Add example of using with Janus video-room

Resolves #268

examples/gstreamer-receive/main.go

@@ -5,8 +5,8 @@ import (
 	"time"
 
 	"github.com/pions/webrtc"
-	"github.com/pions/webrtc/examples/gstreamer-receive/gst"
 	"github.com/pions/webrtc/examples/util"
+	"github.com/pions/webrtc/examples/util/gstreamer-sink"
 	"github.com/pions/webrtc/pkg/ice"
 	"github.com/pions/webrtc/pkg/rtcp"
 )

examples/gstreamer-send-offer/main.go

@@ -4,8 +4,8 @@ import (
 	"fmt"
 
 	"github.com/pions/webrtc"
-	"github.com/pions/webrtc/examples/gstreamer-send/gst"
 	"github.com/pions/webrtc/examples/util"
+	"github.com/pions/webrtc/examples/util/gstreamer-src"
 	"github.com/pions/webrtc/pkg/ice"
 )
 

examples/gstreamer-send/gst/gst.c

@@ -1,88 +0,0 @@
-#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) {
-  switch (GST_MESSAGE_TYPE(msg)) {
-
-  case GST_MESSAGE_EOS:
-    g_print("End of stream\n");
-    exit(1);
-    break;
-
-  case GST_MESSAGE_ERROR: {
-    gchar *debug;
-    GError *error;
-
-    gst_message_parse_error(msg, &error, &debug);
-    g_free(debug);
-
-    g_printerr("Error: %s\n", error->message);
-    g_error_free(error);
-    exit(1);
-    break;
-  }
-  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;
-  SampleHandlerUserData *s = (SampleHandlerUserData *)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), s->pipelineId);
-    }
-    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, int pipelineId) {
-  SampleHandlerUserData *s = calloc(1, sizeof(SampleHandlerUserData));
-  s->pipelineId = pipelineId;
-
-  GstBus *bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline));
-  guint bus_watch_id = gst_bus_add_watch(bus, gstreamer_send_bus_call, NULL);
-  gst_object_unref(bus);
-
-  GstElement *appsink = gst_bin_get_by_name(GST_BIN(pipeline), "appsink");
-  g_object_set(appsink, "emit-signals", TRUE, NULL);
-  g_signal_connect(appsink, "new-sample", G_CALLBACK(gstreamer_send_new_sample_handler), s);
-
-  gst_element_set_state(pipeline, GST_STATE_PLAYING);
-}
-
-void gstreamer_send_stop_pipeline(GstElement *pipeline) {
-  gst_element_set_state(pipeline, GST_STATE_NULL);
-}
-
-

