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.
|