Google Cloud Dialogflow V2 Client - Class AudioEncoding (1.15.1)

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

Namespace

Google \ Cloud \ Dialogflow \ V2

Methods

static::name

Parameter
Name	Description
`value`	`mixed`

static::value

Parameter
Name	Description
`name`	`mixed`

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;

AUDIO_ENCODING_ALAW

Value: 8

8-bit samples that compand 13-bit audio samples using G.711 PCMU/a-law.

Generated from protobuf enum AUDIO_ENCODING_ALAW = 8;