fixes

loadtest/client/gst/compositor.go

@@ -32,7 +32,7 @@ type CompositorPipeline struct {
 func NewCompositorPipeline(extraPipelineStr string) *CompositorPipeline {
 	pipelineStr := `
 		compositor name=vmix background=black ! video/x-raw,width=1920,height=1080,framerate=30/1,format=UYVY ! queue ! tee name=vtee 
-			vtee. ! queue ! autovideosink sync=false 
+			vtee. ! queue ! glimagesink sync=false 
 		audiomixer name=amix ! queue ! tee name=atee 
 			atee. ! queue ! audioconvert ! autoaudiosink
 	` + extraPipelineStr

loadtest/client/producer.go

@@ -30,12 +30,12 @@ type GSTProducer struct {
 // NewGSTProducer will create a new producer for a given client and a videoFile
 func NewGSTProducer(kind string, path string) *GSTProducer {
 	stream := fmt.Sprintf("gst-%v-%v", kind, cuid.New())
-	videoTrack, err := webrtc.NewTrackLocalStaticSample(webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeH264}, cuid.New(), stream)
+	videoTrack, err := webrtc.NewTrackLocalStaticSample(webrtc.RTPCodecCapability{MimeType: "video/h264", ClockRate: 90000}, cuid.New(), stream)
 	if err != nil {
 		panic(err)
 	}
 
-	audioTrack, err := webrtc.NewTrackLocalStaticSample(webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeOpus}, cuid.New(), stream)
+	audioTrack, err := webrtc.NewTrackLocalStaticSample(webrtc.RTPCodecCapability{MimeType: "audio/opus", ClockRate: 48000}, cuid.New(), stream)
 	if err != nil {
 		panic(err)
 	}
@@ -83,7 +83,6 @@ func (t *GSTProducer) Pause(pause bool) {
 
 //Stop the pipeline
 func (t *GSTProducer) Stop() {
-	t.pipeline.Pause()
 }
 
 //Start the pipeline

loadtest/gstreamer-src/gst.c

@@ -0,0 +1,86 @@
+#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);
+  }
+  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));
+  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_object_unref(appsink);
+
+  gst_element_set_state(pipeline, GST_STATE_PLAYING);
+}
+
+void gstreamer_send_stop_pipeline(GstElement *pipeline) {
+  gst_element_set_state(pipeline, GST_STATE_NULL);
+}

loadtest/gstreamer-src/gst.go

@@ -0,0 +1,123 @@
+// Package gst provides an easy API to create an appsink pipeline
+package gstreamergst
+
+/*
+#cgo pkg-config: gstreamer-1.0 gstreamer-app-1.0
+#include "gst.h"
+*/
+import "C"
+import (
+	"fmt"
+	"sync"
+	"time"
+	"unsafe"
+
+	"github.com/pion/webrtc/v3"
+	"github.com/pion/webrtc/v3/pkg/media"
+)
+
+func init() {
+	go C.gstreamer_send_start_mainloop()
+}
+
+// Pipeline is a wrapper for a GStreamer Pipeline
+type Pipeline struct {
+	Pipeline  *C.GstElement
+	tracks    []*webrtc.TrackLocalStaticSample
+	id        int
+	codecName string
+	clockRate float32
+}
+
+var pipelines = make(map[int]*Pipeline)
+var pipelinesLock sync.Mutex
+
+const (
+	videoClockRate = 90000
+	audioClockRate = 48000
+	pcmClockRate   = 8000
+)
+
+// CreatePipeline creates a GStreamer Pipeline
+func CreatePipeline(codecName string, tracks []*webrtc.TrackLocalStaticSample, pipelineSrc string) *Pipeline {
+	pipelineStr := "appsink name=appsink"
+	var clockRate float32
+
+	switch codecName {
+	case "vp8":
+		pipelineStr = pipelineSrc + " ! vp8enc error-resilient=partitions keyframe-max-dist=10 auto-alt-ref=true cpu-used=5 deadline=1 ! " + pipelineStr
+		clockRate = videoClockRate
+
+	case "vp9":
+		pipelineStr = pipelineSrc + " ! vp9enc ! " + pipelineStr
+		clockRate = videoClockRate
+
+	case "h264":
+		pipelineStr = pipelineSrc + " ! video/x-raw,format=I420 ! x264enc speed-preset=ultrafast tune=zerolatency key-int-max=20 ! video/x-h264,stream-format=byte-stream ! " + pipelineStr
+		clockRate = videoClockRate
+
+	case "opus":
+		pipelineStr = pipelineSrc + " ! opusenc ! " + pipelineStr
+		clockRate = audioClockRate
+
+	case "g722":
+		pipelineStr = pipelineSrc + " ! avenc_g722 ! " + pipelineStr
+		clockRate = audioClockRate
+
+	case "pcmu":
+		pipelineStr = pipelineSrc + " ! audio/x-raw, rate=8000 ! mulawenc ! " + pipelineStr
+		clockRate = pcmClockRate
+
+	case "pcma":
+		pipelineStr = pipelineSrc + " ! audio/x-raw, rate=8000 ! alawenc ! " + pipelineStr
+		clockRate = pcmClockRate
+
+	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),
+		tracks:    tracks,
+		id:        len(pipelines),
+		codecName: codecName,
+		clockRate: clockRate,
+	}
+
+	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)
+}
+
+//export goHandlePipelineBuffer
+func goHandlePipelineBuffer(buffer unsafe.Pointer, bufferLen C.int, duration C.int, pipelineID C.int) {
+	pipelinesLock.Lock()
+	pipeline, ok := pipelines[int(pipelineID)]
+	pipelinesLock.Unlock()
+
+	if ok {
+		for _, t := range pipeline.tracks {
+			if err := t.WriteSample(media.Sample{Data: C.GoBytes(buffer, bufferLen), Duration: time.Duration(duration)}); err != nil {
+				panic(err)
+			}
+		}
+	} else {
+		fmt.Printf("discarding buffer, no pipeline with id %d", int(pipelineID))
+	}
+	C.free(buffer)
+}

