As of January 1, 2020 this library no longer supports Python 2 on the latest released version. Library versions released prior to that date will continue to be available. For more information please visit Python 2 support on Google Cloud.

Types for Google Cloud Speech v1 API

class google.cloud.speech_v1.types.LongRunningRecognizeMetadata(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]

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.

Type

int

start_time

Time when the request was received.

Type

Timestamp

last_update_time

Time of the most recent processing update.

Type

Timestamp

class google.cloud.speech_v1.types.LongRunningRecognizeRequest(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]

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.

Type

RecognitionConfig

audio

Required. The audio data to be recognized.

Type

RecognitionAudio

class google.cloud.speech_v1.types.LongRunningRecognizeResponse(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]

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.

Type

Sequence[SpeechRecognitionResult]

class google.cloud.speech_v1.types.RecognitionAudio(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]

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.

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.

Type

bytes

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.

Type

str

class google.cloud.speech_v1.types.RecognitionConfig(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]

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.RecognitionConfig.AudioEncoding].

Type

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.RecognitionConfig.AudioEncoding].

Type

int

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’.

Type

int

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.

Type

bool

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.

Type

str

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.

Type

int

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.

Type

bool

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.

Type

Sequence[SpeechContext]

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.

Type

bool

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.

Type

bool

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.

Type

SpeakerDiarizationConfig

metadata

Metadata regarding this request.

Type

RecognitionMetadata

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.

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.
Type

str

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.

Type

bool

class AudioEncoding(value)[source]

The encoding of the audio data sent in the request.

All encodings support only 1 channel (mono) audio, unless the audio_channel_count and enable_separate_recognition_per_channel fields are set.

For best results, the audio source should be captured and transmitted using a lossless encoding (FLAC or LINEAR16). The accuracy of the speech recognition can be reduced if lossy codecs are used to capture or transmit audio, particularly if background noise is present. Lossy codecs include MULAW, AMR, AMR_WB, OGG_OPUS, SPEEX_WITH_HEADER_BYTE, and MP3.

The FLAC and WAV audio file formats include a header that describes the included audio content. You can request recognition for WAV files that contain either LINEAR16 or MULAW encoded audio. If you send FLAC or WAV audio file format in your request, you do not need to specify an AudioEncoding; the audio encoding format is determined from the file header. If you specify an AudioEncoding when you send send FLAC or WAV audio, the encoding configuration must match the encoding described in the audio header; otherwise the request returns an [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT] error code.

class google.cloud.speech_v1.types.RecognitionMetadata(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]

Description of audio data to be recognized.

interaction_type

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

Type

InteractionType

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/.

Type

int

microphone_distance

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

Type

MicrophoneDistance

original_media_type

The original media the speech was recorded on.

Type

OriginalMediaType

recording_device_type

The type of device the speech was recorded with.

Type

RecordingDeviceType

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’.

Type

str

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

Type

str

audio_topic

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

Type

str

class InteractionType(value)[source]

Use case categories that the audio recognition request can be described by.

class MicrophoneDistance(value)[source]

Enumerates the types of capture settings describing an audio file.

class OriginalMediaType(value)[source]

The original media the speech was recorded on.

class RecordingDeviceType(value)[source]

The type of device the speech was recorded with.

class google.cloud.speech_v1.types.RecognizeRequest(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]

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.

Type

RecognitionConfig

audio

Required. The audio data to be recognized.

Type

RecognitionAudio

class google.cloud.speech_v1.types.RecognizeResponse(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]

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.

Type

Sequence[SpeechRecognitionResult]

class google.cloud.speech_v1.types.SpeakerDiarizationConfig(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]

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.

Type

bool

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.

Type

int

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.

Type

int

speaker_tag

Unused.

Type

int

class google.cloud.speech_v1.types.SpeechContext(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]

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.

Type

Sequence[str]

class google.cloud.speech_v1.types.SpeechRecognitionAlternative(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]

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

transcript

Transcript text representing the words that the user spoke.

Type

str

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.

Type

float

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.

Type

Sequence[WordInfo]

class google.cloud.speech_v1.types.SpeechRecognitionResult(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]

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.

Type

Sequence[SpeechRecognitionAlternative]

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’.

Type

int

class google.cloud.speech_v1.types.StreamingRecognitionConfig(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]

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.

Type

RecognitionConfig

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.

Type

bool

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.

Type

bool

class google.cloud.speech_v1.types.StreamingRecognitionResult(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]

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.

Type

Sequence[SpeechRecognitionAlternative]

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.

Type

bool

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.

Type

float

result_end_time

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

Type

Duration

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’.

Type

int

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.

Type

str

class google.cloud.speech_v1.types.StreamingRecognizeRequest(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]

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_config

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

Type

StreamingRecognitionConfig

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.

Type

bytes

class google.cloud.speech_v1.types.StreamingRecognizeResponse(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]

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.

Type

Status

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).

Type

Sequence[StreamingRecognitionResult]

speech_event_type

Indicates the type of speech event.

Type

SpeechEventType

class SpeechEventType(value)[source]

Indicates the type of speech event.

class google.cloud.speech_v1.types.WordInfo(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]

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.

Type

Duration

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.

Type

Duration

word

The word corresponding to this set of information.

Type

str

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.

Type

int