Field google.cloud.speech.v1.RecognitionConfig.language_code

max_alternatives

Field google.cloud.speech.v1.RecognitionConfig.max_alternatives

metadata

Field google.cloud.speech.v1.RecognitionConfig.metadata

model

Field google.cloud.speech.v1.RecognitionConfig.model

profanity_filter

Field google.cloud.speech.v1.RecognitionConfig.profanity_filter

sample_rate_hertz

Field google.cloud.speech.v1.RecognitionConfig.sample_rate_hertz

speech_contexts

Field google.cloud.speech.v1.RecognitionConfig.speech_contexts

use_enhanced

Field google.cloud.speech.v1.RecognitionConfig.use_enhanced

class google.cloud.speech_v1.types.RecognitionMetadata

Description of audio data to be recognized.

interaction_type

The use case most closely describing the audio content to be recognized.

industry_naics_code_of_audio

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/.

microphone_distance

The audio type that most closely describes the audio being recognized.

original_media_type

The original media the speech was recorded on.

recording_device_type

The type of device the speech was recorded with.

recording_device_name

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

original_mime_type

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

audio_topic

Description of the content. Eg. “Recordings of federal supreme court hearings from 2012”.

audio_topic

Field google.cloud.speech.v1.RecognitionMetadata.audio_topic

industry_naics_code_of_audio

Field google.cloud.speech.v1.RecognitionMetadata.industry_naics_code_of_audio

interaction_type

Field google.cloud.speech.v1.RecognitionMetadata.interaction_type

microphone_distance

Field google.cloud.speech.v1.RecognitionMetadata.microphone_distance

original_media_type

Field google.cloud.speech.v1.RecognitionMetadata.original_media_type

original_mime_type

Field google.cloud.speech.v1.RecognitionMetadata.original_mime_type

recording_device_name

Field google.cloud.speech.v1.RecognitionMetadata.recording_device_name

recording_device_type

Field google.cloud.speech.v1.RecognitionMetadata.recording_device_type

class google.cloud.speech_v1.types.RecognizeRequest

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

config

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

audio

Required. The audio data to be recognized.

audio

Field google.cloud.speech.v1.RecognizeRequest.audio

config

Field google.cloud.speech.v1.RecognizeRequest.config

class google.cloud.speech_v1.types.RecognizeResponse

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

results

Sequential list of transcription results corresponding to sequential portions of audio.

results

Field google.cloud.speech.v1.RecognizeResponse.results

class google.cloud.speech_v1.types.SpeakerDiarizationConfig

Config to enable speaker diarization.

enable_speaker_diarization

If ‘true’, enables speaker detection for each recognized word in the top alternative of the recognition result using a speaker_tag provided in the WordInfo.

min_speaker_count

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.

max_speaker_count

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.

speaker_tag

Unused.

enable_speaker_diarization

Field google.cloud.speech.v1.SpeakerDiarizationConfig.enable_speaker_diarization

max_speaker_count

Field google.cloud.speech.v1.SpeakerDiarizationConfig.max_speaker_count

min_speaker_count

Field google.cloud.speech.v1.SpeakerDiarizationConfig.min_speaker_count

speaker_tag

Field google.cloud.speech.v1.SpeakerDiarizationConfig.speaker_tag

class google.cloud.speech_v1.types.SpeechContext

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

phrases

A list of strings containing words and phrases “hints” so that the speech recognition is more likely to recognize them. This can be used to improve the accuracy for specific words and phrases, for example, if specific commands are typically spoken by the user. This can also be used to add additional words to the vocabulary of the recognizer. See usage limits. 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.

phrases

Field google.cloud.speech.v1.SpeechContext.phrases

class google.cloud.speech_v1.types.SpeechRecognitionAlternative

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

transcript

Transcript text representing the words that the user spoke.

confidence

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.

words

A list of word-specific information for each recognized word. Note: When enable_speaker_diarization is true, you will see all the words from the beginning of the audio.

confidence

Field google.cloud.speech.v1.SpeechRecognitionAlternative.confidence

transcript

Field google.cloud.speech.v1.SpeechRecognitionAlternative.transcript

words

Field google.cloud.speech.v1.SpeechRecognitionAlternative.words

class google.cloud.speech_v1.types.SpeechRecognitionResult

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

alternatives

May contain one or more recognition hypotheses (up to the maximum specified in max_alternatives). These alternatives are ordered in terms of accuracy, with the top (first) alternative being the most probable, as ranked by the recognizer.

