Cloud Speech V1 Client - Class AudioEncoding (1.19.2)

Reference documentation and code samples for the Cloud Speech V1 Client class AudioEncoding.

The encoding of the audio data sent in the request.

All encodings support only 1 channel (mono) audio, unless the audio_channel_count and enable_separate_recognition_per_channel fields are set. For best results, the audio source should be captured and transmitted using a lossless encoding (FLAC or LINEAR16). The accuracy of the speech recognition can be reduced if lossy codecs are used to capture or transmit audio, particularly if background noise is present. Lossy codecs include MULAW, AMR, AMR_WB, OGG_OPUS, SPEEX_WITH_HEADER_BYTE, MP3, and WEBM_OPUS. The FLAC and WAV audio file formats include a header that describes the included audio content. You can request recognition for WAV files that contain either LINEAR16 or MULAW encoded audio. If you send FLAC or WAV audio file format in your request, you do not need to specify an AudioEncoding; the audio encoding format is determined from the file header. If you specify an AudioEncoding when you send send FLAC or WAV audio, the encoding configuration must match the encoding described in the audio header; otherwise the request returns an google.rpc.Code.INVALID_ARGUMENT error code.

Protobuf type google.cloud.speech.v1.RecognitionConfig.AudioEncoding

Namespace

Google \ Cloud \ Speech \ V1 \ RecognitionConfig

Methods

static::name

Parameter
Name	Description
`value`	`mixed`

static::value

Parameter
Name	Description
`name`	`mixed`

Constants

ENCODING_UNSPECIFIED

Value: 0

Not specified.

Generated from protobuf enum ENCODING_UNSPECIFIED = 0;

LINEAR16

Value: 1

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

Generated from protobuf enum LINEAR16 = 1;

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 FLAC = 2;

MULAW

Value: 3

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

Generated from protobuf enum MULAW = 3;

AMR

Value: 4

Adaptive Multi-Rate Narrowband codec. sample_rate_hertz must be 8000.

Generated from protobuf enum AMR = 4;

AMR_WB

Value: 5

Adaptive Multi-Rate Wideband codec. sample_rate_hertz must be 16000.

Generated from protobuf enum AMR_WB = 5;

OGG_OPUS

Value: 6

Opus encoded audio frames in Ogg container (OggOpus).

sample_rate_hertz must be one of 8000, 12000, 16000, 24000, or 48000.

Generated from protobuf enum OGG_OPUS = 6;

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 Cloud Speech 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 SPEEX_WITH_HEADER_BYTE = 7;

MP3

Value: 8

MP3 audio. MP3 encoding is a Beta feature and only available in v1p1beta1. Support all standard MP3 bitrates (which range from 32-320 kbps). When using this encoding, sample_rate_hertz has to match the sample rate of the file being used.

Generated from protobuf enum MP3 = 8;

WEBM_OPUS

Value: 9

Opus encoded audio frames in WebM container (OggOpus). sample_rate_hertz must be one of 8000, 12000, 16000, 24000, or 48000.

Generated from protobuf enum WEBM_OPUS = 9;