Implement sample count generation in gstreamer-send

Audio+Video now works in gstreamer-send

Author: Sean-Der

examples/gstreamer-receive/gst/gst.c

@@ -43,11 +43,8 @@ GstElement *gstreamer_recieve_create_pipeline(char *pipeline) {
 }
 
 void gstreamer_recieve_start_pipeline(GstElement *pipeline) {
-  GstBus *bus;
-  guint bus_watch_id;
-
-  bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline));
-  bus_watch_id = gst_bus_add_watch(bus, gstreamer_recieve_bus_call, NULL);
+  GstBus *bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline));
+  gst_bus_add_watch(bus, gstreamer_recieve_bus_call, NULL);
   gst_object_unref(bus);
 
   gst_element_set_state(pipeline, GST_STATE_PLAYING);

examples/gstreamer-send/gst/gst.c

@@ -52,7 +52,7 @@ GstFlowReturn gstreamer_send_new_sample_handler(GstElement *object, gpointer use
     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, 0, s->pipelineId);
+      goHandlePipelineBuffer(copy, copy_size, GST_BUFFER_DURATION(buffer), s->pipelineId);
     }
     gst_sample_unref (sample);
   }

examples/gstreamer-send/gst/gst.go

@@ -24,6 +24,7 @@ type Pipeline struct {
 	Pipeline *C.GstElement
 	in       chan<- webrtc.RTCSample
 	id       int
+	codec    webrtc.TrackType
 }
 
 var pipelines = make(map[int]*Pipeline)
@@ -53,6 +54,7 @@ func CreatePipeline(codec webrtc.TrackType, in chan<- webrtc.RTCSample) *Pipelin
 		Pipeline: C.gstreamer_send_create_pipeline(pipelineStrUnsafe),
 		in:       in,
 		id:       len(pipelines),
+		codec:    codec,
 	}
 
 	pipelines[pipeline.id] = pipeline
@@ -69,13 +71,24 @@ 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, samples C.int, pipelineId C.int) {
+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 {
-		pipeline.in <- webrtc.RTCSample{C.GoBytes(buffer, bufferLen), uint32(samples)}
+		var samples uint32
+		if pipeline.codec == webrtc.Opus {
+			samples = uint32(audioClockRate * (float32(duration) / 1000000000))
+		} else {
+			samples = uint32(videoClockRate * (float32(duration) / 1000000000))
+		}
+		pipeline.in <- webrtc.RTCSample{C.GoBytes(buffer, bufferLen), samples}
 	} else {
 		fmt.Printf("discarding buffer, no pipeline with id %d", int(pipelineId))
 	}