examples/gstreamer-send/gst/gst.go

@@ -1,99 +0,0 @@
-package gst
-
-/*
-#cgo pkg-config: gstreamer-1.0 gstreamer-app-1.0
-
-#include "gst.h"
-
-*/
-import "C"
-import (
-	"fmt"
-	"sync"
-	"unsafe"
-
-	"github.com/pions/webrtc"
-	"github.com/pions/webrtc/pkg/media"
-)
-
-func init() {
-	go C.gstreamer_send_start_mainloop()
-}
-
-// Pipeline is a wrapper for a GStreamer Pipeline
-type Pipeline struct {
-	Pipeline  *C.GstElement
-	in        chan<- media.RTCSample
-	id        int
-	codecName string
-}
-
-var pipelines = make(map[int]*Pipeline)
-var pipelinesLock sync.Mutex
-
-// CreatePipeline creates a GStreamer Pipeline
-func CreatePipeline(codecName string, in chan<- media.RTCSample) *Pipeline {
-	pipelineStr := "appsink name=appsink"
-	switch codecName {
-	case webrtc.VP8:
-		pipelineStr = "videotestsrc ! vp8enc ! " + pipelineStr
-	case webrtc.VP9:
-		pipelineStr = "videotestsrc ! vp9enc ! " + pipelineStr
-	case webrtc.H264:
-		pipelineStr = "videotestsrc ! video/x-raw,format=I420 ! x264enc bframes=0 speed-preset=veryfast key-int-max=60 ! video/x-h264,stream-format=byte-stream ! " + pipelineStr
-	case webrtc.Opus:
-		pipelineStr = "audiotestsrc ! opusenc ! " + pipelineStr
-	default:
-		panic("Unhandled codec " + codecName)
-	}
-
-	pipelineStrUnsafe := C.CString(pipelineStr)
-	defer C.free(unsafe.Pointer(pipelineStrUnsafe))
-
-	pipelinesLock.Lock()
-	defer pipelinesLock.Unlock()
-
-	pipeline := &Pipeline{
-		Pipeline:  C.gstreamer_send_create_pipeline(pipelineStrUnsafe),
-		in:        in,
-		id:        len(pipelines),
-		codecName: codecName,
-	}
-
-	pipelines[pipeline.id] = pipeline
-	return pipeline
-}
-
-// Start starts the GStreamer Pipeline
-func (p *Pipeline) Start() {
-	C.gstreamer_send_start_pipeline(p.Pipeline, C.int(p.id))
-}
-
-// Stop stops the GStreamer Pipeline
-func (p *Pipeline) Stop() {
-	C.gstreamer_send_stop_pipeline(p.Pipeline)
-}
-
-const (
-	videoClockRate = 90000
-	audioClockRate = 48000
-)
-
-//export goHandlePipelineBuffer
-func goHandlePipelineBuffer(buffer unsafe.Pointer, bufferLen C.int, duration C.int, pipelineID C.int) {
-	pipelinesLock.Lock()
-	defer pipelinesLock.Unlock()
-
-	if pipeline, ok := pipelines[int(pipelineID)]; ok {
-		var samples uint32
-		if pipeline.codecName == webrtc.Opus {
-			samples = uint32(audioClockRate * (float32(duration) / 1000000000))
-		} else {
-			samples = uint32(videoClockRate * (float32(duration) / 1000000000))
-		}
-		pipeline.in <- media.RTCSample{Data: C.GoBytes(buffer, bufferLen), Samples: samples}
-	} else {
-		fmt.Printf("discarding buffer, no pipeline with id %d", int(pipelineID))
-	}
-	C.free(buffer)
-}

examples/gstreamer-send/gst/gst.h

@@ -1,16 +0,0 @@
-#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 pipelineId);
-
-GstElement *gstreamer_send_create_pipeline(char *pipeline);
-void gstreamer_send_start_pipeline(GstElement *pipeline, int pipelineId);
-void gstreamer_send_stop_pipeline(GstElement *pipeline);
-void gstreamer_send_start_mainloop(void);
-
-#endif

examples/gstreamer-send/main.go

@@ -4,8 +4,8 @@ import (
 	"fmt"
 
 	"github.com/pions/webrtc"
-	"github.com/pions/webrtc/examples/gstreamer-send/gst"
 	"github.com/pions/webrtc/examples/util"
+	"github.com/pions/webrtc/examples/util/gstreamer-src"
 	"github.com/pions/webrtc/pkg/ice"
 )
 

