1 files changed, 96 insertions, 0 deletions
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)
+}