Initial commit

6 files changed, 572 insertions, 0 deletions

diff --git a/go.mod b/go.mod
new file mode 100644
index 0000000..d300919
--- /dev/null
+++ b/go.mod
@@ -0,0 +1,8 @@
+module github.com/pion/rtwatch
+
+go 1.13
+
+require (
+	github.com/gorilla/websocket v1.4.1
+	github.com/pion/webrtc/v2 v2.1.4
+)
diff --git a/go.sum b/go.sum
new file mode 100644
index 0000000..cb395e8
--- /dev/null
+++ b/go.sum
@@ -0,0 +1,102 @@
+github.com/cheekybits/genny v1.0.0 h1:uGGa4nei+j20rOSeDeP5Of12XVm7TGUd4dJA9RDitfE=
+github.com/cheekybits/genny v1.0.0/go.mod h1:+tQajlRqAUrPI7DOSpB0XAqZYtQakVtB7wXkRAgjxjQ=
+github.com/davecgh/go-spew v1.1.0/go.mod h1:J7Y8YcW2NihsgmVo/mv3lAwl/skON4iLHjSsI+c5H38=
+github.com/davecgh/go-spew v1.1.1 h1:vj9j/u1bqnvCEfJOwUhtlOARqs3+rkHYY13jYWTU97c=
+github.com/davecgh/go-spew v1.1.1/go.mod h1:J7Y8YcW2NihsgmVo/mv3lAwl/skON4iLHjSsI+c5H38=
+github.com/fsnotify/fsnotify v1.4.7 h1:IXs+QLmnXW2CcXuY+8Mzv/fWEsPGWxqefPtCP5CnV9I=
+github.com/fsnotify/fsnotify v1.4.7/go.mod h1:jwhsz4b93w/PPRr/qN1Yymfu8t87LnFCMoQvtojpjFo=
+github.com/golang/mock v1.2.0 h1:28o5sBqPkBsMGnC6b4MvE2TzSr5/AT4c/1fLqVGIwlk=
+github.com/golang/mock v1.2.0/go.mod h1:oTYuIxOrZwtPieC+H1uAHpcLFnEyAGVDL/k47Jfbm0A=
+github.com/golang/protobuf v1.2.0 h1:P3YflyNX/ehuJFLhxviNdFxQPkGK5cDcApsge1SqnvM=
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diff --git a/gst/gst.c b/gst/gst.c
new file mode 100644
index 0000000..e84316e
--- /dev/null
+++ b/gst/gst.c
@@ -0,0 +1,110 @@
+#include "gst.h"
+
+#include <gst/app/gstappsrc.h>
+
+typedef struct SampleHandlerUserData {
+  int pipelineId;
+} SampleHandlerUserData;
+
+GMainLoop *gstreamer_send_main_loop = NULL;
+void gstreamer_send_start_mainloop(void) {
+  gstreamer_send_main_loop = g_main_loop_new(NULL, FALSE);
+
+  g_main_loop_run(gstreamer_send_main_loop);
+}
+
+static gboolean gstreamer_send_bus_call(GstBus *bus, GstMessage *msg, gpointer data) {
+  GstElement *pipeline = GST_ELEMENT(data);
+
+  switch (GST_MESSAGE_TYPE(msg)) {
+  case GST_MESSAGE_EOS:
+    if (!gst_element_seek (pipeline, 1.0, GST_FORMAT_TIME, GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_KEY_UNIT | GST_SEEK_FLAG_SKIP,
+             GST_SEEK_TYPE_SET, 0,
+             GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE)) {
+        g_print ("EOS restart failed\n");
+        exit(1);
+    }
+    break;
+
+  case GST_MESSAGE_ERROR: {
+    gchar *debug;
+    GError *error;
+
+    gst_message_parse_error(msg, &error, &debug);
+    g_free(debug);
+
+    g_printerr("GStreamer Error: %s\n", error->message);
+    g_error_free(error);
+    exit(1);
+  }
+  default:
+    break;
+  }
+
+  return TRUE;
+}
+
+GstFlowReturn gstreamer_send_new_sample_handler(GstElement *object, gpointer user_data) {
+  GstSample *sample = NULL;
+  GstBuffer *buffer = NULL;
+  gpointer copy = NULL;
+  gsize copy_size = 0;
+  int *isVideo = (int *) user_data;
+
+  g_signal_emit_by_name (object, "pull-sample", &sample);
+  if (sample) {
+    buffer = gst_sample_get_buffer(sample);
+    if (buffer) {
+      gst_buffer_extract_dup(buffer, 0, gst_buffer_get_size(buffer), &copy, &copy_size);
+      goHandlePipelineBuffer(copy, copy_size, GST_BUFFER_DURATION(buffer), *isVideo);
+    }
+    gst_sample_unref (sample);
