AudioEncoding(value)
Audio encoding of the audio content sent in the conversational query
request. Refer to the Cloud Speech API
documentation <https://cloud.google.com/speech-to-text/docs/basics>
__
for more details.
Values:
AUDIO_ENCODING_UNSPECIFIED (0):
Not specified.
AUDIO_ENCODING_LINEAR_16 (1):
Uncompressed 16-bit signed little-endian
samples (Linear PCM).
AUDIO_ENCODING_FLAC (2):
`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.
FLACstream encoding supports 16-bit and
24-bit samples, however, not all fields in
STREAMINFOare supported.
AUDIO_ENCODING_MULAW (3):
8-bit samples that compand 14-bit audio
samples using G.711 PCMU/mu-law.
AUDIO_ENCODING_AMR (4):
Adaptive Multi-Rate Narrowband codec.
sample_rate_hertzmust be 8000.
AUDIO_ENCODING_AMR_WB (5):
Adaptive Multi-Rate Wideband codec.
sample_rate_hertzmust be 16000.
AUDIO_ENCODING_OGG_OPUS (6):
Opus encoded audio frames in Ogg container
(
OggOpus https://wiki.xiph.org/OggOpus__).
sample_rate_hertzmust be 16000.
AUDIO_ENCODING_SPEEX_WITH_HEADER_BYTE (7):
Although the use of lossy encodings is not recommended, if a
very low bitrate encoding is required,
OGG_OPUSis
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
- 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.