com.google.cloud.dialogflow.v2.AudioEncoding Maven / Gradle / Ivy
Go to download
Show more of this group Show more artifacts with this name
Show all versions of proto-google-cloud-dialogflow-v2 Show documentation
Show all versions of proto-google-cloud-dialogflow-v2 Show documentation
PROTO library for proto-google-cloud-dialogflow-v2
// 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)
}