Reference documentation and code samples for the Google Cloud Media Translation V1beta1 Client class TranslateSpeechConfig.
Provides information to the speech translation that specifies how to process the request.
Generated from protobuf message google.cloud.mediatranslation.v1beta1.TranslateSpeechConfig
Methods
__construct
Constructor.
Parameters | |
---|---|
Name | Description |
data |
array
Optional. Data for populating the Message object. |
↳ audio_encoding |
string
Required. Encoding of audio data. Supported formats: - |
↳ source_language_code |
string
Required. Source language code (BCP-47) of the input audio. |
↳ target_language_code |
string
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 |
string
Optional. |
getAudioEncoding
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 oflinear16
.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.
Returns | |
---|---|
Type | Description |
string |
setAudioEncoding
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 oflinear16
.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.
Parameter | |
---|---|
Name | Description |
var |
string
|
Returns | |
---|---|
Type | Description |
$this |
getSourceLanguageCode
Required. Source language code (BCP-47) of the input audio.
Returns | |
---|---|
Type | Description |
string |
setSourceLanguageCode
Required. Source language code (BCP-47) of the input audio.
Parameter | |
---|---|
Name | Description |
var |
string
|
Returns | |
---|---|
Type | Description |
$this |
getTargetLanguageCode
Required. Target language code (BCP-47) of the output.
Returns | |
---|---|
Type | Description |
string |
setTargetLanguageCode
Required. Target language code (BCP-47) of the output.
Parameter | |
---|---|
Name | Description |
var |
string
|
Returns | |
---|---|
Type | Description |
$this |
getSampleRateHertz
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).
Returns | |
---|---|
Type | Description |
int |
setSampleRateHertz
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).
Parameter | |
---|---|
Name | Description |
var |
int
|
Returns | |
---|---|
Type | Description |
$this |
getModel
Optional. google-provided-model/video
and
google-provided-model/enhanced-phone-call
are premium models.
google-provided-model/phone-call
is not premium model.
Returns | |
---|---|
Type | Description |
string |
setModel
Optional. google-provided-model/video
and
google-provided-model/enhanced-phone-call
are premium models.
google-provided-model/phone-call
is not premium model.
Parameter | |
---|---|
Name | Description |
var |
string
|
Returns | |
---|---|
Type | Description |
$this |