examples/janus-gateway/README.md

@@ -19,3 +19,13 @@ Got VP8 track, saving to disk as output.ivf
 ```
 
 You will see output.ivf in the current folder.
+
+## video-room
+This example demonstrates how to stream to a Janus video-room using pion-WebRTC
+
+### Running
+run `main.go` in `github.com/pions/webrtc/examples/janus-gateway/video-room`
+
+If this worked you should see a test video in video-room `1234`
+
+This is the default demo-room that exists in the sample configs, and can quickly be accessed via the Janus demos.

examples/janus-gateway/video-room/go.mod

@@ -0,0 +1,9 @@
+module github.com/pions/webrtc/examples/janus-gateway/streaming
+
+replace github.com/pions/webrtc => ../../../
+
+require (
+	github.com/gorilla/websocket v1.4.0 // indirect
+	github.com/notedit/janus-go v0.0.0-20180821162543-a152adf0cb7b
+	github.com/pions/webrtc v1.1.1
+)

examples/janus-gateway/video-room/go.sum

@@ -0,0 +1,20 @@
+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/google/go-cmp v0.2.0 h1:+dTQ8DZQJz0Mb/HjFlkptS1FeQ4cWSnN941F8aEG4SQ=
+github.com/google/go-cmp v0.2.0/go.mod h1:oXzfMopK8JAjlY9xF4vHSVASa0yLyX7SntLO5aqRK0M=
+github.com/gorilla/websocket v1.4.0 h1:WDFjx/TMzVgy9VdMMQi2K2Emtwi2QcUQsztZ/zLaH/Q=
+github.com/gorilla/websocket v1.4.0/go.mod h1:E7qHFY5m1UJ88s3WnNqhKjPHQ0heANvMoAMk2YaljkQ=
+github.com/notedit/janus-go v0.0.0-20180821162543-a152adf0cb7b h1:GT1/zfKpQHX4Cz7F1QUE/tjE/OP0KM5aYaFiKVRgvkk=
+github.com/notedit/janus-go v0.0.0-20180821162543-a152adf0cb7b/go.mod h1:BN/Txse3qz8tZOmCm2OfajB2wHVujWmX3o9nVdsI6gE=
+github.com/pions/pkg v0.0.0-20181115215726-b60cd756f712 h1:ciXO7F7PusyAzW/EZJt01bETgfTxP/BIGoWQ15pBP54=
+github.com/pions/pkg v0.0.0-20181115215726-b60cd756f712/go.mod h1:r9wKZs+Xxv2acLspex4CHQiIhFjGK1zGP+nUm/8klXA=
+github.com/pkg/errors v0.8.0 h1:WdK/asTD0HN+q6hsWO3/vpuAkAr+tw6aNJNDFFf0+qw=
+github.com/pkg/errors v0.8.0/go.mod h1:bwawxfHBFNV+L2hUp1rHADufV3IMtnDRdf1r5NINEl0=
+github.com/pmezard/go-difflib v1.0.0 h1:4DBwDE0NGyQoBHbLQYPwSUPoCMWR5BEzIk/f1lZbAQM=
+github.com/pmezard/go-difflib v1.0.0/go.mod h1:iKH77koFhYxTK1pcRnkKkqfTogsbg7gZNVY4sRDYZ/4=
+github.com/stretchr/testify v1.2.2 h1:bSDNvY7ZPG5RlJ8otE/7V6gMiyenm9RtJ7IUVIAoJ1w=
+github.com/stretchr/testify v1.2.2/go.mod h1:a8OnRcib4nhh0OaRAV+Yts87kKdq0PP7pXfy6kDkUVs=
+golang.org/x/net v0.0.0-20181129055619-fae4c4e3ad76 h1:xx5MUFyRQRbPk6VjWjIE1epE/K5AoDD8QUN116NCy8k=
+golang.org/x/net v0.0.0-20181129055619-fae4c4e3ad76/go.mod h1:mL1N/T3taQHkDXs73rZJwtUhF3w3ftmwwsq0BUmARs4=
+gotest.tools v2.2.0+incompatible h1:y0IMTfclpMdsdIbr6uwmJn5/WZ7vFuObxDMdrylFM3A=
+gotest.tools v2.2.0+incompatible/go.mod h1:DsYFclhRJ6vuDpmuTbkuFWG+y2sxOXAzmJt81HFBacw=

examples/janus-gateway/video-room/main.go

@@ -0,0 +1,119 @@
+package main
+
+import (
+	"fmt"
+	"log"
+
+	janus "github.com/notedit/janus-go"
+	"github.com/pions/webrtc"
+	"github.com/pions/webrtc/examples/util"
