Interface TranslateSpeechConfigOrBuilder

  • All Superinterfaces:
    com.google.protobuf.MessageLiteOrBuilder, com.google.protobuf.MessageOrBuilder
    All Known Implementing Classes:
    TranslateSpeechConfig, TranslateSpeechConfig.Builder

    public interface TranslateSpeechConfigOrBuilder
    extends com.google.protobuf.MessageOrBuilder
    • Method Detail

      • getAudioEncoding

        String 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](https://wikipedia.org/wiki/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.
         
        string audio_encoding = 1 [(.google.api.field_behavior) = REQUIRED];
        Returns:
        The audioEncoding.
      • getAudioEncodingBytes

        com.google.protobuf.ByteString getAudioEncodingBytes()
         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](https://wikipedia.org/wiki/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.
         
        string audio_encoding = 1 [(.google.api.field_behavior) = REQUIRED];
        Returns:
        The bytes for audioEncoding.
      • getSourceLanguageCode

        String getSourceLanguageCode()
         Required. Source language code (BCP-47) of the input audio.
         
        string source_language_code = 2 [(.google.api.field_behavior) = REQUIRED];
        Returns:
        The sourceLanguageCode.
      • getSourceLanguageCodeBytes

        com.google.protobuf.ByteString getSourceLanguageCodeBytes()
         Required. Source language code (BCP-47) of the input audio.
         
        string source_language_code = 2 [(.google.api.field_behavior) = REQUIRED];
        Returns:
        The bytes for sourceLanguageCode.
      • getTargetLanguageCode

        String getTargetLanguageCode()
         Required. Target language code (BCP-47) of the output.
         
        string target_language_code = 3 [(.google.api.field_behavior) = REQUIRED];
        Returns:
        The targetLanguageCode.
      • getTargetLanguageCodeBytes

        com.google.protobuf.ByteString getTargetLanguageCodeBytes()
         Required. Target language code (BCP-47) of the output.
         
        string target_language_code = 3 [(.google.api.field_behavior) = REQUIRED];
        Returns:
        The bytes for targetLanguageCode.
      • getSampleRateHertz

        int 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).
         
        int32 sample_rate_hertz = 4 [(.google.api.field_behavior) = OPTIONAL];
        Returns:
        The sampleRateHertz.
      • getModel

        String 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.
         
        string model = 5 [(.google.api.field_behavior) = OPTIONAL];
        Returns:
        The model.
      • getModelBytes

        com.google.protobuf.ByteString getModelBytes()
         Optional. `google-provided-model/video` and
         `google-provided-model/enhanced-phone-call` are premium models.
         `google-provided-model/phone-call` is not premium model.
         
        string model = 5 [(.google.api.field_behavior) = OPTIONAL];
        Returns:
        The bytes for model.