public enum AudioEncoding extends java.lang.Enum<AudioEncoding> implements ProtocolMessageEnum
Audio encoding of the audio content sent in the conversational query request. Refer to the [Cloud Speech API documentation](/speech/docs/basics) for more details.Protobuf enum
google.cloud.dialogflow.v2beta1.AudioEncoding
Enum Constant and Description |
---|
AUDIO_ENCODING_AMR
Adaptive Multi-Rate Narrowband codec.
|
AUDIO_ENCODING_AMR_WB
Adaptive Multi-Rate Wideband codec.
|
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`.
|
AUDIO_ENCODING_LINEAR_16
Uncompressed 16-bit signed little-endian samples (Linear PCM).
|
AUDIO_ENCODING_MULAW
8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
|
AUDIO_ENCODING_OGG_OPUS
Opus encoded audio frames in Ogg container
([OggOpus](https://wiki.xiph.org/OggOpus)).
|
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.
|
AUDIO_ENCODING_UNSPECIFIED
Not specified.
|
UNRECOGNIZED |
Modifier and Type | Field and Description |
---|---|
static int |
AUDIO_ENCODING_AMR_VALUE
Adaptive Multi-Rate Narrowband codec.
|
static int |
AUDIO_ENCODING_AMR_WB_VALUE
Adaptive Multi-Rate Wideband codec.
|
static 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`.
|
static int |
AUDIO_ENCODING_LINEAR_16_VALUE
Uncompressed 16-bit signed little-endian samples (Linear PCM).
|
static int |
AUDIO_ENCODING_MULAW_VALUE
8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
|
static int |
AUDIO_ENCODING_OGG_OPUS_VALUE
Opus encoded audio frames in Ogg container
([OggOpus](https://wiki.xiph.org/OggOpus)).
|
static 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.
|
static int |
AUDIO_ENCODING_UNSPECIFIED_VALUE
Not specified.
|
Modifier and Type | Method and Description |
---|---|
static AudioEncoding |
forNumber(int value) |
static Descriptors.EnumDescriptor |
getDescriptor() |
Descriptors.EnumDescriptor |
getDescriptorForType() |
int |
getNumber() |
Descriptors.EnumValueDescriptor |
getValueDescriptor() |
static Internal.EnumLiteMap<AudioEncoding> |
internalGetValueMap() |
static AudioEncoding |
valueOf(Descriptors.EnumValueDescriptor desc) |
static AudioEncoding |
valueOf(int value)
Deprecated.
Use
forNumber(int) instead. |
static AudioEncoding |
valueOf(java.lang.String name)
Returns the enum constant of this type with the specified name.
|
static AudioEncoding[] |
values()
Returns an array containing the constants of this enum type, in
the order they are declared.
|
public static final AudioEncoding AUDIO_ENCODING_UNSPECIFIED
Not specified.
AUDIO_ENCODING_UNSPECIFIED = 0;
public static final AudioEncoding AUDIO_ENCODING_LINEAR_16
Uncompressed 16-bit signed little-endian samples (Linear PCM).
AUDIO_ENCODING_LINEAR_16 = 1;
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;
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;
public static final AudioEncoding AUDIO_ENCODING_AMR
Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
AUDIO_ENCODING_AMR = 4;
public static final AudioEncoding AUDIO_ENCODING_AMR_WB
Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000.
AUDIO_ENCODING_AMR_WB = 5;
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;
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;
public static final AudioEncoding UNRECOGNIZED
public static final int AUDIO_ENCODING_UNSPECIFIED_VALUE
Not specified.
AUDIO_ENCODING_UNSPECIFIED = 0;
public static final int AUDIO_ENCODING_LINEAR_16_VALUE
Uncompressed 16-bit signed little-endian samples (Linear PCM).
AUDIO_ENCODING_LINEAR_16 = 1;
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;
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;
public static final int AUDIO_ENCODING_AMR_VALUE
Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
AUDIO_ENCODING_AMR = 4;
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;
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;
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;
public static AudioEncoding[] values()
for (AudioEncoding c : AudioEncoding.values()) System.out.println(c);
public static AudioEncoding valueOf(java.lang.String name)
name
- the name of the enum constant to be returned.java.lang.IllegalArgumentException
- if this enum type has no constant with the specified namejava.lang.NullPointerException
- if the argument is nullpublic final int getNumber()
getNumber
in interface Internal.EnumLite
getNumber
in interface ProtocolMessageEnum
@Deprecated public static AudioEncoding valueOf(int value)
forNumber(int)
instead.public static AudioEncoding forNumber(int value)
public static Internal.EnumLiteMap<AudioEncoding> internalGetValueMap()
public final Descriptors.EnumValueDescriptor getValueDescriptor()
getValueDescriptor
in interface ProtocolMessageEnum
public final Descriptors.EnumDescriptor getDescriptorForType()
getDescriptorForType
in interface ProtocolMessageEnum
public static final Descriptors.EnumDescriptor getDescriptor()
public static AudioEncoding valueOf(Descriptors.EnumValueDescriptor desc)