Package google.cloud.speech.v1p1beta1

Index

Speech (interface)
RecognitionAudio (message)
RecognitionConfig (message)
RecognitionConfig.AudioEncoding (enum)
RecognizeRequest (message)
RecognizeResponse (message)
SpeakerDiarizationConfig (message)
SpeechRecognitionAlternative (message)
SpeechRecognitionResult (message)
StreamingRecognitionConfig (message)
StreamingRecognitionResult (message)
StreamingRecognizeRequest (message)
StreamingRecognizeResponse (message)
StreamingRecognizeResponse.SpeechEventType (enum)
WordInfo (message)

Speech

Service that implements Google Cloud Speech API.

Recognize

Recognize
`rpc Recognize(RecognizeRequest) returns (RecognizeResponse)` Performs synchronous speech recognition: receive results after all audio has been sent and processed.

rpc Recognize(RecognizeRequest) returns (RecognizeResponse)

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

StreamingRecognize

StreamingRecognize
`rpc StreamingRecognize(StreamingRecognizeRequest) returns (StreamingRecognizeResponse)` Performs bidirectional streaming speech recognition: receive results while sending audio. This method is only available via the gRPC API (not REST).

rpc StreamingRecognize(StreamingRecognizeRequest) returns (StreamingRecognizeResponse)

Performs bidirectional streaming speech recognition: receive results while sending audio. This method is only available via the gRPC API (not REST).

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. See content limits.

Fields

Fields
Union field `audio_source`. The audio source, which is either inline content or a Google Cloud Storage uri. `audio_source` can be only one of the following:
`content`	`bytes` 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.

Union field audio_source. The audio source, which is either inline content or a Google Cloud Storage uri. audio_source can be only one of the following:

content

bytes

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.

RecognitionConfig

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

Fields

encoding

AudioEncoding

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.

sample_rate_hertz

int32

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.

audio_channel_count

int32

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

bool

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

string

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.

enable_automatic_punctuation

bool

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.

diarization_config

SpeakerDiarizationConfig

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.

model

string

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

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.

AudioEncoding

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 error code.

Enums
`ENCODING_UNSPECIFIED`	Not specified.
`LINEAR16`	Uncompressed 16-bit signed little-endian samples (Linear PCM).
`FLAC`	`FLAC` (Free Lossless Audio Codec) is the recommended encoding because it is lossless--therefore recognition is not compromised--and requires only about half the bandwidth of `LINEAR16`. `FLAC` stream encoding supports 16-bit and 24-bit samples, however, not all fields in `STREAMINFO` are supported.
`MULAW`	8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
`OGG_OPUS`	Opus encoded audio frames in Ogg container (OggOpus). `sample_rate_hertz` must be one of 8000, 12000, 16000, 24000, or 48000.
`SPEEX_WITH_HEADER_BYTE`	Although the use of lossy encodings is not recommended, if a very low bitrate encoding is required, `OGG_OPUS` is highly preferred over Speex encoding. The Speex encoding supported by Cloud Speech API has a header byte in each block, as in MIME type `audio/x-speex-with-header-byte`. It is a variant of the RTP Speex encoding defined in RFC 5574. The stream is a sequence of blocks, one block per RTP packet. Each block starts with a byte containing the length of the block, in bytes, followed by one or more frames of Speex data, padded to an integral number of bytes (octets) as specified in RFC 5574. In other words, each RTP header is replaced with a single byte containing the block length. Only Speex wideband is supported. `sample_rate_hertz` must be 16000.

RecognizeRequest

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

Fields

Fields
`config`	`RecognitionConfig` Required. Provides information to the recognizer that specifies how to process the request.
`audio`	`RecognitionAudio` Required. The audio data to be recognized.

config

RecognitionConfig

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

audio

RecognitionAudio

Required. The audio data to be recognized.

