Examples: Make examples/util internal

Resolves #424

examples/internal/gstreamer-src/gst.go

@@ -0,0 +1,116 @@
+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.Sample
+	// stop acts as a signal that this pipeline is stopped
+	// any pending sends to Pipeline.in should be cancelled
+	stop      chan interface{}
+	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.Sample, pipelineSrc string) *Pipeline {
+	pipelineStr := "appsink name=appsink"
+	switch codecName {
+	case webrtc.VP8:
+		pipelineStr = pipelineSrc + " ! vp8enc error-resilient=partitions keyframe-max-dist=10 auto-alt-ref=true cpu-used=5 deadline=1 ! " + pipelineStr
+	case webrtc.VP9:
+		pipelineStr = pipelineSrc + " ! vp9enc ! " + pipelineStr
+	case webrtc.H264:
+		pipelineStr = pipelineSrc + " ! 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 = pipelineSrc + " ! opusenc ! " + pipelineStr
+	case webrtc.G722:
+		pipelineStr = pipelineSrc + " ! avenc_g722 ! " + 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() {
+	// This will signal to goHandlePipelineBuffer
+	// and provide a method for cancelling sends.
+	p.stop = make(chan interface{})
+	C.gstreamer_send_start_pipeline(p.Pipeline, C.int(p.id))
+}
+
+// Stop stops the GStreamer Pipeline
+func (p *Pipeline) Stop() {
+	// To run gstreamer_send_stop_pipeline we need to make sure
+	// that appsink isn't being hung by any goHandlePipelineBuffers
+	close(p.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()
+	pipeline, ok := pipelines[int(pipelineID)]
+	pipelinesLock.Unlock()
+
+	if ok {
+		var samples uint32
+		if pipeline.codecName == webrtc.Opus {
+			samples = uint32(audioClockRate * (float32(duration) / 1000000000))
+		} else {
+			samples = uint32(videoClockRate * (float32(duration) / 1000000000))
+		}
+		// We need to be able to cancel this function even f pipeline.in isn't being serviced
+		// When pipeline.stop is closed the sending of data will be cancelled.
+		select {
+		case pipeline.in <- media.Sample{Data: C.GoBytes(buffer, bufferLen), Samples: samples}:
+		case <-pipeline.stop:
+		}
+	} else {
+		fmt.Printf("discarding buffer, no pipeline with id %d", int(pipelineID))
+	}
+	C.free(buffer)
+}