Reference documentation and code samples for the Google Cloud Dialogflow V2 Client class AudioEncoding.
Audio encoding of the audio content sent in the conversational query request.
Refer to the Cloud Speech API documentation for more details.
Protobuf type google.cloud.dialogflow.v2.AudioEncoding
Methods
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Constants
AUDIO_ENCODING_UNSPECIFIED
Value: 0
Not specified.
Generated from protobuf enum AUDIO_ENCODING_UNSPECIFIED = 0;
AUDIO_ENCODING_LINEAR_16
Value: 1
Uncompressed 16-bit signed little-endian samples (Linear PCM).
Generated from protobuf enum AUDIO_ENCODING_LINEAR_16 = 1;
AUDIO_ENCODING_FLAC
Value: 2
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.
Generated from protobuf enum AUDIO_ENCODING_FLAC = 2;
AUDIO_ENCODING_MULAW
Value: 3
8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
Generated from protobuf enum AUDIO_ENCODING_MULAW = 3;
AUDIO_ENCODING_AMR
Value: 4
Adaptive Multi-Rate Narrowband codec. sample_rate_hertz
must be 8000.
Generated from protobuf enum AUDIO_ENCODING_AMR = 4;
AUDIO_ENCODING_AMR_WB
Value: 5
Adaptive Multi-Rate Wideband codec. sample_rate_hertz
must be 16000.
Generated from protobuf enum AUDIO_ENCODING_AMR_WB = 5;
AUDIO_ENCODING_OGG_OPUS
Value: 6
Opus encoded audio frames in Ogg container (OggOpus).
sample_rate_hertz
must be 16000.
Generated from protobuf enum AUDIO_ENCODING_OGG_OPUS = 6;
AUDIO_ENCODING_SPEEX_WITH_HEADER_BYTE
Value: 7
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.
Generated from protobuf enum AUDIO_ENCODING_SPEEX_WITH_HEADER_BYTE = 7;