Dialogflow V2 API - Module Google::Cloud::Dialogflow::V2::AudioEncoding (v1.0.0)

Reference documentation and code samples for the Dialogflow V2 API module Google::Cloud::Dialogflow::V2::AudioEncoding.

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



value: 0
Not specified.


value: 1
Uncompressed 16-bit signed little-endian samples (Linear PCM).


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.


value: 3
8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.


value: 4
Adaptive Multi-Rate Narrowband codec. sample_rate_hertz must be 8000.


value: 5
Adaptive Multi-Rate Wideband codec. sample_rate_hertz must be 16000.


value: 6
Opus encoded audio frames in Ogg container (OggOpus). sample_rate_hertz must be 16000.


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.