package google.cloud.speech.v1
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service Speech
cloud_speech.proto:34Service that implements Google Cloud Speech API.
Performs asynchronous speech recognition: receive results via the google.longrunning.Operations interface. Returns either an `Operation.error` or an `Operation.response` which contains a `LongRunningRecognizeResponse` message.
message
cloud_speech.proto:66The top-level message sent by the client for the `LongRunningRecognize` method.
optional config = 1
*Required* Provides information to the recognizer that specifies how to process the request.
optional audio = 2
*Required* The audio data to be recognized.
rpc Recognize (, )
cloud_speech.proto:37Performs synchronous speech recognition: receive results after all audio has been sent and processed.
message
cloud_speech.proto:55The top-level message sent by the client for the `Recognize` method.
optional config = 1
*Required* Provides information to the recognizer that specifies how to process the request.
optional audio = 2
*Required* The audio data to be recognized.
message
cloud_speech.proto:264The only message returned to the client by the `Recognize` method. It contains the result as zero or more sequential `SpeechRecognitionResult` messages.
repeated results = 2
*Output-only* Sequential list of transcription results corresponding to sequential portions of audio.
rpc StreamingRecognize (stream , stream )
cloud_speech.proto:51Performs bidirectional streaming speech recognition: receive results while sending audio. This method is only available via the gRPC API (not REST).
message
cloud_speech.proto:80The 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.
oneof streaming_request
The streaming request, which is either a streaming config or audio content.
streaming_config = 1
Provides information to the recognizer that specifies how to process the request. The first `StreamingRecognizeRequest` message must contain a `streaming_config` message.
bytes audio_content = 2
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](https://cloud.google.com/speech/limits#content).
message
cloud_speech.proto:345`StreamingRecognizeResponse` is the only message returned to the client by `StreamingRecognize`. A series of zero or more `StreamingRecognizeResponse` messages are streamed back to the client. If there is no recognizable audio, and `single_utterance` is set to false, then no messages are streamed back to the client. Here's an example of a series of ten `StreamingRecognizeResponse`s that might be returned while processing audio: 1. results { alternatives { transcript: "tube" } stability: 0.01 } 2. results { alternatives { transcript: "to be a" } stability: 0.01 } 3. results { alternatives { transcript: "to be" } stability: 0.9 } results { alternatives { transcript: " or not to be" } stability: 0.01 } 4. results { alternatives { transcript: "to be or not to be" confidence: 0.92 } alternatives { transcript: "to bee or not to bee" } is_final: true } 5. results { alternatives { transcript: " that's" } stability: 0.01 } 6. results { alternatives { transcript: " that is" } stability: 0.9 } results { alternatives { transcript: " the question" } stability: 0.01 } 7. results { alternatives { transcript: " that is the question" confidence: 0.98 } alternatives { transcript: " that was the question" } is_final: true } Notes: - Only two of the above responses #4 and #7 contain final results; they are indicated by `is_final: true`. Concatenating these together generates the full transcript: "to be or not to be that is the question". - The others contain interim `results`. #3 and #6 contain two interim `results`: the first portion has a high stability and is less likely to change; the second portion has a low stability and is very likely to change. A UI designer might choose to show only high stability `results`. - The specific `stability` and `confidence` values shown above are only for illustrative purposes. Actual values may vary. - In each response, only one of these fields will be set: `error`, `speech_event_type`, or one or more (repeated) `results`.
optional error = 1
*Output-only* If set, returns a [google.rpc.Status][google.rpc.Status] message that specifies the error for the operation.
repeated results = 2
*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).
StreamingRecognizeResponse.SpeechEventType speech_event_type = 4
*Output-only* Indicates the type of speech event.
message
cloud_speech.proto:284Describes 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.
int32 progress_percent = 1
Approximate percentage of audio processed thus far. Guaranteed to be 100 when the audio is fully processed and the results are available.
optional start_time = 2
Time when the request was received.
optional last_update_time = 3
Time of the most recent processing update.
message
cloud_speech.proto:275The 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.
repeated results = 2
*Output-only* Sequential list of transcription results corresponding to sequential portions of audio.
message
cloud_speech.proto:242Contains 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 [audio limits](https://cloud.google.com/speech/limits#content).
