mirror of
https://github.com/pion/webrtc.git
synced 2025-10-05 07:06:51 +08:00
0e7086d37a
Let's pull off the bandaid! * Reduces studdering: webrtc.RTCTrack -> webrtc.Track * Makes it easier to find types by editor autocomplete * Makes code read more fluently (less repetition) Since we're breaking the API in 2.0, our only chance to do this is now. Relates to #408
117 lines
3.2 KiB
Go
117 lines
3.2 KiB
Go
package gst
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/*
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#cgo pkg-config: gstreamer-1.0 gstreamer-app-1.0
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#include "gst.h"
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*/
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import "C"
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import (
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"fmt"
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"sync"
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"unsafe"
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"github.com/pions/webrtc"
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"github.com/pions/webrtc/pkg/media"
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)
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func init() {
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go C.gstreamer_send_start_mainloop()
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}
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// Pipeline is a wrapper for a GStreamer Pipeline
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type Pipeline struct {
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Pipeline *C.GstElement
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in chan<- media.Sample
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// stop acts as a signal that this pipeline is stopped
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// any pending sends to Pipeline.in should be cancelled
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stop chan interface{}
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id int
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codecName string
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}
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var pipelines = make(map[int]*Pipeline)
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var pipelinesLock sync.Mutex
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// CreatePipeline creates a GStreamer Pipeline
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func CreatePipeline(codecName string, in chan<- media.Sample, pipelineSrc string) *Pipeline {
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pipelineStr := "appsink name=appsink"
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switch codecName {
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case webrtc.VP8:
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pipelineStr = pipelineSrc + " ! vp8enc error-resilient=partitions keyframe-max-dist=10 auto-alt-ref=true cpu-used=5 deadline=1 ! " + pipelineStr
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case webrtc.VP9:
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pipelineStr = pipelineSrc + " ! vp9enc ! " + pipelineStr
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case webrtc.H264:
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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
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case webrtc.Opus:
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pipelineStr = pipelineSrc + " ! opusenc ! " + pipelineStr
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case webrtc.G722:
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pipelineStr = pipelineSrc + " ! avenc_g722 ! " + pipelineStr
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default:
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panic("Unhandled codec " + codecName)
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}
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pipelineStrUnsafe := C.CString(pipelineStr)
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defer C.free(unsafe.Pointer(pipelineStrUnsafe))
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pipelinesLock.Lock()
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defer pipelinesLock.Unlock()
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pipeline := &Pipeline{
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Pipeline: C.gstreamer_send_create_pipeline(pipelineStrUnsafe),
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in: in,
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id: len(pipelines),
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codecName: codecName,
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}
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pipelines[pipeline.id] = pipeline
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return pipeline
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}
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// Start starts the GStreamer Pipeline
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func (p *Pipeline) Start() {
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// This will signal to goHandlePipelineBuffer
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// and provide a method for cancelling sends.
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p.stop = make(chan interface{})
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C.gstreamer_send_start_pipeline(p.Pipeline, C.int(p.id))
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}
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// Stop stops the GStreamer Pipeline
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func (p *Pipeline) Stop() {
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// To run gstreamer_send_stop_pipeline we need to make sure
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// that appsink isn't being hung by any goHandlePipelineBuffers
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close(p.stop)
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C.gstreamer_send_stop_pipeline(p.Pipeline)
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}
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const (
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videoClockRate = 90000
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audioClockRate = 48000
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)
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//export goHandlePipelineBuffer
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func goHandlePipelineBuffer(buffer unsafe.Pointer, bufferLen C.int, duration C.int, pipelineID C.int) {
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pipelinesLock.Lock()
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pipeline, ok := pipelines[int(pipelineID)]
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pipelinesLock.Unlock()
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if ok {
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var samples uint32
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if pipeline.codecName == webrtc.Opus {
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samples = uint32(audioClockRate * (float32(duration) / 1000000000))
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} else {
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samples = uint32(videoClockRate * (float32(duration) / 1000000000))
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}
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// We need to be able to cancel this function even f pipeline.in isn't being serviced
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// When pipeline.stop is closed the sending of data will be cancelled.
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select {
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case pipeline.in <- media.Sample{Data: C.GoBytes(buffer, bufferLen), Samples: samples}:
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case <-pipeline.stop:
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}
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} else {
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fmt.Printf("discarding buffer, no pipeline with id %d", int(pipelineID))
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}
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C.free(buffer)
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}
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