如何接收 SIP 音频并将 wav 流发送到节点中的 Google 语音识别 API？

Question

到目前为止，我一直在尝试sipster，但它有一些令人生畏的限制 (e.g. lack of configurability)。任何想法如何做到这一点？也许使用像asterisk-manager 这样的星号节点包装器？

在一些更详细的基本思想是

• 运行虚拟 sip 客户端，可以接收 SIP 连接
• 将该连接中的音频转换为常规 wav 格式
• 将 wav 音频流式传输到 Google 语音 API
• 有其他方法可以通过节点作用于 sip 流，例如播放声音

Answer 1

这篇文章已经很老了，看起来 Google 方面的情况已经有了很大的改进，无论是语音处理器本身，它变得越来越准确，还有 Node.js 方面，如 @987654321 @ 与 Google Cloud Speech API 的接口会定期更新。

根据@arheops 的建议，您可能想看看 Asterisk 的 EAGI 和 Node.js，以便获得由 Google 转录的音频样本。

以下 EAGI bash 脚本可能在这方面有所帮助（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

希望这会有所帮助！

Answer 2

最简单的方法 - 使用星号 EAGI 界面并将声音从标准输入/流读取到谷歌。

然而，目前谷歌语音识别 api 并不稳定。有些日子它只是停止工作，然后第二天开始工作。