Use phone as microphone

android-mic.sh

#!/bin/bash

# Use phone as microphone
#
# This is a striped down version of the ipwebcam-gst tool, initially 
# made for learning purposes, with two little addons:
#   - Remove forwarded ports after streaming is finished
#   - Increase microphone volume
#   - No waiting for enter to end streaming, just press ctrl-C (habit)
#
# Original tool can be found on https://github.com/bluezio/ipwebcam-gst

GST_AUDIO_MIMETYPE="audio/x-raw"
GST_AUDIO_FORMAT="format=S16LE"
GST_AUDIO_LAYOUT=",layout=interleaved"
GST_AUDIO_RATE="rate=44100"
GST_AUDIO_CHANNELS="channels=1"

##########################################################
#		   ADB forward ports                     #
##########################################################

adb forward tcp:4747 tcp:4747

##########################################################
#		     Pulse Audio setup                   #
##########################################################

DEFAULT_SINK=$(pacmd dump | grep set-default-sink | cut -f2 -d " ")
DEFAULT_SOURCE=$(pacmd dump | grep set-default-source | cut -f2 -d " ")

module_id_by_sinkname() {
    pacmd list-sinks | grep -e 'name:' -e 'module:' | grep -A1 "name: <$1>" | grep module: | cut -f2 -d: | tr -d ' '
}

SINK_NAME="ipwebcam"
SINK_ID=$(module_id_by_sinkname $SINK_NAME)
ECANCEL_ID=$(module_id_by_sinkname "${SINK_NAME}_echo_cancel")


PA_AUDIO_CAPS="$GST_AUDIO_FORMAT $GST_AUDIO_RATE $GST_AUDIO_CHANNELS"

# Registering audio device if not yet registered
if [ -z $SINK_ID ] ; then
    SINK_ID=$(pactl load-module module-null-sink \
                    sink_name="$SINK_NAME" \
                    $PA_AUDIO_CAPS \
                    sink_properties="device.description='IP\ Webcam'")
fi

if [ -z $ECANCEL_ID ] ; then
    ECANCEL_ID=$(pactl load-module module-echo-cancel \
                       sink_name="${SINK_NAME}_echo_cancel" \
                       source_master="$SINK_NAME.monitor" \
                       sink_master="$DEFAULT_SINK" \
                       $PA_AUDIO_CAPS \
                       aec_method="webrtc" save_aec=true use_volume_sharing=true)
fi

pactl set-default-source $SINK_NAME.monitor

# Either sleep or use queue in gstreamer audio caps
sleep 2

##########################################################
#		   Start streaming                       #
##########################################################

GST_AUDIO_CAPS="$GST_AUDIO_MIMETYPE,$GST_AUDIO_FORMAT$GST_AUDIO_LAYOUT,$GST_AUDIO_RATE,$GST_AUDIO_CHANNELS"

pipeline_audio() {
  echo souphttpsrc location=http://localhost:4747/audio.wav do-timestamp=true is-live=true \
    ! $GST_AUDIO_CAPS \
    ! volume volume=5 \
    ! pulsesink device="$SINK_NAME" 
}

gst-launch-1.0 -e -vt --gst-plugin-spew --gst-debug="souphttpsrc:0,pulse:0" $( pipeline_audio ) 

##########################################################
#		   Clean after streaming                 #
##########################################################

pactl set-default-source ${DEFAULT_SOURCE}
pactl unload-module ${ECANCEL_ID}
pactl unload-module ${SINK_ID}

adb forward --remove tcp:4747

echo "Bye"