AudioEncoding(value)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][google.rpc.Code.INVALID_ARGUMENT]
error code.
Values:
ENCODING_UNSPECIFIED (0):
Not specified.
LINEAR16 (1):
Uncompressed 16-bit signed little-endian
samples (Linear PCM).
FLAC (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.
MULAW (3):
8-bit samples that compand 14-bit audio
samples using G.711 PCMU/mu-law.
AMR (4):
Adaptive Multi-Rate Narrowband codec. sample_rate_hertz
must be 8000.
AMR_WB (5):
Adaptive Multi-Rate Wideband codec. sample_rate_hertz
must be 16000.
OGG_OPUS (6):
Opus encoded audio frames in Ogg container
(OggOpus <https://wiki.xiph.org/OggOpus>).
sample_rate_hertz must be one of 8000, 12000, 16000,
24000, or 48000.
SPEEX_WITH_HEADER_BYTE (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 <https://speex.org/> 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 <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
- In other words, each RTP header is replaced with a
single byte containing the block length. Only Speex wideband
is supported.
sample_rate_hertzmust be 16000. MP3 (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_hertzhas to match the sample rate of the file being used. WEBM_OPUS (9): Opus encoded audio frames in WebM container (OggOpus <https://wiki.xiph.org/OggOpus>__).sample_rate_hertzmust be one of 8000, 12000, 16000, 24000, or 48000.