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

JSON representation
  "encoding": enum(AudioEncoding),
  "sampleRateHertz": number,
  "languageCode": string,
  "maxAlternatives": number,
  "profanityFilter": boolean,
  "speechContexts": [
  "enableWordTimeOffsets": boolean,
  "enableAutomaticPunctuation": boolean,
  "metadata": {
  "model": string,
  "useEnhanced": boolean


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 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 and required for all other audio formats. For details, see AudioEncoding.



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.



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 maxAlternatives. 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.



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.



Optional A means to provide context to assist the speech recognition.



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.



Optional 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 Metadata regarding this request.



Optional 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


Best for short queries such as voice commands or voice search.


Best for audio that originated from a phone call (typically recorded at an 8khz sampling rate).


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.


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.



Optional Set to true to use an enhanced model for speech recognition. You must also set the model field to a valid, enhanced model. If useEnhanced is set to true and the model field is not set, then useEnhanced is ignored. If useEnhanced 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.

You must opt-in to the audio logging using the instructions in the data logging documentation. If you set useEnhanced to true and you have not enabled audio logging, then you will receive an error.


The encoding of the audio data sent in the request.

All encodings support only 1 channel (mono) audio.

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, and SPEEX_WITH_HEADER_BYTE.

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.

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. sampleRateHertz must be 8000.
AMR_WB Adaptive Multi-Rate Wideband codec. sampleRateHertz must be 16000.
OGG_OPUS Opus encoded audio frames in Ogg container (OggOpus). sampleRateHertz 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. sampleRateHertz must be 16000.


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

JSON representation
  "phrases": [


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.


Description of audio data to be recognized.

JSON representation
  "interactionType": enum(InteractionType),
  "industryNaicsCodeOfAudio": number,
  "microphoneDistance": enum(MicrophoneDistance),
  "originalMediaType": enum(OriginalMediaType),
  "recordingDeviceType": enum(RecordingDeviceType),
  "recordingDeviceName": string,
  "originalMimeType": string,
  "obfuscatedId": string,
  "audioTopic": string


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


number (uint32 format)

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



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



The original media the speech was recorded on.



The type of device the speech was recorded with.



The device used to make the recording. Examples 'Nexus 5X' or 'Polycom SoundStation IP 6000' or 'POTS' or 'VoIP' or 'Cardioid Microphone'.



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


string (int64 format)

Obfuscated (privacy-protected) ID of the user, to identify number of unique users using the service.



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.

INTERACTION_TYPE_UNSPECIFIED Use case is either unknown or is something other than one of the other values below.
DISCUSSION 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 One or more persons lecturing or presenting to others, mostly uninterrupted.
PHONE_CALL A phone-call or video-conference in which two or more people, who are not in the same room, are actively participating.
VOICEMAIL A recorded message intended for another person to listen to.
PROFESSIONALLY_PRODUCED Professionally produced audio (eg. TV Show, Podcast).
VOICE_COMMAND Transcribe voice commands, such as for controlling a device.
DICTATION 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.

NEARFIELD 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 The speaker if within 3 meters of the microphone.
FARFIELD The speaker is more than 3 meters away from the microphone.


The original media the speech was recorded on.

ORIGINAL_MEDIA_TYPE_UNSPECIFIED Unknown original media type.
AUDIO The speech data is an audio recording.
VIDEO The speech data originally recorded on a video.


The type of device the speech was recorded with.

RECORDING_DEVICE_TYPE_UNSPECIFIED The recording device is unknown.
SMARTPHONE Speech was recorded on a smartphone.
PC Speech was recorded using a personal computer or tablet.
PHONE_LINE Speech was recorded over a phone line.
VEHICLE Speech was recorded in a vehicle.
OTHER_OUTDOOR_DEVICE Speech was recorded outdoors.
OTHER_INDOOR_DEVICE Speech was recorded indoors.
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Cloud Speech-to-Text API