Class TranslateSpeechConfig (0.11.17) 

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

Provides information to the speech translation that specifies how to process the request.

Attributes
Name Description
audio_encoding str
Required. Encoding of audio data. Supported formats: - 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. - mulaw 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law. - amr Adaptive Multi-Rate Narrowband codec. sample_rate_hertz must be 8000. - amr-wb Adaptive Multi-Rate Wideband codec. sample_rate_hertz must be 16000. - ogg-opus Opus encoded audio frames in Ogg __ container. sample_rate_hertz must be one of 8000, 12000, 16000, 24000, or 48000. - mp3 MP3 audio. 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.
source_language_code str
Required. Source language code (BCP-47) of the input audio.
target_language_code str
Required. Target language code (BCP-47) of the output.
sample_rate_hertz int
Optional. Sample rate in Hertz of the audio data. 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).
model str
Optional. google-provided-model/video and google-provided-model/enhanced-phone-call are premium models. google-provided-model/phone-call is not premium model.