channel_tag

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

Field google.cloud.speech.v1.SpeechRecognitionResult.alternatives

channel_tag

Field google.cloud.speech.v1.SpeechRecognitionResult.channel_tag

class google.cloud.speech_v1.types.Status
code

Field google.rpc.Status.code

details

Field google.rpc.Status.details

message

Field google.rpc.Status.message

class google.cloud.speech_v1.types.StreamingRecognitionConfig

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

config

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

single_utterance

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 StreamingRecognitionResults 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.

interim_results

If true, interim results (tentative hypotheses) may be returned as they become available (these interim results are indicated with the is_final=false flag). If false or omitted, only is_final=true result(s) are returned.

config

Field google.cloud.speech.v1.StreamingRecognitionConfig.config

interim_results

Field google.cloud.speech.v1.StreamingRecognitionConfig.interim_results

single_utterance

Field google.cloud.speech.v1.StreamingRecognitionConfig.single_utterance

class google.cloud.speech_v1.types.StreamingRecognitionResult

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

alternatives

May contain one or more recognition hypotheses (up to the maximum specified in max_alternatives). These alternatives are ordered in terms of accuracy, with the top (first) alternative being the most probable, as ranked by the recognizer.

is_final

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.

stability

An estimate of the likelihood that the recognizer will not change its guess about this interim result. Values range from 0.0 (completely unstable) to 1.0 (completely stable). This field is only provided for interim results (is_final=false). The default of 0.0 is a sentinel value indicating stability was not set.

result_end_time

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

channel_tag

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’.

language_code

The BCP-47 language tag of the language in this result. This language code was detected to have the most likelihood of being spoken in the audio.

alternatives

Field google.cloud.speech.v1.StreamingRecognitionResult.alternatives

channel_tag

Field google.cloud.speech.v1.StreamingRecognitionResult.channel_tag

is_final

Field google.cloud.speech.v1.StreamingRecognitionResult.is_final

language_code

Field google.cloud.speech.v1.StreamingRecognitionResult.language_code

result_end_time

Field google.cloud.speech.v1.StreamingRecognitionResult.result_end_time

stability

Field google.cloud.speech.v1.StreamingRecognitionResult.stability

class google.cloud.speech_v1.types.StreamingRecognizeRequest

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_content. All subsequent messages must contain audio_content and must not contain a streaming_config message.

streaming_request

The streaming request, which is either a streaming config or audio content.

streaming_config

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

audio_content

The audio data to be recognized. Sequential chunks of audio data are sent in sequential StreamingRecognizeRequest messages. The first StreamingRecognizeRequest message must not contain audio_content data and all subsequent StreamingRecognizeRequest messages must contain audio_content data. The audio bytes must be encoded as specified in RecognitionConfig. Note: as with all bytes fields, proto buffers use a pure binary representation (not base64). See content limits.

audio_content

Field google.cloud.speech.v1.StreamingRecognizeRequest.audio_content

streaming_config

Field google.cloud.speech.v1.StreamingRecognizeRequest.streaming_config

class google.cloud.speech_v1.types.StreamingRecognizeResponse

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 StreamingRecognizeResponses 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.

error

If set, returns a [google.rpc.Status][google.rpc.Status] message that specifies the error for the operation.

results

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).

speech_event_type

Indicates the type of speech event.

error

Field google.cloud.speech.v1.StreamingRecognizeResponse.error

results

Field google.cloud.speech.v1.StreamingRecognizeResponse.results

speech_event_type

Field google.cloud.speech.v1.StreamingRecognizeResponse.speech_event_type

class google.cloud.speech_v1.types.Timestamp
nanos

Field google.protobuf.Timestamp.nanos

seconds

Field google.protobuf.Timestamp.seconds

class google.cloud.speech_v1.types.WordInfo

Word-specific information for recognized words.

start_time

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.

end_time

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

The word corresponding to this set of information.

speaker_tag

A distinct integer value is assigned for every speaker within the audio. This field specifies which one of those speakers was detected to have spoken this word. Value ranges from ‘1’ to diarization_speaker_count. speaker_tag is set if enable_speaker_diarization = ‘true’ and only in the top alternative.

end_time

Field google.cloud.speech.v1.WordInfo.end_time

speaker_tag

Field google.cloud.speech.v1.WordInfo.speaker_tag

start_time

Field google.cloud.speech.v1.WordInfo.start_time

word

Field google.cloud.speech.v1.WordInfo.word