Google Cloud Media Translation V1beta1 Client - Class TranslateSpeechConfig (0.3.2)

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

Namespace

Google \ Cloud \ MediaTranslation \ V1beta1

Methods

__construct

Constructor.

Parameters
NameDescription
data array

Optional. Data for populating the Message object.

↳ audio_encoding string

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 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. google-provided-model/video and google-provided-model/enhanced-phone-call are premium models. google-provided-model/phone-call is not premium model.

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 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.
Returns
TypeDescription
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 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.
Parameter
NameDescription
var string
Returns
TypeDescription
$this

getSourceLanguageCode

Required. Source language code (BCP-47) of the input audio.

Returns
TypeDescription
string

setSourceLanguageCode

Required. Source language code (BCP-47) of the input audio.

Parameter
NameDescription
var string
Returns
TypeDescription
$this

getTargetLanguageCode

Required. Target language code (BCP-47) of the output.

Returns
TypeDescription
string

setTargetLanguageCode

Required. Target language code (BCP-47) of the output.

Parameter
NameDescription
var string
Returns
TypeDescription
$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
TypeDescription
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
NameDescription
var int
Returns
TypeDescription
$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
TypeDescription
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
NameDescription
var string
Returns
TypeDescription
$this