@google-cloud/speech 1.4.0 » Namespace: v1

google.cloud.speech.v1

namespace   static

Enumerations

AudioEncoding

static

number

Audio encoding of the data sent in the audio message. All encodings support only 1 channel (mono) audio. Only FLAC and WAV include a header that describes the bytes of audio that follow the header. The other encodings are raw audio bytes with no header.

For best results, the audio source should be captured and transmitted using a lossless encoding (FLAC or LINEAR16). Recognition accuracy may be reduced if lossy codecs, which include the other codecs listed in this section, are used to capture or transmit the audio, particularly if background noise is present.

Value
ENCODING_UNSPECIFIED	Not specified. Will return result google.rpc.Code.INVALID_ARGUMENT.
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.
AMR	Adaptive Multi-Rate Narrowband codec. sample_rate_hertz must be 8000.
AMR_WB	Adaptive Multi-Rate Wideband codec. sample_rate_hertz must be 16000.
OGG_OPUS	Opus encoded audio frames in Ogg container ( OggOpus). sample_rate_hertz must be 16000.
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.

SpeechEventType

static

number

Indicates the type of speech event.

Value
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.

Properties

AudioEncoding

static

number

Audio encoding of the data sent in the audio message. All encodings support only 1 channel (mono) audio. Only FLAC and WAV include a header that describes the bytes of audio that follow the header. The other encodings are raw audio bytes with no header.

For best results, the audio source should be captured and transmitted using a lossless encoding (FLAC or LINEAR16). Recognition accuracy may be reduced if lossy codecs, which include the other codecs listed in this section, are used to capture or transmit the audio, particularly if background noise is present.

Value
ENCODING_UNSPECIFIED	Not specified. Will return result google.rpc.Code.INVALID_ARGUMENT.
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.
AMR	Adaptive Multi-Rate Narrowband codec. sample_rate_hertz must be 8000.
AMR_WB	Adaptive Multi-Rate Wideband codec. sample_rate_hertz must be 16000.
OGG_OPUS	Opus encoded audio frames in Ogg container ( OggOpus). sample_rate_hertz must be 16000.
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.

SpeechEventType

static

number

Indicates the type of speech event.

Value
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.

Abstract types

LongRunningRecognizeMetadata

static

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.

Properties

Parameter
progressPercent	number Approximate percentage of audio processed thus far. Guaranteed to be 100 when the audio is fully processed and the results are available.
startTime	Object Time when the request was received. This object should have the same structure as Timestamp
lastUpdateTime	Object Time of the most recent processing update. This object should have the same structure as Timestamp

See also

google.cloud.speech.v1.LongRunningRecognizeMetadata definition in proto format

LongRunningRecognizeRequest

static

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

Properties

Parameter
config	Object Required Provides information to the recognizer that specifies how to process the request. This object should have the same structure as RecognitionConfig
audio	Object Required The audio data to be recognized. This object should have the same structure as RecognitionAudio

See also

google.cloud.speech.v1.LongRunningRecognizeRequest definition in proto format

LongRunningRecognizeResponse

static

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.

Property

Parameter
results	Array of Object Output-only Sequential list of transcription results corresponding to sequential portions of audio. This object should have the same structure as SpeechRecognitionResult

See also

google.cloud.speech.v1.LongRunningRecognizeResponse definition in proto format

RecognitionAudio

static

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 audio limits.

Properties

Parameter
content	string 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.
uri	string URI that points to a file that contains audio data bytes as specified in RecognitionConfig. 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). For more information, see Request URIs.

See also

google.cloud.speech.v1.RecognitionAudio definition in proto format

RecognitionConfig

static

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

Properties

Parameter
encoding	number Required Encoding of audio data sent in all RecognitionAudio messages. The number should be among the values of AudioEncoding
sampleRateHertz	number Required 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).
languageCode	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.
maxAlternatives	number 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. 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.
profanityFilter	boolean 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.
speechContexts	Array of Object Optional A means to provide context to assist the speech recognition. This object should have the same structure as SpeechContext
enableWordTimeOffsets	boolean 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.

See also

google.cloud.speech.v1.RecognitionConfig definition in proto format

RecognizeRequest

static

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

Properties

Parameter
config	Object Required Provides information to the recognizer that specifies how to process the request. This object should have the same structure as RecognitionConfig
audio	Object Required The audio data to be recognized. This object should have the same structure as RecognitionAudio

See also

google.cloud.speech.v1.RecognizeRequest definition in proto format

RecognizeResponse

static

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

Property

Parameter
results	Array of Object Output-only Sequential list of transcription results corresponding to sequential portions of audio. This object should have the same structure as SpeechRecognitionResult

See also

google.cloud.speech.v1.RecognizeResponse definition in proto format

SpeechContext

static

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

Property

Parameter
phrases	Array of string 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.

See also

google.cloud.speech.v1.SpeechContext definition in proto format

SpeechRecognitionAlternative

static

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

Properties

Parameter
transcript	string Output-only Transcript text representing the words that the user spoke.
confidence	number 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 typically provided only for the top hypothesis, and only for is_final=true results. Clients should not rely on the confidence field as it is not guaranteed to be accurate or consistent. The default of 0.0 is a sentinel value indicating confidence was not set.
words	Array of Object Output-only A list of word-specific information for each recognized word. This object should have the same structure as WordInfo

See also

google.cloud.speech.v1.SpeechRecognitionAlternative definition in proto format

SpeechRecognitionResult

static

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

Property

Parameter
alternatives	Array of Object 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. This object should have the same structure as SpeechRecognitionAlternative

See also

google.cloud.speech.v1.SpeechRecognitionResult definition in proto format

StreamingRecognitionConfig

static

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

Properties

Parameter
config	Object Required Provides information to the recognizer that specifies how to process the request. This object should have the same structure as RecognitionConfig
singleUtterance	boolean 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 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.
interimResults	boolean Optional 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.

See also

google.cloud.speech.v1.StreamingRecognitionConfig definition in proto format

StreamingRecognitionResult

static

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

Properties

Parameter
alternatives	Array of Object Output-only May contain one or more recognition hypotheses (up to the maximum specified in max_alternatives). This object should have the same structure as SpeechRecognitionAlternative
isFinal	boolean 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.
stability	number Output-only 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.

See also

google.cloud.speech.v1.StreamingRecognitionResult definition in proto format

StreamingRecognizeRequest

static

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.

Properties

Parameter
streamingConfig	Object Provides information to the recognizer that specifies how to process the request. The first StreamingRecognizeRequest message must contain a streaming_config message. This object should have the same structure as StreamingRecognitionConfig
audioContent	string 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, protobuffers use a pure binary representation (not base64). See audio limits.

See also

google.cloud.speech.v1.StreamingRecognizeRequest definition in proto format

StreamingRecognizeResponse

static

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.

Properties

Parameter
error	Object Output-only If set, returns a google.rpc.Status message that specifies the error for the operation. This object should have the same structure as Status
results	Array of Object 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 more is_final=false results followed by zero or one is_final=true result (the newly settled portion). This object should have the same structure as StreamingRecognitionResult
speechEventType	number Output-only Indicates the type of speech event. The number should be among the values of SpeechEventType

See also

google.cloud.speech.v1.StreamingRecognizeResponse definition in proto format

WordInfo

static

Word-specific information for recognized words. Word information is only included in the response when certain request parameters are set, such as enable_word_time_offsets.

Properties

Parameter
startTime	Object 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. This object should have the same structure as Duration
endTime	Object 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. This object should have the same structure as Duration
word	string Output-only The word corresponding to this set of information.

See also

google.cloud.speech.v1.WordInfo definition in proto format