Package com.google.cloud.dialogflow.v2beta1

Enum AudioEncoding

  • All Implemented Interfaces:
    com.google.protobuf.Internal.EnumLite, com.google.protobuf.ProtocolMessageEnum, Serializable, Comparable<AudioEncoding>
    public enum AudioEncoding
extends Enum<AudioEncoding>
implements com.google.protobuf.ProtocolMessageEnum
     Audio encoding of the audio content sent in the conversational query request.
 Refer to the
 [Cloud Speech API
 documentation](https://cloud.google.com/speech-to-text/docs/basics) for more
 details.
    Protobuf enum google.cloud.dialogflow.v2beta1.AudioEncoding

    • Enum Constant Detail

      • AUDIO_ENCODING_UNSPECIFIED

        public static final AudioEncoding AUDIO_ENCODING_UNSPECIFIED
         Not specified.
        AUDIO_ENCODING_UNSPECIFIED = 0;

      • AUDIO_ENCODING_LINEAR_16

        public static final AudioEncoding AUDIO_ENCODING_LINEAR_16
         Uncompressed 16-bit signed little-endian samples (Linear PCM).
        AUDIO_ENCODING_LINEAR_16 = 1;

      • AUDIO_ENCODING_FLAC

        public static final AudioEncoding AUDIO_ENCODING_FLAC
         [`FLAC`](https://xiph.org/flac/documentation.html) (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.
        AUDIO_ENCODING_FLAC = 2;

      • AUDIO_ENCODING_MULAW

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

      • AUDIO_ENCODING_AMR

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

      • AUDIO_ENCODING_AMR_WB

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

      • AUDIO_ENCODING_OGG_OPUS

        public static final AudioEncoding AUDIO_ENCODING_OGG_OPUS
         Opus encoded audio frames in Ogg container
 ([OggOpus](https://wiki.xiph.org/OggOpus)).
 `sample_rate_hertz` must be 16000.
        AUDIO_ENCODING_OGG_OPUS = 6;

      • AUDIO_ENCODING_SPEEX_WITH_HEADER_BYTE

        public static final AudioEncoding AUDIO_ENCODING_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
 Dialogflow 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.
        AUDIO_ENCODING_SPEEX_WITH_HEADER_BYTE = 7;

    • Field Detail

      • AUDIO_ENCODING_UNSPECIFIED_VALUE

        public static final int AUDIO_ENCODING_UNSPECIFIED_VALUE
         Not specified.
        AUDIO_ENCODING_UNSPECIFIED = 0;
        See Also:
        Constant Field Values

      • AUDIO_ENCODING_LINEAR_16_VALUE

        public static final int AUDIO_ENCODING_LINEAR_16_VALUE
         Uncompressed 16-bit signed little-endian samples (Linear PCM).
        AUDIO_ENCODING_LINEAR_16 = 1;
        See Also:
        Constant Field Values

      • AUDIO_ENCODING_FLAC_VALUE

        public static final int AUDIO_ENCODING_FLAC_VALUE
         [`FLAC`](https://xiph.org/flac/documentation.html) (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.
        AUDIO_ENCODING_FLAC = 2;
        See Also:
        Constant Field Values

      • AUDIO_ENCODING_MULAW_VALUE

        public static final int AUDIO_ENCODING_MULAW_VALUE
         8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
        AUDIO_ENCODING_MULAW = 3;
        See Also:
        Constant Field Values

      • AUDIO_ENCODING_AMR_VALUE

        public static final int AUDIO_ENCODING_AMR_VALUE
         Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
        AUDIO_ENCODING_AMR = 4;
        See Also:
        Constant Field Values

      • AUDIO_ENCODING_AMR_WB_VALUE

        public static final int AUDIO_ENCODING_AMR_WB_VALUE
         Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000.
        AUDIO_ENCODING_AMR_WB = 5;
        See Also:
        Constant Field Values

      • AUDIO_ENCODING_OGG_OPUS_VALUE

        public static final int AUDIO_ENCODING_OGG_OPUS_VALUE
         Opus encoded audio frames in Ogg container
 ([OggOpus](https://wiki.xiph.org/OggOpus)).
 `sample_rate_hertz` must be 16000.
        AUDIO_ENCODING_OGG_OPUS = 6;
        See Also:
        Constant Field Values

      • AUDIO_ENCODING_SPEEX_WITH_HEADER_BYTE_VALUE

        public static final int AUDIO_ENCODING_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
 Dialogflow 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.
        AUDIO_ENCODING_SPEEX_WITH_HEADER_BYTE = 7;
        See Also:
        Constant Field Values

    • Method Detail

      • values

        public static AudioEncoding[] values()
        Returns an array containing the constants of this enum type, in the order they are declared. This method may be used to iterate over the constants as follows: 
        for (AudioEncoding c : AudioEncoding.values())
    System.out.println(c);
        Returns:
        an array containing the constants of this enum type, in the order they are declared

      • valueOf

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

      • valueOf

        public static 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