Used in: ,
oneof audio_source
The audio source, which is either inline content or a Google Cloud Storage uri.
bytes content = 1
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.
string uri = 2
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][google.rpc.Code.INVALID_ARGUMENT]). For more information, see [Request URIs](https://cloud.google.com/storage/docs/reference-uris).
message
cloud_speech.proto:129Provides information to the recognizer that specifies how to process the request.
Used in: , , , , ,
RecognitionConfig.AudioEncoding encoding = 1
*Required* Encoding of audio data sent in all `RecognitionAudio` messages.
int32 sample_rate_hertz = 2
*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).
string language_code = 3
*Required* The language of the supplied audio as a [BCP-47](https://www.rfc-editor.org/rfc/bcp/bcp47.txt) language tag. Example: "en-US". See [Language Support](https://cloud.google.com/speech/docs/languages) for a list of the currently supported language codes.
int32 max_alternatives = 4
*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.
bool profanity_filter = 5
*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.
repeated speech_contexts = 6
*Optional* A means to provide context to assist the speech recognition.
bool enable_word_time_offsets = 8
*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`.
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.
Used in:
ENCODING_UNSPECIFIED = 0
Not specified. Will return result [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT].
LINEAR16 = 1
Uncompressed 16-bit signed little-endian samples (Linear PCM).
FLAC = 2
[`FLAC`](https://xiph.org/flac/documentation.html) (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 = 3
8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
AMR = 4
Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
AMR_WB = 5
Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000.
OGG_OPUS = 6
Opus encoded audio frames in Ogg container ([OggOpus](https://wiki.xiph.org/OggOpus)). `sample_rate_hertz` must be 16000.
SPEEX_WITH_HEADER_BYTE = 7
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](https://speex.org/) 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](https://tools.ietf.org/html/rfc5574). 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.
message
cloud_speech.proto:228Provides "hints" to the speech recognizer to favor specific words and phrases in the results.
Used in:
repeated string phrases = 1
*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](https://cloud.google.com/speech/limits#content).
message
cloud_speech.proto:407Alternative hypotheses (a.k.a. n-best list).
Used in: ,
string transcript = 1
*Output-only* Transcript text representing the words that the user spoke.
float confidence = 2
*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.
repeated words = 3
*Output-only* A list of word-specific information for each recognized word.
message
cloud_speech.proto:398A speech recognition result corresponding to a portion of the audio.
Used in: ,
repeated alternatives = 1
*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.
message
cloud_speech.proto:102Provides information to the recognizer that specifies how to process the request.
Used in:
optional config = 1
*Required* Provides information to the recognizer that specifies how to process the request.
bool single_utterance = 2
*Optional* If `false` or omitted, the recognizer will perform continuous recognition (continuing to wait for and process audio even if the user pauses speaking) until the client closes the input stream (gRPC API) or until the maximum time limit has been reached. May return multiple `StreamingRecognitionResult`s with the `is_final` flag set to `true`. If `true`, the recognizer will detect a single spoken utterance. When it detects that the user has paused or stopped speaking, it will return an `END_OF_SINGLE_UTTERANCE` event and cease recognition. It will return no more than one `StreamingRecognitionResult` with the `is_final` flag set to `true`.
bool interim_results = 3
*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.
message
cloud_speech.proto:377A streaming speech recognition result corresponding to a portion of the audio that is currently being processed.
Used in:
repeated alternatives = 1
*Output-only* May contain one or more recognition hypotheses (up to the maximum specified in `max_alternatives`).
bool is_final = 2
*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.
float stability = 3
*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.
Indicates the type of speech event.
Used in:
SPEECH_EVENT_UNSPECIFIED = 0
No speech event specified.
END_OF_SINGLE_UTTERANCE = 1
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.
message
cloud_speech.proto:426Word-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`.
Used in:
optional start_time = 1
*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.
optional end_time = 2
*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.
string word = 3
*Output-only* The word corresponding to this set of information.