I know Asterisk has a speech recognition interface built in, but I need to go beyond that, with APIs like Lex, Wit or Luis etc.
These APIs can repond really quickly and accurately as they can receive and interpret an audio stream, but in all the Asterisk speech recognition tools I can find, they all say that they need to save the speech to a file, then convert, then upload via whatever API.
There’s also the very cheap/free high quality speech synthesis services like Amazon Polly, which can also return an audio stream object (or save a file).
Removing the “record in Asterisk/store as file/convert file/upload file <> receive stream/save file/convert file/playback in Asterisk” part of the sequence would save vital seconds of silence and caller annoyance.
I am looking at, for example, Google’s speech to text service, and it can cope with the following codecs, some of which can be used directly from Asterisk, yes? So there is no need to transcode? As Google returns “live” word results, I could set an API to “watch” for a trigger word, and then use the AGI to trigger something in Asterisk.
Well, that is how I am assuming it can/should be done!
Also, there is JackAudio. And I’m thinking to myself that surely something could be done here, but then also, surely one of the people far more skilled than me would have already done this?
So, have they either missed a trick here, or is there something in a recent version of Asterisk
LINEAR16 Uncompressed 16-bit signed little-endian samples (Linear PCM).
FLAC 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 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
AMR Adaptive Multi-Rate Narrowband codec. sample_rate_hertz must be 8000.
AMR_WB Adaptive Multi-Rate Wideband codec. sample_rate_hertz must be 16000.
OGG_OPUS Opus encoded audio frames in Ogg container (OggOpus). sample_rate_hertz must be 16000.
SPEEX_WITH_HEADER_BYTE 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.
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