Class RecognitionConfig (2.23.0)

RecognitionConfig(mapping=None, *, ignore_unknown_fields=False, **kwargs)

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

Attributes

Name Description

encoding google.cloud.speech_v1p1beta1.types.RecognitionConfig.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.

sample_rate_hertz int
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.

audio_channel_count int
The number of channels in the input audio data. ONLY set this for MULTI-CHANNEL recognition. Valid values for LINEAR16, OGG_OPUS and FLAC are 1-8. 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'.

enable_separate_recognition_per_channel bool
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.

language_code str
Required. The language of the supplied audio as a BCP-47 __ language tag. Example: "en-US". See `Language Support

alternative_language_codes MutableSequence[str]
A list of up to 3 additional BCP-47 __ language tags, listing possible alternative languages of the supplied audio. See `Language Support

max_alternatives int
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.

profanity_filter bool
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.

adaptation google.cloud.speech_v1p1beta1.types.SpeechAdaptation
Speech adaptation configuration improves the accuracy of speech recognition. For more information, see the `speech adaptation

transcript_normalization google.cloud.speech_v1p1beta1.types.TranscriptNormalization
Use transcription normalization to automatically replace parts of the transcript with phrases of your choosing. For StreamingRecognize, this normalization only applies to stable partial transcripts (stability > 0.8) and final transcripts.

speech_contexts MutableSequence[google.cloud.speech_v1p1beta1.types.SpeechContext]
Array of SpeechContext. A means to provide context to assist the speech recognition. For more information, see `speech adaptation

enable_word_time_offsets bool
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.

enable_word_confidence bool
If true, the top result includes a list of words and the confidence for those words. If false, no word-level confidence information is returned. The default is false.

enable_automatic_punctuation bool
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.

enable_spoken_punctuation google.protobuf.wrappers_pb2.BoolValue
The spoken punctuation behavior for the call If not set, uses default behavior based on model of choice e.g. command_and_search will enable spoken punctuation by default If 'true', replaces spoken punctuation with the corresponding symbols in the request. For example, "how are you question mark" becomes "how are you?". See https://cloud.google.com/speech-to-text/docs/spoken-punctuation for support. If 'false', spoken punctuation is not replaced.

enable_spoken_emojis google.protobuf.wrappers_pb2.BoolValue
The spoken emoji behavior for the call If not set, uses default behavior based on model of choice If 'true', adds spoken emoji formatting for the request. This will replace spoken emojis with the corresponding Unicode symbols in the final transcript. If 'false', spoken emojis are not replaced.

enable_speaker_diarization bool
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. Note: Use diarization_config instead.

diarization_speaker_count int
If set, specifies the estimated number of speakers in the conversation. Defaults to '2'. Ignored unless enable_speaker_diarization is set to true. Note: Use diarization_config instead.

diarization_config google.cloud.speech_v1p1beta1.types.SpeakerDiarizationConfig
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.

metadata google.cloud.speech_v1p1beta1.types.RecognitionMetadata
Metadata regarding this request.

model

str
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. .. raw:: html

Model	Description
`latest_long`	Best for long form content like media or conversation.
`latest_short`	Best for short form content like commands or single shot directed speech.
`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 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.
`medical_conversation`	Best for audio that originated from a conversation between a medical provider and patient.
`medical_dictation`	Best for audio that originated from dictation notes by a medical provider.

use_enhanced bool
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.

Classes

AudioEncoding

AudioEncoding(value)

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, MP3, and WEBM_OPUS.

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.

Values: 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

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. MP3 (8): MP3 audio. MP3 encoding is a Beta feature and only available in v1p1beta1. Support all standard MP3 bitrates (which range from 32-320 kbps). When using this encoding, sample_rate_hertz has to match the sample rate of the file being used. WEBM_OPUS (9): Opus encoded audio frames in WebM container (OggOpus <https://wiki.xiph.org/OggOpus>__). sample_rate_hertz must be one of 8000, 12000, 16000, 24000, or 48000.