loadtest/gstreamer-src/gst.h

@@ -0,0 +1,16 @@
+// #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

loadtest/load.go

@@ -53,10 +53,9 @@ func getFileByType(file string) string {
 	if file == "h264" {
 		//TODO not working as of now need to debug
 		// load is there but doesn't play on browser. track should play on browser also
-		// issue is on browser side
-		filepath = "/var/tmp/Big_Buck_Bunny_720_10s_1MB.mp4"
+		filepath = "/var/tmp/Jellyfish_360_10s_1MB.mp4"
 		if _, err := os.Stat(filepath); os.IsNotExist(err) {
-			err := util.DownloadFile(filepath, "https://test-videos.co.uk/vids/bigbuckbunny/mp4/h264/720/Big_Buck_Bunny_720_10s_1MB.mp4")
+			err := util.DownloadFile(filepath, "https://test-videos.co.uk/vids/jellyfish/mp4/h264/360/Jellyfish_360_10s_1MB.mp4")
 			if err != nil {
 				log.Infof("error downloading file %v", err)
 				filepath = "test"
@@ -139,6 +138,57 @@ func run(e *sdk.Engine, addr, session, file, role string, total, duration, cycle
 				util.HandleDataChannel(c, "loadtest", i, cid)
 
 				if !strings.Contains(file, ".webm") {
+
+					// if file == "test2" {
+
+					// 	audioSrc := "audiotestsrc"
+					// 	videoSrc := "videotestsrc"
+
+					// 	videoTrack, err := webrtc.NewTrackLocalStaticSample(webrtc.RTPCodecCapability{MimeType: "video/vp8"}, "video", "pion2")
+					// 	if err != nil {
+					// 		panic(err)
+					// 	}
+
+					// 	// _, err = peerConnection.AddTrack(videoTrack)
+					// 	// if err != nil {
+					// 	// 	panic(err)
+					// 	// }
+
+					// 	audioTrack, err := webrtc.NewTrackLocalStaticSample(webrtc.RTPCodecCapability{MimeType: "audio/opus"}, "audio", "pion1")
+					// 	if err != nil {
+					// 		panic(err)
+					// 	}
+					// 	// _, err = peerConnection.AddTrack(audioTrack)
+					// 	// if err != nil {
+					// 	// 	panic(err)
+					// 	// }
+
+					// 	// client join a session
+					// 	c.Join(new_session, nil)
+					// 	defer e.DelClient(c)
+
+					// 	t, _ := c.GetPubTransport().GetPeerConnection().AddTransceiverFromTrack(audioTrack, webrtc.RTPTransceiverInit{
+					// 		Direction: webrtc.RTPTransceiverDirectionSendonly,
+					// 	})
+					// 	defer c.UnPublish(t)
+
+					// 	t2, _ := c.GetPubTransport().GetPeerConnection().AddTransceiverFromTrack(videoTrack, webrtc.RTPTransceiverInit{
+					// 		Direction: webrtc.RTPTransceiverDirectionSendonly,
+					// 	})
+					// 	defer c.UnPublish(t2)
+					// 	c.OnNegotiationNeeded()
+
+					// 	if err != nil {
+					// 		log.Errorf("join err=%v", err)
+					// 		panic(err)
+					// 	}
+
+					// 	// Start pushing buffers on these tracks
+					// 	gstreamergst.CreatePipeline("opus", []*webrtc.TrackLocalStaticSample{audioTrack}, audioSrc).Start()
+					// 	gstreamergst.CreatePipeline("vp8", []*webrtc.TrackLocalStaticSample{videoTrack}, videoSrc).Start()
+
+					// } else {
+
 					//this stopped working. need to debug
 					//i.e video is not showing on webapp
 					c.Join(new_session, nil)
@@ -162,6 +212,26 @@ func run(e *sdk.Engine, addr, session, file, role string, total, duration, cycle
 					defer c.UnPublish(t2)
 					c.OnNegotiationNeeded()
 
+					go func() {
+						rtcpBuf := make([]byte, 1500)
+						for {
+							if _, _, rtcpErr := t2.Sender().Read(rtcpBuf); rtcpErr != nil {
+								log.Infof("videoSender rtcp error %v", err)
+								return
+							}
+						}
+					}()
+
+					go func() {
+						rtcpBuf := make([]byte, 1500)
+						for {
+							if _, _, rtcpErr := t.Sender().Read(rtcpBuf); rtcpErr != nil {
+								log.Infof("audioSender rtcp error %v", err)
+								return
+							}
+						}
+					}()
+
 					go func() {
 						ticker := time.NewTicker(3 * time.Second)
 						defer ticker.Stop()
@@ -180,14 +250,15 @@ func run(e *sdk.Engine, addr, session, file, role string, total, duration, cycle
 						}
 					}()
 
-					producer.Start()
+					go producer.Start()
 					defer producer.Stop()
 
 					defer func() {
 						log.Infof("closing tracks")
 					}()
 
 					log.Infof("tracks published")
+					// }
 
 				} else {
 					c.Join(new_session, nil)

tracktodisk/cloudwriter.go

@@ -43,6 +43,7 @@ func NewCloudFileWriter(path string) *FileWriter {
 		log.Errorf("Bucket(%v).Create: %v", bucketName, err)
 	}
 
+	util.UpdateMeta(bucketName + "/" + path)
 	w := bucket.Object(path).NewWriter(ctx)
 	fw.wr = w
 	fw.client = client