+  }
+
+  return GST_FLOW_OK;
+}
+
+GstElement *gstreamer_send_create_pipeline(char *pipeline) {
+  gst_init(NULL, NULL);
+  GError *error = NULL;
+  return gst_parse_launch(pipeline, &error);
+}
+
+void gstreamer_send_start_pipeline(GstElement *pipeline) {
+  GstBus *bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline));
+  gst_bus_add_watch(bus, gstreamer_send_bus_call, pipeline);
+  gst_object_unref(bus);
+
+  GstElement *audio = gst_bin_get_by_name(GST_BIN(pipeline), "audio"),
+             *video = gst_bin_get_by_name(GST_BIN(pipeline), "video");
+
+  int *isAudio = malloc(sizeof(int)),
+      *isVideo = malloc(sizeof(int));
+
+  *isVideo = 1;
+  *isAudio = 0;
+
+  g_object_set(audio, "emit-signals", TRUE, NULL);
+  g_signal_connect(audio, "new-sample", G_CALLBACK(gstreamer_send_new_sample_handler), isAudio);
+
+  g_object_set(video, "emit-signals", TRUE, NULL);
+  g_signal_connect(video, "new-sample", G_CALLBACK(gstreamer_send_new_sample_handler), isVideo);
+
+  gstreamer_send_play_pipeline(pipeline);
+}
+
+void gstreamer_send_play_pipeline(GstElement *pipeline) {
+  gst_element_set_state(pipeline, GST_STATE_PLAYING);
+}
+
+void gstreamer_send_pause_pipeline(GstElement *pipeline) {
+  gst_element_set_state(pipeline, GST_STATE_PAUSED);
+}
+
+void gstreamer_send_seek(GstElement *pipeline, int64_t seek_pos) {
+    if (!gst_element_seek (pipeline, 1.0, GST_FORMAT_TIME, GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_KEY_UNIT | GST_SEEK_FLAG_SKIP,
+             GST_SEEK_TYPE_SET, seek_pos * GST_SECOND,
+             GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE)) {
+        g_print ("Seek failed!\n");
+    }
+}
diff --git a/gst/gst.go b/gst/gst.go
new file mode 100644
index 0000000..f8a5385
--- /dev/null
+++ b/gst/gst.go
@@ -0,0 +1,96 @@
+package gst
+
+/*
+#cgo pkg-config: gstreamer-1.0 gstreamer-app-1.0
+
+#include "gst.h"
+
+*/
+import "C"
+import (
+	"fmt"
+	"io"
+	"sync"
+	"unsafe"
+
+	"github.com/pion/webrtc/v2"
+	"github.com/pion/webrtc/v2/pkg/media"
+)
+
+func init() {
+	go C.gstreamer_send_start_mainloop()
+}
+
+// Pipeline is a wrapper for a GStreamer Pipeline
+type Pipeline struct {
+	Pipeline   *C.GstElement
+	audioTrack *webrtc.Track
+	videoTrack *webrtc.Track
+}
+
+var pipeline = &Pipeline{}
+var pipelinesLock sync.Mutex
+
+// CreatePipeline creates a GStreamer Pipeline
+func CreatePipeline(containerPath string, audioTrack, videoTrack *webrtc.Track) *Pipeline {
+	pipelineStr := fmt.Sprintf("filesrc location=\"%s\" ! decodebin name=demux ! queue ! x264enc bframes=0 speed-preset=veryfast key-int-max=60 ! video/x-h264,stream-format=byte-stream ! appsink name=video demux. ! queue ! audioconvert ! audioresample ! opusenc ! appsink name=audio", containerPath)
+
+	pipelineStrUnsafe := C.CString(pipelineStr)
+	defer C.free(unsafe.Pointer(pipelineStrUnsafe))
+
+	pipelinesLock.Lock()
+	defer pipelinesLock.Unlock()
+	pipeline = &Pipeline{
+		Pipeline:   C.gstreamer_send_create_pipeline(pipelineStrUnsafe),
+		audioTrack: audioTrack,
+		videoTrack: videoTrack,
+	}
+	return pipeline
+}
+
+// Start starts the GStreamer Pipeline
+func (p *Pipeline) Start() {
+	// This will signal to goHandlePipelineBuffer
+	// and provide a method for cancelling sends.
