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;
private static $valueToName = [
self::AUDIO_ENCODING_UNSPECIFIED => 'AUDIO_ENCODING_UNSPECIFIED',
self::AUDIO_ENCODING_LINEAR_16 => 'AUDIO_ENCODING_LINEAR_16',
self::AUDIO_ENCODING_FLAC => 'AUDIO_ENCODING_FLAC',
self::AUDIO_ENCODING_MULAW => 'AUDIO_ENCODING_MULAW',
self::AUDIO_ENCODING_AMR => 'AUDIO_ENCODING_AMR',
self::AUDIO_ENCODING_AMR_WB => 'AUDIO_ENCODING_AMR_WB',
self::AUDIO_ENCODING_OGG_OPUS => 'AUDIO_ENCODING_OGG_OPUS',
self::AUDIO_ENCODING_SPEEX_WITH_HEADER_BYTE => 'AUDIO_ENCODING_SPEEX_WITH_HEADER_BYTE',
];
public static function name($value)
{
if (!isset(self::$valueToName[$value])) {
throw new UnexpectedValueException(sprintf(
'Enum %s has no name defined for value %s', __CLASS__, $value));
}
return self::$valueToName[$value];
}
public static function value($name)
{
$const = __CLASS__ . '::' . strtoupper($name);
if (!defined($const)) {
throw new UnexpectedValueException(sprintf(
'Enum %s has no value defined for name %s', __CLASS__, $name));
}
return constant($const);
}
}