RecognitionConfig

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

JSON representation
{
  "encoding": enum(AudioEncoding),
  "sampleRateHertz": number,
  "audioChannelCount": number,
  "enableSeparateRecognitionPerChannel": boolean,
  "languageCode": string,
  "maxAlternatives": number,
  "profanityFilter": boolean,
  "speechContexts": [
    {
      object(SpeechContext)
    }
  ],
  "enableWordTimeOffsets": boolean,
  "model": string,
  "useEnhanced": boolean
}
Fields
encoding

enum(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.

sampleRateHertz

number

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, WAV. and 'MP3' audio files, and is required for all other audio formats. For details, see AudioEncoding.

audioChannelCount

number

Optional 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 enableSeparateRecognitionPerChannel to 'true'.

enableSeparateRecognitionPerChannel

boolean

This needs to be set to true explicitly and audioChannelCount > 1 to get each channel recognized separately. The recognition result will contain a channelTag 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: audioChannelCount multiplied by the length of the audio.

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 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.

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[]

object(SpeechContext)

Optional array of SpeechContext. A means to provide context to assist the speech recognition. For more information, see Phrase Hints.

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.

model

string

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

command_and_search

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

phone_call

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

video

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.

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.

useEnhanced

boolean

Optional Set to true to use an enhanced model for speech recognition. If useEnhanced is set to true and the model field is not set, then an appropriate enhanced model is chosen if: 1. project is eligible for requesting enhanced models 2. an enhanced model exists for the audio

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.

Enhanced speech models require that you opt-in to data logging using instructions in the documentation. If you set useEnhanced to true and you have not enabled audio logging, then you will receive an error.

AudioEncoding

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, 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.
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.

SpeechContext

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

JSON representation
{
  "phrases": [
    string
  ]
}
Fields
phrases[]

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.

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