+	C.gstreamer_send_start_pipeline(p.Pipeline)
+}
+
+// Play sets the pipeline to PLAYING
+func (p *Pipeline) Play() {
+	C.gstreamer_send_play_pipeline(p.Pipeline)
+}
+
+// Pause sets the pipeline to PAUSED
+func (p *Pipeline) Pause() {
+	C.gstreamer_send_pause_pipeline(p.Pipeline)
+}
+
+// SeekToTime seeks on the pipeline
+func (p *Pipeline) SeekToTime(seekPos int64) {
+	C.gstreamer_send_seek(p.Pipeline, C.int64_t(seekPos))
+}
+
+const (
+	videoClockRate = 90000
+	audioClockRate = 48000
+)
+
+//export goHandlePipelineBuffer
+func goHandlePipelineBuffer(buffer unsafe.Pointer, bufferLen C.int, duration C.int, isVideo C.int) {
+	var track *webrtc.Track
+	var samples uint32
+
+	if isVideo == 1 {
+		samples = uint32(videoClockRate * (float32(duration) / 1000000000))
+		track = pipeline.videoTrack
+	} else {
+		samples = uint32(audioClockRate * (float32(duration) / 1000000000))
+		track = pipeline.audioTrack
+	}
+
+	if err := track.WriteSample(media.Sample{Data: C.GoBytes(buffer, bufferLen), Samples: samples}); err != nil && err != io.ErrClosedPipe {
+		panic(err)
+	}
+
+	C.free(buffer)
+}
diff --git a/gst/gst.h b/gst/gst.h
new file mode 100644
index 0000000..1103a6a
--- /dev/null
+++ b/gst/gst.h
@@ -0,0 +1,20 @@
+#ifndef GST_H
+#define GST_H
+
+#include <glib.h>
+#include <gst/gst.h>
+#include <stdint.h>
+#include <stdlib.h>
+
+extern void goHandlePipelineBuffer(void *buffer, int bufferLen, int samples, int isVideo);
+
+GstElement *gstreamer_send_create_pipeline(char *pipeline);
+void gstreamer_send_start_pipeline(GstElement *pipeline);
+
+void gstreamer_send_play_pipeline(GstElement *pipeline);
+void gstreamer_send_pause_pipeline(GstElement *pipeline);
+void gstreamer_send_seek(GstElement *pipeline, int64_t seek_pos);
+
+void gstreamer_send_start_mainloop(void);
+
+#endif
diff --git a/main.go b/main.go
new file mode 100644
index 0000000..8e6befc
--- /dev/null
+++ b/main.go
@@ -0,0 +1,236 @@
+package main
+
+import (
+	"encoding/json"
+	"flag"
+	"fmt"
+	"log"
+	"net/http"
+	"strconv"
+
+	"github.com/gorilla/websocket"
+	"github.com/pion/rtwatch/gst"
+	"github.com/pion/webrtc/v2"
+)
+
+const homeHTML = `<!DOCTYPE html>
+<html lang="en">
+	<head>
+		<title>synced-playback</title>
+	</head>
+	<body id="body">
+		<video id="video1" autoplay playsinline></video>
+
+		<div>
+		  <input type="number" id="seekTime" value="30">
+		  <button type="button" onClick="seekClick()">Seek</button>
+		  <button type="button" onClick="playClick()">Play</button>
+		  <button type="button" onClick="pauseClick()">Pause</button>
+		</div>
+
+		<script>
+			let conn = new WebSocket('ws://' + window.location.host + '/ws')
+			let pc = new RTCPeerConnection()
+
+			window.seekClick = () => {
+				conn.send(JSON.stringify({event: 'seek', data: document.getElementById('seekTime').value}))
+			}
+			window.playClick = () => {
+				conn.send(JSON.stringify({event: 'play', data: ''}))
+			}
+			window.pauseClick = () => {
+				conn.send(JSON.stringify({event: 'pause', data: ''}))
+			}
+
+			pc.ontrack = function (event) {
+			  if (event.track.kind === 'audio') {
+				return
+			  }
+			  var el = document.getElementById('video1')
+			  el.srcObject = event.streams[0]
+			  el.autoplay = true
+			  el.controls = true
+			}
+
+			conn.onopen = () => {
+				pc.createOffer({offerToReceiveVideo: true, offerToReceiveAudio: true}).then(offer => {
+					pc.setLocalDescription(offer)
+					conn.send(JSON.stringify({event: 'offer', data: JSON.stringify(offer)}))
+				})
+			}
+			conn.onclose = evt => {
+				console.log('Connection closed')
+			}
+			conn.onmessage = evt => {
+				let msg = JSON.parse(evt.data)
+				if (!msg) {
+					return console.log('failed to parse msg')
+				}
+
+				switch (msg.event) {
+				case 'answer':
+					answer = JSON.parse(msg.data)
+					if (!answer) {
