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// Generated by the protocol buffer compiler.  DO NOT EDIT!
// source: google/cloud/dialogflow/v2/session.proto

package com.google.cloud.dialogflow.v2;

/**
 *
 *
 * 
 * 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 {@code google.cloud.dialogflow.v2.AudioEncoding} */ public enum AudioEncoding implements com.google.protobuf.ProtocolMessageEnum { /** * * *
   * Not specified.
   * 
* * AUDIO_ENCODING_UNSPECIFIED = 0; */ AUDIO_ENCODING_UNSPECIFIED(0), /** * * *
   * Uncompressed 16-bit signed little-endian samples (Linear PCM).
   * 
* * AUDIO_ENCODING_LINEAR_16 = 1; */ AUDIO_ENCODING_LINEAR_16(1), /** * * *
   * [`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_FLAC(2), /** * * *
   * 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
   * 
* * AUDIO_ENCODING_MULAW = 3; */ AUDIO_ENCODING_MULAW(3), /** * * *
   * Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
   * 
* * AUDIO_ENCODING_AMR = 4; */ AUDIO_ENCODING_AMR(4), /** * * *
   * Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000.
   * 
* * AUDIO_ENCODING_AMR_WB = 5; */ AUDIO_ENCODING_AMR_WB(5), /** * * *
   * 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_OGG_OPUS(6), /** * * *
   * 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; */ AUDIO_ENCODING_SPEEX_WITH_HEADER_BYTE(7), UNRECOGNIZED(-1), ; /** * * *
   * Not specified.
   * 
* * AUDIO_ENCODING_UNSPECIFIED = 0; */ public static final int AUDIO_ENCODING_UNSPECIFIED_VALUE = 0; /** * * *
   * Uncompressed 16-bit signed little-endian samples (Linear PCM).
   * 
* * AUDIO_ENCODING_LINEAR_16 = 1; */ public static final int AUDIO_ENCODING_LINEAR_16_VALUE = 1; /** * * *
   * [`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; */ public static final int AUDIO_ENCODING_FLAC_VALUE = 2; /** * * *
   * 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
   * 
* * AUDIO_ENCODING_MULAW = 3; */ public static final int AUDIO_ENCODING_MULAW_VALUE = 3; /** * * *
   * Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
   * 
* * AUDIO_ENCODING_AMR = 4; */ public static final int AUDIO_ENCODING_AMR_VALUE = 4; /** * * *
   * Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000.
   * 
* * AUDIO_ENCODING_AMR_WB = 5; */ public static final int AUDIO_ENCODING_AMR_WB_VALUE = 5; /** * * *
   * Opus encoded audio frames in Ogg container
   * ([OggOpus](https://wiki.xiph.org/OggOpus)).
   * `sample_rate_hertz` must be 16000.
   * 
* * AUDIO_ENCODING_OGG_OPUS = 6; */ public static final int AUDIO_ENCODING_OGG_OPUS_VALUE = 6; /** * * *
   * 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; */ public static final int AUDIO_ENCODING_SPEEX_WITH_HEADER_BYTE_VALUE = 7; public final int getNumber() { if (this == UNRECOGNIZED) { throw new java.lang.IllegalArgumentException( "Can't get the number of an unknown enum value."); } return value; } /** @deprecated Use {@link #forNumber(int)} instead. */ @java.lang.Deprecated public static AudioEncoding valueOf(int value) { return forNumber(value); } public static AudioEncoding forNumber(int value) { switch (value) { case 0: return AUDIO_ENCODING_UNSPECIFIED; case 1: return AUDIO_ENCODING_LINEAR_16; case 2: return AUDIO_ENCODING_FLAC; case 3: return AUDIO_ENCODING_MULAW; case 4: return AUDIO_ENCODING_AMR; case 5: return AUDIO_ENCODING_AMR_WB; case 6: return AUDIO_ENCODING_OGG_OPUS; case 7: return AUDIO_ENCODING_SPEEX_WITH_HEADER_BYTE; default: return null; } } public static com.google.protobuf.Internal.EnumLiteMap internalGetValueMap() { return internalValueMap; } private static final com.google.protobuf.Internal.EnumLiteMap internalValueMap = new com.google.protobuf.Internal.EnumLiteMap() { public AudioEncoding findValueByNumber(int number) { return AudioEncoding.forNumber(number); } }; public final com.google.protobuf.Descriptors.EnumValueDescriptor getValueDescriptor() { return getDescriptor().getValues().get(ordinal()); } public final com.google.protobuf.Descriptors.EnumDescriptor getDescriptorForType() { return getDescriptor(); } public static final com.google.protobuf.Descriptors.EnumDescriptor getDescriptor() { return com.google.cloud.dialogflow.v2.SessionProto.getDescriptor().getEnumTypes().get(0); } private static final AudioEncoding[] VALUES = values(); public static AudioEncoding valueOf(com.google.protobuf.Descriptors.EnumValueDescriptor desc) { if (desc.getType() != getDescriptor()) { throw new java.lang.IllegalArgumentException("EnumValueDescriptor is not for this type."); } if (desc.getIndex() == -1) { return UNRECOGNIZED; } return VALUES[desc.getIndex()]; } private final int value; private AudioEncoding(int value) { this.value = value; } // @@protoc_insertion_point(enum_scope:google.cloud.dialogflow.v2.AudioEncoding) }




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