mirror of
https://github.com/ossrs/srs.git
synced 2025-02-15 04:42:04 +00:00
292 lines
8.4 KiB
Go
292 lines
8.4 KiB
Go
// The MIT License (MIT)
|
|
//
|
|
// Copyright (c) 2021 Winlin
|
|
//
|
|
// Permission is hereby granted, free of charge, to any person obtaining a copy of
|
|
// this software and associated documentation files (the "Software"), to deal in
|
|
// the Software without restriction, including without limitation the rights to
|
|
// use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of
|
|
// the Software, and to permit persons to whom the Software is furnished to do so,
|
|
// subject to the following conditions:
|
|
//
|
|
// The above copyright notice and this permission notice shall be included in all
|
|
// copies or substantial portions of the Software.
|
|
//
|
|
// THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
|
|
// IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS
|
|
// FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR
|
|
// COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER
|
|
// IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN
|
|
// CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
|
|
package srs
|
|
|
|
import (
|
|
"context"
|
|
"fmt"
|
|
"strings"
|
|
"sync"
|
|
"time"
|
|
|
|
"github.com/ossrs/go-oryx-lib/errors"
|
|
"github.com/ossrs/go-oryx-lib/logger"
|
|
"github.com/pion/interceptor"
|
|
"github.com/pion/rtcp"
|
|
"github.com/pion/sdp/v3"
|
|
"github.com/pion/webrtc/v3"
|
|
"github.com/pion/webrtc/v3/pkg/media"
|
|
"github.com/pion/webrtc/v3/pkg/media/h264writer"
|
|
"github.com/pion/webrtc/v3/pkg/media/ivfwriter"
|
|
"github.com/pion/webrtc/v3/pkg/media/oggwriter"
|
|
)
|
|
|
|
// @see https://github.com/pion/webrtc/blob/master/examples/save-to-disk/main.go
|
|
func startPlay(ctx context.Context, r, dumpAudio, dumpVideo string, enableAudioLevel, enableTWCC bool, pli int) error {
|
|
ctx = logger.WithContext(ctx)
|
|
|
|
logger.Tf(ctx, "Run play url=%v, audio=%v, video=%v, audio-level=%v, twcc=%v",
|
|
r, dumpAudio, dumpVideo, enableAudioLevel, enableTWCC)
|
|
|
|
// For audio-level.
|
|
webrtcNewPeerConnection := func(configuration webrtc.Configuration) (*webrtc.PeerConnection, error) {
|
|
m := &webrtc.MediaEngine{}
|
|
if err := m.RegisterDefaultCodecs(); err != nil {
|
|
return nil, err
|
|
}
|
|
|
|
for _, extension := range []string{sdp.SDESMidURI, sdp.SDESRTPStreamIDURI, sdp.TransportCCURI} {
|
|
if extension == sdp.TransportCCURI && !enableTWCC {
|
|
continue
|
|
}
|
|
if err := m.RegisterHeaderExtension(webrtc.RTPHeaderExtensionCapability{URI: extension}, webrtc.RTPCodecTypeVideo); err != nil {
|
|
return nil, err
|
|
}
|
|
}
|
|
|
|
// https://github.com/pion/ion/issues/130
|
|
// https://github.com/pion/ion-sfu/pull/373/files#diff-6f42c5ac6f8192dd03e5a17e9d109e90cb76b1a4a7973be6ce44a89ffd1b5d18R73
|
|
for _, extension := range []string{sdp.SDESMidURI, sdp.SDESRTPStreamIDURI, sdp.AudioLevelURI} {
|
|
if extension == sdp.AudioLevelURI && !enableAudioLevel {
|
|
continue
|
|
}
|
|