RecognizeResponse

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

Fields

Fields
`results[]`	`SpeechRecognitionResult` Sequential list of transcription results corresponding to sequential portions of audio.

results[]

SpeechRecognitionResult

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

SpeakerDiarizationConfig

Config to enable speaker diarization.

Fields

Fields
`enable_speaker_diarization`	`bool` 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`	`int32` 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`	`int32` 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.

enable_speaker_diarization

bool

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

int32

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

int32

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.

SpeechRecognitionAlternative

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

Fields

Fields
`transcript`	`string` Transcript text representing the words that the user spoke.
`confidence`	`float` 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[]`	`WordInfo` 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.

transcript

string

Transcript text representing the words that the user spoke.

confidence

float

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[]

WordInfo

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.

SpeechRecognitionResult

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

Fields

alternatives[]

SpeechRecognitionAlternative

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

int32

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

string

Output only. 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.

StreamingRecognitionConfig

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

Fields

Fields
`config`	`RecognitionConfig` Required. Provides information to the recognizer that specifies how to process the request.

config

RecognitionConfig

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

StreamingRecognitionResult

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

Fields
`alternatives[]`	`SpeechRecognitionAlternative` 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`	`bool` 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`	`float` 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`	`Duration` Time offset of the end of this result relative to the beginning of the audio.
`channel_tag`	`int32` 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`	`string` Output only. 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.

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.

Fields

Fields
Union field `streaming_request`. The streaming request, which is either a streaming config or audio content. `streaming_request` can be only one of the following:
`streaming_config`	`StreamingRecognitionConfig` Provides information to the recognizer that specifies how to process the request. The first `StreamingRecognizeRequest` message must contain a `streaming_config` message.
`audio_content`	`bytes` 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.

Union field streaming_request. The streaming request, which is either a streaming config or audio content. streaming_request can be only one of the following:

streaming_config

StreamingRecognitionConfig

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

audio_content

bytes

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.

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:

results { alternatives { transcript: "tube" } stability: 0.01 }
results { alternatives { transcript: "to be a" } stability: 0.01 }
results { alternatives { transcript: "to be" } stability: 0.9 } results { alternatives { transcript: " or not to be" } stability: 0.01 }
results { alternatives { transcript: "to be or not to be" confidence: 0.92 } alternatives { transcript: "to bee or not to bee" } is_final: true }
results { alternatives { transcript: " that's" } stability: 0.01 }
results { alternatives { transcript: " that is" } stability: 0.9 } results { alternatives { transcript: " the question" } stability: 0.01 }
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.

Fields

Fields
`error`	`Status` If set, returns a `google.rpc.Status` message that specifies the error for the operation.
`results[]`	`StreamingRecognitionResult` 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`	`SpeechEventType` Indicates the type of speech event.

error

Status

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

results[]

StreamingRecognitionResult

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

SpeechEventType

Indicates the type of speech event.

SpeechEventType

Indicates the type of speech event.

Enums

SPEECH_EVENT_UNSPECIFIED No speech event specified.

END_OF_SINGLE_UTTERANCE This event indicates that the server has detected the end of the user's speech utterance and expects no additional speech. Therefore, the server will not process additional audio (although it may subsequently return additional results). The client should stop sending additional audio data, half-close the gRPC connection, and wait for any additional results until the server closes the gRPC connection. This event is only sent if single_utterance was set to true, and is not used otherwise.

Enums
`SPEECH_EVENT_UNSPECIFIED`	No speech event specified.
`END_OF_SINGLE_UTTERANCE`	This event indicates that the server has detected the end of the user's speech utterance and expects no additional speech. Therefore, the server will not process additional audio (although it may subsequently return additional results). The client should stop sending additional audio data, half-close the gRPC connection, and wait for any additional results until the server closes the gRPC connection. This event is only sent if `single_utterance` was set to `true`, and is not used otherwise.

WordInfo

Word-specific information for recognized words.

Fields
`start_time`	`Duration` 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`	`Duration` 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`	`string` The word corresponding to this set of information.
`speaker_tag`	`int32` Output only. 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.