+						return console.log('failed to parse answer')
+					}
+					pc.setRemoteDescription(answer)
+				}
+			}
+			window.conn = conn
+		</script>
+	</body>
+</html>
+`
+
+var (
+	upgrader = websocket.Upgrader{
+		ReadBufferSize:  1024,
+		WriteBufferSize: 1024,
+	}
+
+	peerConnectionConfig = webrtc.Configuration{}
+
+	audioTrack = &webrtc.Track{}
+	videoTrack = &webrtc.Track{}
+	pipeline   = &gst.Pipeline{}
+)
+
+type websocketMessage struct {
+	Event string `json:"event"`
+	Data  string `json:"data"`
+}
+
+func main() {
+	containerPath := ""
+	httpListenAddress := ""
+	flag.StringVar(&containerPath, "container-path", "", "path to the media file you want to playback")
+	flag.StringVar(&httpListenAddress, "http-listen-address", ":8080", "address for HTTP server to listen on")
+	flag.Parse()
+
+	if containerPath == "" {
+		panic("-container-path must be specified")
+	}
+
+	pc, err := webrtc.NewPeerConnection(webrtc.Configuration{
+		ICEServers: []webrtc.ICEServer{
+			{
+				URLs: []string{"stun:stun.l.google.com:19302"},
+			},
+		},
+	})
+	if err != nil {
+		log.Fatal(err)
+	}
+
+	videoTrack, err = pc.NewTrack(webrtc.DefaultPayloadTypeH264, 5000, "synced-video", "synced-video")
+	if err != nil {
+		log.Fatal(err)
+	}
+
+	audioTrack, err = pc.NewTrack(webrtc.DefaultPayloadTypeOpus, 5001, "synced-video", "synced-video")
+	if err != nil {
+		log.Fatal(err)
+	}
+
+	pipeline = gst.CreatePipeline(containerPath, audioTrack, videoTrack)
+	pipeline.Start()
+
+	http.HandleFunc("/", serveHome)
+	http.HandleFunc("/ws", serveWs)
+
+	fmt.Println(fmt.Sprintf("Video file '%s' is now available on '%s', have fun!", containerPath, httpListenAddress))
+	log.Fatal(http.ListenAndServe(httpListenAddress, nil))
+}
+
+func handleWebsocketMessage(pc *webrtc.PeerConnection, ws *websocket.Conn, message *websocketMessage) error {
+	switch message.Event {
+	case "play":
+		pipeline.Play()
+	case "pause":
+		pipeline.Pause()
+	case "seek":
+		i, err := strconv.ParseInt(message.Data, 0, 64)
+		if err != nil {
+			log.Print(err)
+		}
+		pipeline.SeekToTime(i)
+	case "offer":
+		offer := webrtc.SessionDescription{}
+		if err := json.Unmarshal([]byte(message.Data), &offer); err != nil {
+			return err
+		}
+
+		if err := pc.SetRemoteDescription(offer); err != nil {
+			return err
+		}
+
+		answer, err := pc.CreateAnswer(nil)
+		if err != nil {
+			return err
+		}
+		if err := pc.SetLocalDescription(answer); err != nil {
+			return err
+		}
+
+		answerString, err := json.Marshal(answer)
+		if err != nil {
+			return err
+		}
+
+		if err = ws.WriteJSON(&websocketMessage{
+			Event: "answer",
+			Data:  string(answerString),
+		}); err != nil {
+			return err
+		}
+	}
+	return nil
+}
+
+func serveWs(w http.ResponseWriter, r *http.Request) {
+	ws, err := upgrader.Upgrade(w, r, nil)
+	if err != nil {
+		if _, ok := err.(websocket.HandshakeError); !ok {
+			log.Println(err)
+		}
+		return
+	}
+
+	peerConnection, err := webrtc.NewPeerConnection(peerConnectionConfig)
+	if err != nil {
+		log.Print(err)
+		return
+	} else if _, err = peerConnection.AddTrack(audioTrack); err != nil {
+		log.Print(err)
+		return
+	} else if _, err = peerConnection.AddTrack(videoTrack); err != nil {
+		log.Print(err)
+		return
+	}
+
+	defer func() {
+		if err := peerConnection.Close(); err != nil {
+			log.Println(err)
+		}
+	}()
+
+	message := &websocketMessage{}
+	for {
+		_, msg, err := ws.ReadMessage()
+		if err != nil {
+			break
+		} else if err := json.Unmarshal(msg, &message); err != nil {
+			log.Print(err)
+			return
+		}
+
+		if err := handleWebsocketMessage(peerConnection, ws, message); err != nil {
+			log.Print(err)
+		}
+	}
+}
+
+func serveHome(w http.ResponseWriter, r *http.Request) {
+	w.Header().Set("Content-Type", "text/html; charset=utf-8")
+	fmt.Fprintf(w, homeHTML)
+}