if err := m.RegisterHeaderExtension(webrtc.RTPHeaderExtensionCapability{URI: extension}, webrtc.RTPCodecTypeAudio); err != nil {
|
|
return nil, err
|
|
}
|
|
}
|
|
|
|
i := &interceptor.Registry{}
|
|
if err := webrtc.RegisterDefaultInterceptors(m, i); err != nil {
|
|
return nil, err
|
|
}
|
|
|
|
api := webrtc.NewAPI(webrtc.WithMediaEngine(m), webrtc.WithInterceptorRegistry(i))
|
|
return api.NewPeerConnection(configuration)
|
|
}
|
|
|
|
pc, err := webrtcNewPeerConnection(webrtc.Configuration{})
|
|
if err != nil {
|
|
return errors.Wrapf(err, "Create PC")
|
|
}
|
|
|
|
var receivers []*webrtc.RTPReceiver
|
|
defer func() {
|
|
pc.Close()
|
|
for _, receiver := range receivers {
|
|
receiver.Stop()
|
|
}
|
|
}()
|
|
|
|
pc.AddTransceiverFromKind(webrtc.RTPCodecTypeAudio, webrtc.RTPTransceiverInit{
|
|
Direction: webrtc.RTPTransceiverDirectionRecvonly,
|
|
})
|
|
pc.AddTransceiverFromKind(webrtc.RTPCodecTypeVideo, webrtc.RTPTransceiverInit{
|
|
Direction: webrtc.RTPTransceiverDirectionRecvonly,
|
|
})
|
|
|
|
offer, err := pc.CreateOffer(nil)
|
|
if err != nil {
|
|
return errors.Wrapf(err, "Create Offer")
|
|
}
|
|
|
|
if err := pc.SetLocalDescription(offer); err != nil {
|
|
return errors.Wrapf(err, "Set offer %v", offer)
|
|
}
|
|
|
|
answer, err := apiRtcRequest(ctx, "/rtc/v1/play", r, offer.SDP)
|
|
if err != nil {
|
|
return errors.Wrapf(err, "Api request offer=%v", offer.SDP)
|
|
}
|
|
|
|
if err := pc.SetRemoteDescription(webrtc.SessionDescription{
|
|
Type: webrtc.SDPTypeAnswer, SDP: answer,
|
|
}); err != nil {
|
|
return errors.Wrapf(err, "Set answer %v", answer)
|
|
}
|
|
|
|
var da media.Writer
|
|
var dv_vp8 media.Writer
|
|
var dv_h264 media.Writer
|
|
defer func() {
|
|
if da != nil {
|
|
da.Close()
|
|
}
|
|
if dv_vp8 != nil {
|
|
dv_vp8.Close()
|
|
}
|
|
if dv_h264 != nil {
|
|
dv_h264.Close()
|
|
}
|
|
}()
|
|
|
|
handleTrack := func(ctx context.Context, track *webrtc.TrackRemote, receiver *webrtc.RTPReceiver) error {
|
|
// Send a PLI on an interval so that the publisher is pushing a keyframe
|
|
go func() {
|
|
if track.Kind() == webrtc.RTPCodecTypeAudio {
|
|
return
|
|
}
|
|
|
|
if pli <= 0 {
|
|
return
|
|
}
|
|
|
|
for {
|
|
select {
|
|
case <-ctx.Done():
|
|
return
|
|
case <-time.After(time.Duration(pli) * time.Second):
|
|
_ = pc.WriteRTCP([]rtcp.Packet{&rtcp.PictureLossIndication{
|
|
MediaSSRC: uint32(track.SSRC()),
|
|
}})
|
|
}
|
|
}
|
|
}()
|
|
|
|
receivers = append(receivers, receiver)
|
|
|
|
codec := track.Codec()
|
|
|
|
trackDesc := fmt.Sprintf("channels=%v", codec.Channels)
|
|
if track.Kind() == webrtc.RTPCodecTypeVideo {
|
|
trackDesc = fmt.Sprintf("fmtp=%v", codec.SDPFmtpLine)
|
|
}
|
|
if headers := receiver.GetParameters().HeaderExtensions; len(headers) > 0 {
|
|
trackDesc = fmt.Sprintf("%v, header=%v", trackDesc, headers)
|
|
}
|
|
logger.Tf(ctx, "Got track %v, pt=%v, tbn=%v, %v",
|
|
codec.MimeType, codec.PayloadType, codec.ClockRate, trackDesc)
|
|
|
|
if codec.MimeType == "audio/opus" {
|
|
if da == nil && dumpAudio != "" {
|
|
if da, err = oggwriter.New(dumpAudio, codec.ClockRate, codec.Channels); err != nil {
