Google\Cloud\Dialogflow\V2\AudioEncoding */ class AudioEncoding { /** * Not specified. * * Generated from protobuf enum AUDIO_ENCODING_UNSPECIFIED = 0; */ const AUDIO_ENCODING_UNSPECIFIED = 0; /** * Uncompressed 16-bit signed little-endian samples (Linear PCM). * * Generated from protobuf enum AUDIO_ENCODING_LINEAR_16 = 1; */ const AUDIO_ENCODING_LINEAR_16 = 1; /** * [`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. * * Generated from protobuf enum AUDIO_ENCODING_FLAC = 2; */ const AUDIO_ENCODING_FLAC = 2; /** * 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law. * * Generated from protobuf enum AUDIO_ENCODING_MULAW = 3; */ const AUDIO_ENCODING_MULAW = 3; /** * Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000. * * Generated from protobuf enum AUDIO_ENCODING_AMR = 4; */ const AUDIO_ENCODING_AMR = 4; /** * Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000. * * Generated from protobuf enum AUDIO_ENCODING_AMR_WB = 5; */ const AUDIO_ENCODING_AMR_WB = 5; /** * Opus encoded audio frames in Ogg container * ([OggOpus](https://wiki.xiph.org/OggOpus)). * `sample_rate_hertz` must be 16000. * * Generated from protobuf enum AUDIO_ENCODING_OGG_OPUS = 6; */ const AUDIO_ENCODING_OGG_OPUS = 6; /** * 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. * * Generated from protobuf enum AUDIO_ENCODING_SPEEX_WITH_HEADER_BYTE = 7; */ const AUDIO_ENCODING_SPEEX_WITH_HEADER_BYTE = 7; }