Package com.google.cloud.speech.v1

Enum RecognitionConfig.AudioEncoding

java.lang.Object
java.lang.Enum<RecognitionConfig.AudioEncoding>
com.google.cloud.speech.v1.RecognitionConfig.AudioEncoding
All Implemented Interfaces:
com.google.protobuf.Internal.EnumLite, com.google.protobuf.ProtocolMessageEnum, Serializable, Comparable<RecognitionConfig.AudioEncoding>
Enclosing class:
RecognitionConfig
public static enum RecognitionConfig.AudioEncoding extends Enum<RecognitionConfig.AudioEncoding> implements com.google.protobuf.ProtocolMessageEnum
 The encoding of the audio data sent in the request.

 All encodings support only 1 channel (mono) audio, unless the
 `audio_channel_count` and `enable_separate_recognition_per_channel` fields
 are set.

 For best results, the audio source should be captured and transmitted using
 a lossless encoding (`FLAC` or `LINEAR16`). The accuracy of the speech
 recognition can be reduced if lossy codecs are used to capture or transmit
 audio, particularly if background noise is present. Lossy codecs include
 `MULAW`, `AMR`, `AMR_WB`, `OGG_OPUS`, `SPEEX_WITH_HEADER_BYTE`, `MP3`,
 and `WEBM_OPUS`.

 The `FLAC` and `WAV` audio file formats include a header that describes the
 included audio content. You can request recognition for `WAV` files that
 contain either `LINEAR16` or `MULAW` encoded audio.
 If you send `FLAC` or `WAV` audio file format in
 your request, you do not need to specify an `AudioEncoding`; the audio
 encoding format is determined from the file header. If you specify
 an `AudioEncoding` when you send  send `FLAC` or `WAV` audio, the
 encoding configuration must match the encoding described in the audio
 header; otherwise the request returns an
 [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT] error
 code.
Protobuf enum google.cloud.speech.v1.RecognitionConfig.AudioEncoding

  • Enum Constant Details

    • ENCODING_UNSPECIFIED

      public static final RecognitionConfig.AudioEncoding ENCODING_UNSPECIFIED
       Not specified.
      ENCODING_UNSPECIFIED = 0;

    • LINEAR16

      public static final RecognitionConfig.AudioEncoding LINEAR16
       Uncompressed 16-bit signed little-endian samples (Linear PCM).
      LINEAR16 = 1;

    • FLAC

      public static final RecognitionConfig.AudioEncoding 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`. `FLAC` stream
 encoding supports 16-bit and 24-bit samples, however, not all fields in
 `STREAMINFO` are supported.
      FLAC = 2;

    • MULAW

      public static final RecognitionConfig.AudioEncoding MULAW
       8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
      MULAW = 3;

    • AMR

      public static final RecognitionConfig.AudioEncoding AMR
       Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
      AMR = 4;

    • AMR_WB

      public static final RecognitionConfig.AudioEncoding AMR_WB
       Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000.
      AMR_WB = 5;

    • OGG_OPUS

      public static final RecognitionConfig.AudioEncoding OGG_OPUS
       Opus encoded audio frames in Ogg container
 ([OggOpus](https://wiki.xiph.org/OggOpus)).
 `sample_rate_hertz` must be one of 8000, 12000, 16000, 24000, or 48000.
      OGG_OPUS = 6;

    • SPEEX_WITH_HEADER_BYTE

      public static final RecognitionConfig.AudioEncoding SPEEX_WITH_HEADER_BYTE
       Although the use of lossy encodings is not recommended, if a very low
 bitrate encoding is required, `OGG_OPUS` is highly preferred over
 Speex encoding. The [Speex](https://speex.org/)  encoding supported by
 Cloud Speech API has a header byte in each block, as in MIME type
 `audio/x-speex-with-header-byte`.
 It is a variant of the RTP Speex encoding defined in
 [RFC 5574](https://tools.ietf.org/html/rfc5574).
 The stream is a sequence of blocks, one block per RTP packet. Each block
 starts with a byte containing the length of the block, in bytes, followed
 by one or more frames of Speex data, padded to an integral number of
 bytes (octets) as specified in RFC 5574. In other words, each RTP header
 is replaced with a single byte containing the block length. Only Speex
 wideband is supported. `sample_rate_hertz` must be 16000.
      SPEEX_WITH_HEADER_BYTE = 7;

    • MP3

      public static final RecognitionConfig.AudioEncoding MP3
       MP3 audio. MP3 encoding is a Beta feature and only available in
 v1p1beta1. 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.
      MP3 = 8;

    • WEBM_OPUS

      public static final RecognitionConfig.AudioEncoding WEBM_OPUS
       Opus encoded audio frames in WebM container
 ([OggOpus](https://wiki.xiph.org/OggOpus)). `sample_rate_hertz` must be
 one of 8000, 12000, 16000, 24000, or 48000.
      WEBM_OPUS = 9;

    • UNRECOGNIZED

      public static final RecognitionConfig.AudioEncoding UNRECOGNIZED

  • Field Details

    • ENCODING_UNSPECIFIED_VALUE

      public static final int ENCODING_UNSPECIFIED_VALUE
       Not specified.
      ENCODING_UNSPECIFIED = 0;
      See Also:

    • LINEAR16_VALUE

      public static final int LINEAR16_VALUE
       Uncompressed 16-bit signed little-endian samples (Linear PCM).
      LINEAR16 = 1;
      See Also:

    • FLAC_VALUE

      public static final int FLAC_VALUE
       `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`. `FLAC` stream
 encoding supports 16-bit and 24-bit samples, however, not all fields in
 `STREAMINFO` are supported.
      FLAC = 2;
      See Also:

    • MULAW_VALUE

      public static final int MULAW_VALUE
       8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
      MULAW = 3;
      See Also:

    • AMR_VALUE

      public static final int AMR_VALUE
       Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
      AMR = 4;
      See Also:

    • AMR_WB_VALUE

      public static final int AMR_WB_VALUE
       Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000.
      AMR_WB = 5;
      See Also:

    • OGG_OPUS_VALUE

      public static final int OGG_OPUS_VALUE
       Opus encoded audio frames in Ogg container
 ([OggOpus](https://wiki.xiph.org/OggOpus)).
 `sample_rate_hertz` must be one of 8000, 12000, 16000, 24000, or 48000.
      OGG_OPUS = 6;
      See Also:

    • SPEEX_WITH_HEADER_BYTE_VALUE

      public static final int SPEEX_WITH_HEADER_BYTE_VALUE
       Although the use of lossy encodings is not recommended, if a very low
 bitrate encoding is required, `OGG_OPUS` is highly preferred over
 Speex encoding. The [Speex](https://speex.org/)  encoding supported by
 Cloud Speech API has a header byte in each block, as in MIME type
 `audio/x-speex-with-header-byte`.
 It is a variant of the RTP Speex encoding defined in
 [RFC 5574](https://tools.ietf.org/html/rfc5574).
 The stream is a sequence of blocks, one block per RTP packet. Each block
 starts with a byte containing the length of the block, in bytes, followed
 by one or more frames of Speex data, padded to an integral number of
 bytes (octets) as specified in RFC 5574. In other words, each RTP header
 is replaced with a single byte containing the block length. Only Speex
 wideband is supported. `sample_rate_hertz` must be 16000.
      SPEEX_WITH_HEADER_BYTE = 7;
      See Also:

    • MP3_VALUE

      public static final int MP3_VALUE
       MP3 audio. MP3 encoding is a Beta feature and only available in
 v1p1beta1. 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.
      MP3 = 8;
      See Also:

    • WEBM_OPUS_VALUE

      public static final int WEBM_OPUS_VALUE
       Opus encoded audio frames in WebM container
 ([OggOpus](https://wiki.xiph.org/OggOpus)). `sample_rate_hertz` must be
 one of 8000, 12000, 16000, 24000, or 48000.
      WEBM_OPUS = 9;
      See Also:

  • Method Details

    • values

      public static RecognitionConfig.AudioEncoding[] values()
      Returns an array containing the constants of this enum type, in the order they are declared.
      Returns:
      an array containing the constants of this enum type, in the order they are declared

    • valueOf

      public static RecognitionConfig.AudioEncoding valueOf(String name)
      Returns the enum constant of this type with the specified name. The string must match exactly an identifier used to declare an enum constant in this type. (Extraneous whitespace characters are not permitted.)
      Parameters:
      name - the name of the enum constant to be returned.
      Returns:
      the enum constant with the specified name
      Throws:
      IllegalArgumentException - if this enum type has no constant with the specified name
      NullPointerException - if the argument is null

    • getNumber

      public final int getNumber()
      Specified by:
      getNumber in interface com.google.protobuf.Internal.EnumLite
      Specified by:
      getNumber in interface com.google.protobuf.ProtocolMessageEnum

    • valueOf

      @Deprecated public static RecognitionConfig.AudioEncoding valueOf(int value)
      Deprecated.
      Returns the enum constant of this type with the specified name. The string must match exactly an identifier used to declare an enum constant in this type. (Extraneous whitespace characters are not permitted.)
      Parameters:
      value - the name of the enum constant to be returned.
      Returns:
      the enum constant with the specified name
      Throws:
      IllegalArgumentException - if this enum type has no constant with the specified name
      NullPointerException - if the argument is null

    • forNumber

      public static RecognitionConfig.AudioEncoding forNumber(int value)
      Parameters:
      value - The numeric wire value of the corresponding enum entry.
      Returns:
      The enum associated with the given numeric wire value.

    • internalGetValueMap

      public static com.google.protobuf.Internal.EnumLiteMap<RecognitionConfig.AudioEncoding> internalGetValueMap()

    • getValueDescriptor

      public final com.google.protobuf.Descriptors.EnumValueDescriptor getValueDescriptor()
      Specified by:
      getValueDescriptor in interface com.google.protobuf.ProtocolMessageEnum

    • getDescriptorForType

      public final com.google.protobuf.Descriptors.EnumDescriptor getDescriptorForType()
      Specified by:
      getDescriptorForType in interface com.google.protobuf.ProtocolMessageEnum

    • getDescriptor

      public static com.google.protobuf.Descriptors.EnumDescriptor getDescriptor()

    • valueOf

      public static RecognitionConfig.AudioEncoding valueOf(com.google.protobuf.Descriptors.EnumValueDescriptor desc)
      Returns the enum constant of this type with the specified name. The string must match exactly an identifier used to declare an enum constant in this type. (Extraneous whitespace characters are not permitted.)
      Parameters:
      desc - the name of the enum constant to be returned.
      Returns:
      the enum constant with the specified name
      Throws:
      IllegalArgumentException - if this enum type has no constant with the specified name
      NullPointerException - if the argument is null