|
|
return errors.Wrapf(err, "New audio dumper")
|
|
}
|
|
logger.Tf(ctx, "Open ogg writer file=%v, tbn=%v, channels=%v",
|
|
dumpAudio, codec.ClockRate, codec.Channels)
|
|
}
|
|
|
|
if err = writeTrackToDisk(ctx, da, track); err != nil {
|
|
return errors.Wrapf(err, "Write audio disk")
|
|
}
|
|
} else if codec.MimeType == "video/VP8" {
|
|
if dumpVideo != "" && !strings.HasSuffix(dumpVideo, ".ivf") {
|
|
return errors.Errorf("%v should be .ivf for VP8", dumpVideo)
|
|
}
|
|
|
|
if dv_vp8 == nil && dumpVideo != "" {
|
|
if dv_vp8, err = ivfwriter.New(dumpVideo); err != nil {
|
|
return errors.Wrapf(err, "New video dumper")
|
|
}
|
|
logger.Tf(ctx, "Open ivf writer file=%v", dumpVideo)
|
|
}
|
|
|
|
if err = writeTrackToDisk(ctx, dv_vp8, track); err != nil {
|
|
return errors.Wrapf(err, "Write video disk")
|
|
}
|
|
} else if codec.MimeType == "video/H264" {
|
|
if dumpVideo != "" && !strings.HasSuffix(dumpVideo, ".h264") {
|
|
return errors.Errorf("%v should be .h264 for H264", dumpVideo)
|
|
}
|
|
|
|
if dv_h264 == nil && dumpVideo != "" {
|
|
if dv_h264, err = h264writer.New(dumpVideo); err != nil {
|
|
return errors.Wrapf(err, "New video dumper")
|
|
}
|
|
logger.Tf(ctx, "Open h264 writer file=%v", dumpVideo)
|
|
}
|
|
|
|
if err = writeTrackToDisk(ctx, dv_h264, track); err != nil {
|
|
return errors.Wrapf(err, "Write video disk")
|
|
}
|
|
} else {
|
|
logger.Wf(ctx, "Ignore track %v pt=%v", codec.MimeType, codec.PayloadType)
|
|
}
|
|
return nil
|
|
}
|
|
|
|
ctx, cancel := context.WithCancel(ctx)
|
|
pc.OnTrack(func(track *webrtc.TrackRemote, receiver *webrtc.RTPReceiver) {
|
|
err = handleTrack(ctx, track, receiver)
|
|
if err != nil {
|
|
codec := track.Codec()
|
|
err = errors.Wrapf(err, "Handle track %v, pt=%v", codec.MimeType, codec.PayloadType)
|
|
cancel()
|
|
}
|
|
})
|
|
|
|
pc.OnICEConnectionStateChange(func(state webrtc.ICEConnectionState) {
|
|
logger.If(ctx, "ICE state %v", state)
|
|
|
|
if state == webrtc.ICEConnectionStateFailed || state == webrtc.ICEConnectionStateClosed {
|
|
if ctx.Err() != nil {
|
|
return
|
|
}
|
|
|
|
logger.Wf(ctx, "Close for ICE state %v", state)
|
|
cancel()
|
|
}
|
|
})
|
|
|
|
// Wait for event from context or tracks.
|
|
var wg sync.WaitGroup
|
|
|
|
wg.Add(1)
|
|
go func() {
|
|
defer wg.Done()
|
|
|
|
for {
|
|
select {
|
|
case <-ctx.Done():
|
|
return
|
|
case <-time.After(5 * time.Second):
|
|
gStatRTC.PeerConnection = pc.GetStats()
|
|
}
|
|
}
|
|
}()
|
|
|
|
wg.Wait()
|
|
return err
|
|
}
|
|
|
|
func writeTrackToDisk(ctx context.Context, w media.Writer, track *webrtc.TrackRemote) error {
|
|
for ctx.Err() == nil {
|
|
pkt, _, err := track.ReadRTP()
|
|
if err != nil {
|
|
if ctx.Err() != nil {
|
|
return nil
|
|
}
|
|
return errors.Wrapf(err, "Read RTP")
|
|
}
|
|
|
|
if w == nil {
|
|
continue
|
|
}
|
|
|
|
if err := w.WriteRTP(pkt); err != nil {
|
|
if len(pkt.Payload) <= 2 {
|
|
continue
|
|
}
|
|
logger.Wf(ctx, "Ignore write RTP %vB err %+v", len(pkt.Payload), err)
|
|
}
|
|
}
|
|
|
|
return ctx.Err()
|
|
}
|