+	"github.com/pions/webrtc/examples/util/gstreamer-src"
+	"github.com/pions/webrtc/pkg/ice"
+)
+
+func watchHandle(handle *janus.Handle) {
+	// wait for event
+	for {
+		msg := <-handle.Events
+		switch msg := msg.(type) {
+		case *janus.SlowLinkMsg:
+			log.Println("SlowLinkMsg type ", handle.Id)
+		case *janus.MediaMsg:
+			log.Println("MediaEvent type", msg.Type, " receiving ", msg.Receiving)
+		case *janus.WebRTCUpMsg:
+			log.Println("WebRTCUp type ", handle.Id)
+		case *janus.HangupMsg:
+			log.Println("HangupEvent type ", handle.Id)
+		case *janus.EventMsg:
+			log.Printf("EventMsg %+v", msg.Plugindata.Data)
+		}
+
+	}
+
+}
+
+func main() {
+	// Everything below is the pion-WebRTC API! Thanks for using it ❤️.
+
+	// Setup the codecs you want to use.
+	// We'll use the default ones but you can also define your own
+	webrtc.RegisterDefaultCodecs()
+
+	// Prepare the configuration
+	config := webrtc.RTCConfiguration{
+		IceServers: []webrtc.RTCIceServer{
+			{
+				URLs: []string{"stun:stun.l.google.com:19302"},
+			},
+		},
+	}
+
+	// Create a new RTCPeerConnection
+	peerConnection, err := webrtc.New(config)
+	util.Check(err)
+
+	peerConnection.OnICEConnectionStateChange(func(connectionState ice.ConnectionState) {
+		fmt.Printf("Connection State has changed %s \n", connectionState.String())
+	})
+
+	// Create a audio track
+	opusTrack, err := peerConnection.NewRTCSampleTrack(webrtc.DefaultPayloadTypeOpus, "audio", "pion1")
+	util.Check(err)
+	_, err = peerConnection.AddTrack(opusTrack)
+	util.Check(err)
+
+	// Create a video track
+	vp8Track, err := peerConnection.NewRTCSampleTrack(webrtc.DefaultPayloadTypeVP8, "video", "pion2")
+	util.Check(err)
+	_, err = peerConnection.AddTrack(vp8Track)
+	util.Check(err)
+
+	offer, err := peerConnection.CreateOffer(nil)
+	util.Check(err)
+
+	gateway, err := janus.Connect("ws://localhost:8188/janus")
+	util.Check(err)
+
+	session, err := gateway.Create()
+	util.Check(err)
+
+	handle, err := session.Attach("janus.plugin.videoroom")
+	util.Check(err)
+
+	go watchHandle(handle)
+
+	_, err = handle.Message(map[string]interface{}{
+		"request": "join",
+		"ptype":   "publisher",
+		"room":    1234,
+		"id":      1,
+	}, nil)
+	util.Check(err)
+
+	msg, err := handle.Message(map[string]interface{}{
+		"request": "publish",
+		"audio":   true,
+		"video":   true,
+		"data":    false,
+	}, map[string]interface{}{
+		"type":    "offer",
+		"sdp":     offer.Sdp,
+		"trickle": false,
+	})
+	util.Check(err)
+
+	if msg.Jsep != nil {
+		err = peerConnection.SetRemoteDescription(webrtc.RTCSessionDescription{
+			Type: webrtc.RTCSdpTypeAnswer,
+			Sdp:  msg.Jsep["sdp"].(string),
+		})
+		util.Check(err)
+
+		// Start pushing buffers on these tracks
+		gst.CreatePipeline(webrtc.Opus, opusTrack.Samples).Start()
+		gst.CreatePipeline(webrtc.VP8, vp8Track.Samples).Start()
+	}
+
+	select {}
+
+}