到目前为止,我一直在尝试sipster,但是它有一些令人讨厌的限制(e.g. lack of configurability)。任何想法如何做到这一点?也许有一个像asterisk-manager这样的星号的节点包装?
更详细地说,基本思想是
最佳答案
这篇文章很老,看起来在Google方面已经有了很大的改进,在语音处理器本身(变得越来越准确)以及在Node.js方面,作为与Google交互的Node.js client方面, Cloud Speech API会定期更新。
根据@arheops的建议,您可能想看看Asterisk的EAGI和Node.js,以便让音频样本被Google转录。
以下EAGI bash脚本可能会在这方面有所帮助(详细说明可用here):
#!/bin/bash
# Read all variables sent by Asterisk store them as an array, but won't use them
declare -a array
while read -e ARG && [ "$ARG" ] ; do
array=(` echo $ARG | sed -e 's/://'`)
export ${array[0]}=${array[1]}
done
# First argument is language
case "$1" in
"fr-FR" | "en-GB" | "es-ES" | "it-IT" )
LANG=$1
;;
*)
LANG=en-US
;;
esac
NODECMD=$(which node)
# Second argument is a timeout, in seconds. The duration to wait for voice input form the caller.
DURATION=$2
SAMPLE_RATE=8000
SAMPLE_SIZE_BYTES=2
let "SAMPLE_SIZE_BITS = SAMPLE_SIZE_BYTES * 8"
# EAGI_AUDIO_FORMAT is an asterisk variable that specifies the sample rate and
# sample size (usually 16 bits per sample) of the caller's voice stream.
# Depending on the codec used here, you can get sample rate values ranging from
# 8000Hz (e.g. G.711 uLaw) to 48000Hz (e.g. opus).
echo "GET VARIABLE EAGI_AUDIO_FORMAT"
read line
EAGI_AUDIO_FORMAT=$(echo $line | sed -r 's/.*\((.*)\).*/\1/')
# 5 seconds of audio input are gathered in ( SAMPLE_RATE / sample_size ) * 5 bytes
# - SAMPLE_RATE is set as per EAGI_AUDIO_FORMAT
# - sample_size is set to 2 (16 bits per sample)
#
# We don't do much here to adapt the sample rate, this code should be improved
case "${EAGI_AUDIO_FORMAT}" in
"slin48")
SAMPLE_RATE=48000
;;
*)
SAMPLE_RATE=8000
;;
esac
# Temporary file to store raw audio samples
AUDIO_FILE=/tmp/audio-${SAMPLE_SIZE_BITS}_bits-${SAMPLE_RATE}_hz-${DURATION}_sec.raw
# We use `dd` here to copy the raw audio samples we're getting from file
# descriptor 3 (this is the Enhanced version in EAGI) to the temporary file.
# The number of blocks to copy is a function of the DURATION to record audio and
# the sample rate. SAMPLE_SIZE_BYTES cannot be changed as it is assumed that each
# sample is 16 bits in size.
let "COUNT = SAMPLE_RATE * SAMPLE_SIZE_BYTES * DURATION"
# By default, dd stores blocks of 512 bytes
let "BLOCKS = COUNT / 512"
echo "exec noop \"Number of bytes to store : ${COUNT}\""
read line
echo "exec noop \"Number of dd blocks to store : ${BLOCKS}\""
read line
echo "exec playback \"beep\""
read line
dd if=/dev/fd/3 count=${BLOCKS} of=${AUDIO_FILE}
echo "exec noop \"File saved !\""
echo "exec noop \"AUDIO_FILE : ${AUDIO_FILE}\""
read line
echo "exec noop \"SAMPLE_RATE : ${SAMPLE_RATE}\""
read line
echo "exec noop \"LANG : ${LANG}\""
read line
# Submit audio to Google Cloud Speech API and get the result
export GOOGLE_APPLICATION_CREDENTIALS=/usr/local/node_programs/service_account_file.json
RES=$(${NODECMD} /usr/local/node_programs/nodejs-speech/samples/recognize.js sync ${AUDIO_FILE} -e LINEAR16 -r ${SAMPLE_RATE} -l ${LANG})
# clean up result returned from recognize.js :
# - remove new lines
# - remove 'Transcription :' header
RES=$(echo $RES | tr -d '\n' | sed -e 's/Transcription: \(.*$\)/\1/')
# Set GOOGLE_TRANSCRIPTION_RESULT variable, remove temporary file
# and continue dialplan execution
echo "set variable GOOGLE_TRANSCRIPTION_RESULT \"${RES}\""
read line
/bin/rm -f ${AUDIO_FILE}
exit 0
希望这可以帮助!
关于node.js - 如何在 Node 中接收SIP音频并将WAV流发送到Google语音识别API?,我们在Stack Overflow上找到一个类似的问题: https://stackoverflow.com/questions/40555589/