package google.cloud.speech.v1
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service Speech
cloud_speech.proto:37Service that implements Google Cloud Speech API.
rpc Recognize (, )
cloud_speech.proto:43Performs synchronous speech recognition: receive results after all audio has been sent and processed.
message
cloud_speech.proto:76The 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:543The only message returned to the client by the `Recognize` method. It contains the result as zero or more sequential `SpeechRecognitionResult` messages.
repeated results = 2
Sequential list of transcription results corresponding to sequential portions of audio.
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. For more information on asynchronous speech recognition, see the [how-to](https://cloud.google.com/speech-to-text/docs/async-recognize).
message
cloud_speech.proto:87The 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 StreamingRecognize (stream , stream )
cloud_speech.proto:71Performs bidirectional streaming speech recognition: receive results while sending audio. This method is only available via the gRPC API (not REST).
message
cloud_speech.proto:101The 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.
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, proto buffers use a pure binary representation (not base64). See [content limits](https://cloud.google.com/speech-to-text/quotas#content).
message
cloud_speech.proto:624`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
If set, returns a [google.rpc.Status][google.rpc.Status] message that specifies the error for the operation.
repeated results = 2
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).
StreamingRecognizeResponse.SpeechEventType speech_event_type = 4
Indicates the type of speech event.
message
cloud_speech.proto:563Describes 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:554The 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
Sequential list of transcription results corresponding to sequential portions of audio.
message
cloud_speech.proto:520Contains 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](https://cloud.google.com/speech-to-text/quotas#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, proto buffers 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`. 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](https://cloud.google.com/storage/docs/reference-uris).
message
cloud_speech.proto:150Provides information to the recognizer that specifies how to process the request.
Used in: , ,
RecognitionConfig.AudioEncoding encoding = 1
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].
int32 sample_rate_hertz = 2
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].
int32 audio_channel_count = 7
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'.
bool enable_separate_recognition_per_channel = 12
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.
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-to-text/docs/languages) for a list of the currently supported language codes.
int32 max_alternatives = 4
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
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
Array of [SpeechContext][google.cloud.speech.v1.SpeechContext]. A means to provide context to assist the speech recognition. For more information, see [speech adaptation](https://cloud.google.com/speech-to-text/docs/context-strength).
bool enable_word_time_offsets = 8
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`.
bool enable_automatic_punctuation = 11
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.
optional diarization_config = 19
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.
optional metadata = 9
Metadata regarding this request.
string model = 13
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. <table> <tr> <td><b>Model</b></td> <td><b>Description</b></td> </tr> <tr> <td><code>command_and_search</code></td> <td>Best for short queries such as voice commands or voice search.</td> </tr> <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> </tr> <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> </tr> <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> </tr> </table>
bool use_enhanced = 14
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.
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.
Used in:
ENCODING_UNSPECIFIED = 0
Not specified.
LINEAR16 = 1
Uncompressed 16-bit signed little-endian samples (Linear PCM).
FLAC = 2
`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 = 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 one of 8000, 12000, 16000, 24000, or 48000.
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:372Description of audio data to be recognized.
Used in:
RecognitionMetadata.InteractionType interaction_type = 1
The use case most closely describing the audio content to be recognized.
uint32 industry_naics_code_of_audio = 3
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/.
RecognitionMetadata.MicrophoneDistance microphone_distance = 4
The audio type that most closely describes the audio being recognized.
RecognitionMetadata.OriginalMediaType original_media_type = 5
The original media the speech was recorded on.
RecognitionMetadata.RecordingDeviceType recording_device_type = 6
The type of device the speech was recorded with.
string recording_device_name = 7
The device used to make the recording. Examples 'Nexus 5X' or 'Polycom SoundStation IP 6000' or 'POTS' or 'VoIP' or 'Cardioid Microphone'.
string original_mime_type = 8
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
string audio_topic = 10
Description of the content. Eg. "Recordings of federal supreme court hearings from 2012".
