#!/usr/bin/env bash

function show_usage()
{
	echo
	echo "USAGE"
	echo "-----"
	echo
	echo "  SERVER_URL=https://my.mediasoup-demo.org:4443 ROOM_ID=test MEDIA_URL=https://foo.com/test.mp4 LOCAL_VIDEO_PATH=/tmp/video.mp4 ./gstreamer.sh"
	echo
	echo "  where:"
	echo "  - SERVER_URL is the URL of the mediasoup-demo API server"
	echo "  - ROOM_ID is the id of the mediasoup-demo room (it must exist in advance)"
	echo "  - MEDIA_URL is the HTTP URL to a audio+video file (such as a .mp4 file), optional if this command was already run and LOCAL_VIDEO_PATH is already generated)"
	echo "  - LOCAL_VIDEO_PATH is full file path to the generated mp4 file after fetching it from MEDIA_URL"
	echo
	echo "REQUIREMENTS"
	echo "------------"
	echo
	echo "  - gstreamer: stream audio and video (https://gstreamer.freedesktop.org)"
	echo "  - httpie: command line HTTP client (https://httpie.org)"
	echo "  - jq: command-line JSON processor (https://stedolan.github.io/jq)"
	echo
}

echo

if [ -z "${SERVER_URL}" ] ; then
	>&2 echo "ERROR: missing SERVER_URL environment variable"
	show_usage
	exit 1
fi

if [ -z "${ROOM_ID}" ] ; then
	>&2 echo "ERROR: missing ROOM_ID environment variable"
	show_usage
	exit 1
fi

if [ -z "${LOCAL_VIDEO_PATH}" ] ; then
	>&2 echo "ERROR: missing LOCAL_VIDEO_PATH environment variable"
	show_usage
	exit 1
fi

if [ -z "${MEDIA_URL}" ] && [ ! -f "${LOCAL_VIDEO_PATH}" ] ; then
	>&2 echo "ERROR: missing MEDIA_URL environment variable and local video ${LOCAL_VIDEO_PATH} doesn't exist"
	show_usage
	exit 1
fi

if [ "$(command -v gst-launch-1.0)" == "" ] ; then
	>&2 echo "ERROR: gst-launch-1.0 command not found, must install GStreamer"
	show_usage
	exit 1
fi

if [ "$(command -v http)" == "" ] ; then
	>&2 echo "ERROR: http command not found, must install httpie"
	show_usage
	exit 1
fi

if [ "$(command -v jq)" == "" ] ; then
	>&2 echo "ERROR: jq command not found, must install jq"
	show_usage
	exit 1
fi

set -e

BROADCASTER_ID=$(LC_CTYPE=C tr -dc A-Za-z0-9 < /dev/urandom | fold -w ${1:-32} | head -n 1)
HTTPIE_COMMAND="http --check-status"
AUDIO_SSRC=1111
AUDIO_PT=100
VIDEO_SSRC=2222
VIDEO_PT=101

#
# Verify that a room with id ROOM_ID does exist by sending a simlpe HTTP GET. If
# not abort since we are not allowed to initiate a room..
#
echo ">>> verifying that room '${ROOM_ID}' exists..."

${HTTPIE_COMMAND} \
	GET ${SERVER_URL}/rooms/${ROOM_ID} > /dev/null

#
# Create a Broadcaster entity in the server by sending a POST with our metadata.
# Note that this is not related to mediasoup at all, but will become just a JS
# object in the Node.js application to hold our metadata and mediasoup Transports
# and Producers.
#
echo ">>> creating Broadcaster..."

${HTTPIE_COMMAND} \
	POST ${SERVER_URL}/rooms/${ROOM_ID}/broadcasters \
	id="${BROADCASTER_ID}" \
	displayName="Broadcaster" \
	device:='{"name": "GStreamer"}' \
	> /dev/null

#
# Upon script termination delete the Broadcaster in the server by sending a
# HTTP DELETE.
#
trap 'echo ">>> script exited with status code $?"; ${HTTPIE_COMMAND} DELETE ${SERVER_URL}/rooms/${ROOM_ID}/broadcasters/${BROADCASTER_ID} > /dev/null' EXIT

#
# Create a PlainTransport in the mediasoup to send our audio using plain RTP
# over UDP. Do it via HTTP post specifying type:"plain" and comedia:true and
# rtcpMux:false.
#
echo ">>> creating mediasoup PlainTransport for producing audio..."

