Files
webrtc/examples/util/gstreamer-src/gst.go
Woodrow Douglass c1bfcc8367 Tweak g722 support code
* allow GST examples to use g722
* Assume G722 is mono -- that's what chrome likes,
  and i don't know if anyone's doing stereo g722 anyway

Relates to #283
2018-12-18 08:33:19 -05:00

102 lines
2.5 KiB
Go

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
case webrtc.G722:
pipelineStr = "audiotestsrc ! 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() {
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)
}