Class AudioEncoding (2.19.1)

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.

Values: AUDIO_ENCODING_UNSPECIFIED (0): Not specified. AUDIO_ENCODING_LINEAR_16 (1): Uncompressed 16-bit signed little-endian samples (Linear PCM). AUDIO_ENCODING_FLAC (2): `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 ofLINEAR16.FLACstream encoding supports 16-bit and 24-bit samples, however, not all fields inSTREAMINFOare supported. AUDIO_ENCODING_MULAW (3): 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law. AUDIO_ENCODING_AMR (4): Adaptive Multi-Rate Narrowband codec.sample_rate_hertzmust be 8000. AUDIO_ENCODING_AMR_WB (5): Adaptive Multi-Rate Wideband codec.sample_rate_hertzmust be 16000. AUDIO_ENCODING_OGG_OPUS (6): Opus encoded audio frames in Ogg container (OggOpus https://wiki.xiph.org/OggOpus__).sample_rate_hertzmust be 16000. AUDIO_ENCODING_SPEEX_WITH_HEADER_BYTE (7): Although the use of lossy encodings is not recommended, if a very low bitrate encoding is required,OGG_OPUSis highly preferred over Speex encoding. TheSpeex https://speex.org/__ encoding supported by Dialogflow API has a header byte in each block, as in MIME typeaudio/x-speex-with-header-byte. It is a variant of the RTP Speex encoding defined inRFC 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

1. 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.