res=$(${HTTPIE_COMMAND} \
	POST ${SERVER_URL}/rooms/${ROOM_ID}/broadcasters/${BROADCASTER_ID}/transports \
	type="plain" \
	comedia:=true \
	rtcpMux:=false \
	2> /dev/null)

#
# Parse JSON response into Shell variables and extract the PlainTransport id,
# IP, port and RTCP port.
#
eval "$(echo ${res} | jq -r '@sh "audioTransportId=\(.id) audioTransportIp=\(.ip) audioTransportPort=\(.port) audioTransportRtcpPort=\(.rtcpPort)"')"

#
# Create a PlainTransport in the mediasoup to send our video using plain RTP
# over UDP. Do it via HTTP post specifying type:"plain" and comedia:true and
# rtcpMux:false.
#
echo ">>> creating mediasoup PlainTransport for producing video..."

res=$(${HTTPIE_COMMAND} \
	POST ${SERVER_URL}/rooms/${ROOM_ID}/broadcasters/${BROADCASTER_ID}/transports \
	type="plain" \
	comedia:=true \
	rtcpMux:=false \
	2> /dev/null)

#
# Parse JSON response into Shell variables and extract the PlainTransport id,
# IP, port and RTCP port.
#
eval "$(echo ${res} | jq -r '@sh "videoTransportId=\(.id) videoTransportIp=\(.ip) videoTransportPort=\(.port) videoTransportRtcpPort=\(.rtcpPort)"')"

#
# Create a mediasoup Producer to send audio by sending our RTP parameters via a
# HTTP POST.
#
echo ">>> creating mediasoup audio Producer..."

${HTTPIE_COMMAND} -v \
	POST ${SERVER_URL}/rooms/${ROOM_ID}/broadcasters/${BROADCASTER_ID}/transports/${audioTransportId}/producers \
	kind="audio" \
	rtpParameters:="{ \"codecs\": [{ \"mimeType\":\"audio/opus\", \"payloadType\":${AUDIO_PT}, \"clockRate\":48000, \"channels\":2, \"parameters\":{ \"sprop-stereo\":1 } }], \"encodings\": [{ \"ssrc\":${AUDIO_SSRC} }] }" \
	> /dev/null

#
# Create a mediasoup Producer to send video by sending our RTP parameters via a
# HTTP POST.
#
echo ">>> creating mediasoup video Producer..."

${HTTPIE_COMMAND} -v \
	POST ${SERVER_URL}/rooms/${ROOM_ID}/broadcasters/${BROADCASTER_ID}/transports/${videoTransportId}/producers \
	kind="video" \
	rtpParameters:="{ \"codecs\": [{ \"mimeType\":\"video/h264\", \"payloadType\":${VIDEO_PT}, \"clockRate\":90000, \"parameters\":{ \"packetization-mode\":1, \"profile-level-id\":\"42e032\", \"level-asymmetry-allowed\":1 } }], \"encodings\": [{ \"ssrc\":${VIDEO_SSRC} }] }" \
	> /dev/null

#
# NOTE: This is just to fetch and convert the file so must be run just once.
# If the local video file already exists don't run this.
if [ ! -f "${LOCAL_VIDEO_PATH}" ] ; then
	echo ">>> running ffmpeg..."

	ffmpeg \
		-analyzeduration 20M \
		-probesize 20M \
		-re -i ${MEDIA_URL} \
		-map 0:v:0 \
		-c:v copy \
		-async 10000 \
		-f tee [select=v:f=h264]${LOCAL_VIDEO_PATH}
fi

echo ">>> running gstreamer..."

gst-launch-1.0 \
	-v \
	rtpbin name=rtpbin \
	rtp-profile=avpf \
	filesrc location=${LOCAL_VIDEO_PATH} \
	do-timestamp=true \
	! queue \
	! h264parse \
	! rtph264pay \
	mtu=1300 \
	ssrc=${VIDEO_SSRC} \
	pt=${VIDEO_PT} \
	! rtprtxqueue \
	max-size-time=1000 \
	max-size-packets=0 \
	! rtpbin.send_rtp_sink_0 rtpbin.send_rtp_src_0 \
	! udpsink host=${videoTransportIp} port=${videoTransportPort} \
	rtpbin.send_rtcp_src_0 \
	! udpsink host=${videoTransportIp} port=${videoTransportRtcpPort} \
	sync=false \
	async=false