Use case categories that the audio recognition request can be described by.
Used in:
INTERACTION_TYPE_UNSPECIFIED = 0
Use case is either unknown or is something other than one of the other values below.
DISCUSSION = 1
Multiple people in a conversation or discussion. For example in a meeting with two or more people actively participating. Typically all the primary people speaking would be in the same room (if not, see PHONE_CALL)
PRESENTATION = 2
One or more persons lecturing or presenting to others, mostly uninterrupted.
PHONE_CALL = 3
A phone-call or video-conference in which two or more people, who are not in the same room, are actively participating.
VOICEMAIL = 4
A recorded message intended for another person to listen to.
PROFESSIONALLY_PRODUCED = 5
Professionally produced audio (eg. TV Show, Podcast).
VOICE_SEARCH = 6
Transcribe spoken questions and queries into text.
VOICE_COMMAND = 7
Transcribe voice commands, such as for controlling a device.
DICTATION = 8
Transcribe speech to text to create a written document, such as a text-message, email or report.
Enumerates the types of capture settings describing an audio file.
Used in:
MICROPHONE_DISTANCE_UNSPECIFIED = 0
Audio type is not known.
NEARFIELD = 1
The audio was captured from a closely placed microphone. Eg. phone, dictaphone, or handheld microphone. Generally if there speaker is within 1 meter of the microphone.
MIDFIELD = 2
The speaker if within 3 meters of the microphone.
FARFIELD = 3
The speaker is more than 3 meters away from the microphone.
The original media the speech was recorded on.
Used in:
ORIGINAL_MEDIA_TYPE_UNSPECIFIED = 0
Unknown original media type.
AUDIO = 1
The speech data is an audio recording.
VIDEO = 2
The speech data originally recorded on a video.
The type of device the speech was recorded with.
Used in:
RECORDING_DEVICE_TYPE_UNSPECIFIED = 0
The recording device is unknown.
SMARTPHONE = 1
Speech was recorded on a smartphone.
PC = 2
Speech was recorded using a personal computer or tablet.
PHONE_LINE = 3
Speech was recorded over a phone line.
VEHICLE = 4
Speech was recorded in a vehicle.
OTHER_OUTDOOR_DEVICE = 5
Speech was recorded outdoors.
OTHER_INDOOR_DEVICE = 6
Speech was recorded indoors.
message
cloud_speech.proto:354Config to enable speaker diarization.
Used in:
bool enable_speaker_diarization = 1
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.
int32 min_speaker_count = 2
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.
int32 max_speaker_count = 3
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.
message
cloud_speech.proto:500Provides "hints" to the speech recognizer to favor specific words and phrases in the results.
Used in:
repeated string phrases = 1
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-to-text/quotas#content). 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.
message
cloud_speech.proto:708Alternative hypotheses (a.k.a. n-best list).
Used in: ,
string transcript = 1
Transcript text representing the words that the user spoke.
float confidence = 2
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.
repeated words = 3
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.
message
cloud_speech.proto:694A speech recognition result corresponding to a portion of the audio.
Used in: ,
repeated alternatives = 1
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.
int32 channel_tag = 2
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'.
message
cloud_speech.proto:123Provides 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
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
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:656A streaming speech recognition result corresponding to a portion of the audio that is currently being processed.
Used in:
repeated alternatives = 1
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.
bool is_final = 2
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
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.
optional result_end_time = 4
Time offset of the end of this result relative to the beginning of the audio.
int32 channel_tag = 5
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'.
string language_code = 6
The [BCP-47](https://www.rfc-editor.org/rfc/bcp/bcp47.txt) language tag of the language in this result. This language code was detected to have the most likelihood of being spoken in the audio.
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:728Word-specific information for recognized words.
Used in:
optional start_time = 1
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
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
The word corresponding to this set of information.
int32 speaker_tag = 5
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.