Enum AudioEncoding (4.61.0)

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public enum AudioEncoding extends Enum<AudioEncoding> implements ProtocolMessageEnum

Audio encoding of the audio content sent in the conversational query request. Refer to the Cloud Speech API documentation for more details.

Protobuf enum google.cloud.dialogflow.v2beta1.AudioEncoding

Implements

ProtocolMessageEnum

Inherited Members

Enum.<T>valueOf(Class<T>,String)

Enum.clone()

Enum.compareTo(E)

Enum.equals(Object)

Enum.finalize()

Enum.getDeclaringClass()

Enum.hashCode()

Enum.name()

Enum.ordinal()

Enum.toString()

Object.getClass()

Object.notify()

Object.notifyAll()

Object.wait()

Object.wait(long)

Object.wait(long,int)

Static Fields
Name	Description
AUDIO_ENCODING_ALAW	8-bit samples that compand 13-bit audio samples using G.711 PCMU/a-law. AUDIO_ENCODING_ALAW = 8;
AUDIO_ENCODING_ALAW_VALUE	8-bit samples that compand 13-bit audio samples using G.711 PCMU/a-law. AUDIO_ENCODING_ALAW = 8;
AUDIO_ENCODING_AMR	Adaptive Multi-Rate Narrowband codec. sample_rate_hertz must be 8000. AUDIO_ENCODING_AMR = 4;
AUDIO_ENCODING_AMR_VALUE	Adaptive Multi-Rate Narrowband codec. sample_rate_hertz must be 8000. AUDIO_ENCODING_AMR = 4;
AUDIO_ENCODING_AMR_WB	Adaptive Multi-Rate Wideband codec. sample_rate_hertz must be 16000. AUDIO_ENCODING_AMR_WB = 5;
AUDIO_ENCODING_AMR_WB_VALUE	Adaptive Multi-Rate Wideband codec. sample_rate_hertz must be 16000. AUDIO_ENCODING_AMR_WB = 5;
AUDIO_ENCODING_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. AUDIO_ENCODING_FLAC = 2;
AUDIO_ENCODING_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. AUDIO_ENCODING_FLAC = 2;
AUDIO_ENCODING_LINEAR_16	Uncompressed 16-bit signed little-endian samples (Linear PCM). AUDIO_ENCODING_LINEAR_16 = 1;
AUDIO_ENCODING_LINEAR_16_VALUE	Uncompressed 16-bit signed little-endian samples (Linear PCM). AUDIO_ENCODING_LINEAR_16 = 1;
AUDIO_ENCODING_MULAW	8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law. AUDIO_ENCODING_MULAW = 3;
AUDIO_ENCODING_MULAW_VALUE	8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law. AUDIO_ENCODING_MULAW = 3;
AUDIO_ENCODING_OGG_OPUS	Opus encoded audio frames in Ogg container (OggOpus). sample_rate_hertz must be 16000. AUDIO_ENCODING_OGG_OPUS = 6;
AUDIO_ENCODING_OGG_OPUS_VALUE	Opus encoded audio frames in Ogg container (OggOpus). sample_rate_hertz must be 16000. AUDIO_ENCODING_OGG_OPUS = 6;
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 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. 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_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 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. 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_UNSPECIFIED	Not specified. AUDIO_ENCODING_UNSPECIFIED = 0;
AUDIO_ENCODING_UNSPECIFIED_VALUE	Not specified. AUDIO_ENCODING_UNSPECIFIED = 0;
UNRECOGNIZED

Static Methods
Name	Description
forNumber(int value)
getDescriptor()
internalGetValueMap()
valueOf(Descriptors.EnumValueDescriptor desc)
valueOf(int value)	Deprecated. Use #forNumber(int) instead.
valueOf(String name)
values()

Methods
Name	Description
getDescriptorForType()
getNumber()
getValueDescriptor()