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generate RTP packets after H264 remuxing

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Previously, RTP packets coming from sources other than RTSP (that
actually are RTMP and HLS) were generated before the H264 remuxing, and
that leaded to invalid streams, expecially when sourceOnDemand is true
and the stream has invalid or dynamic SPS/PPS.
This commit is contained in:
aler9
2022-08-14 13:03:04 +02:00
parent 5ba46e6c8f
commit 3606472e82
14 changed files with 397 additions and 281 deletions
Download Patch File Download Diff File Expand all files Collapse all files
internal/core/data.go
+15 -4
View File
@@ -6,10 +6,21 @@ import (
"github.com/pion/rtp"
)
// data is the data unit routed across the server.
// it must contain one or more of the following:
// - a single RTP packet
// - a group of H264 NALUs (grouped by timestamp)
// - a single AAC AU
type data struct {
trackID int
rtp *rtp.Packet
trackID int
rtpPacket *rtp.Packet
// timing
ptsEqualsDTS bool
h264NALUs [][]byte
h264PTS time.Duration
pts time.Duration
h264NALUs [][]byte
mpeg4AudioAU []byte
}
internal/core/hls_muxer.go
+3 -3
View File
@@ -366,17 +366,17 @@ func (m *hlsMuxer) runInner(innerCtx context.Context, innerReady chan struct{})
}
if videoInitialPTS == nil {
v := data.h264PTS
v := data.pts
videoInitialPTS = &v
}
pts := data.h264PTS - *videoInitialPTS
pts := data.pts - *videoInitialPTS
err = m.muxer.WriteH264(pts, data.h264NALUs)
if err != nil {
return fmt.Errorf("muxer error: %v", err)
}
} else if audioTrack != nil && data.trackID == audioTrackID {
aus, pts, err := aacDecoder.Decode(data.rtp)
aus, pts, err := aacDecoder.Decode(data.rtpPacket)
if err != nil {
if err != rtpmpeg4audio.ErrMorePacketsNeeded {
m.log(logger.Warn, "unable to decode audio track: %v", err)
internal/core/hls_source.go
+16 -50
View File
@@ -6,8 +6,6 @@ import (
"github.com/aler9/gortsplib"
"github.com/aler9/gortsplib/pkg/h264"
"github.com/aler9/gortsplib/pkg/rtph264"
"github.com/aler9/gortsplib/pkg/rtpmpeg4audio"
"github.com/aler9/rtsp-simple-server/internal/hls"
"github.com/aler9/rtsp-simple-server/internal/logger"
@@ -46,8 +44,6 @@ func (s *hlsSource) run(ctx context.Context) error {
var stream *stream
var videoTrackID int
var audioTrackID int
var videoEnc *rtph264.Encoder
var audioEnc *rtpmpeg4audio.Encoder
defer func() {
if stream != nil {
@@ -60,25 +56,18 @@ func (s *hlsSource) run(ctx context.Context) error {
if videoTrack != nil {
videoTrackID = len(tracks)
videoEnc = &rtph264.Encoder{PayloadType: 96}
videoEnc.Init()
tracks = append(tracks, videoTrack)
}
if audioTrack != nil {
audioTrackID = len(tracks)
audioEnc = &rtpmpeg4audio.Encoder{
PayloadType: 96,
SampleRate: audioTrack.ClockRate(),
SizeLength: 13,
IndexLength: 3,
IndexDeltaLength: 3,
}
audioEnc.Init()
tracks = append(tracks, audioTrack)
}
res := s.parent.sourceStaticImplSetReady(pathSourceStaticSetReadyReq{tracks: tracks})
res := s.parent.sourceStaticImplSetReady(pathSourceStaticSetReadyReq{
tracks: tracks,
generateRTPPackets: true,
})
if res.err != nil {
return res.err
}
@@ -94,29 +83,12 @@ func (s *hlsSource) run(ctx context.Context) error {
return
}
pkts, err := videoEnc.Encode(nalus, pts)
if err != nil {
return
}
lastPkt := len(pkts) - 1
for i, pkt := range pkts {
if i != lastPkt {
stream.writeData(&data{
trackID: videoTrackID,
rtp: pkt,
ptsEqualsDTS: false,
})
} else {
stream.writeData(&data{
trackID: videoTrackID,
rtp: pkt,
ptsEqualsDTS: h264.IDRPresent(nalus),
h264NALUs: nalus,
h264PTS: pts,
})
}
}
stream.writeData(&data{
trackID: videoTrackID,
ptsEqualsDTS: h264.IDRPresent(nalus),
pts: pts,
h264NALUs: nalus,
})
}
onAudioData := func(pts time.Duration, au []byte) {
@@ -124,18 +96,12 @@ func (s *hlsSource) run(ctx context.Context) error {
return
}
pkts, err := audioEnc.Encode([][]byte{au}, pts)
if err != nil {
return
}
for _, pkt := range pkts {
stream.writeData(&data{
trackID: audioTrackID,
rtp: pkt,
ptsEqualsDTS: true,
})
}
stream.writeData(&data{
trackID: audioTrackID,
ptsEqualsDTS: true,
pts: pts,
mpeg4AudioAU: au,
})
}
c, err := hls.NewClient(
internal/core/hls_source_test.go
+5 -1
View File
@@ -135,7 +135,11 @@ func TestHLSSource(t *testing.T) {
c := gortsplib.Client{
OnPacketRTP: func(ctx *gortsplib.ClientOnPacketRTPCtx) {
require.Equal(t, []byte{0x05}, ctx.Packet.Payload)
require.Equal(t, [][]byte{
{0x07, 0x01, 0x02, 0x03},
{0x08},
{0x05},
}, ctx.H264NALUs)
close(frameRecv)
},
}
internal/core/path.go
+41 -25
View File
@@ -92,8 +92,9 @@ type pathSourceStaticSetReadyRes struct {
}
type pathSourceStaticSetReadyReq struct {
tracks gortsplib.Tracks
res chan pathSourceStaticSetReadyRes
tracks gortsplib.Tracks
generateRTPPackets bool
res chan pathSourceStaticSetReadyRes
}
type pathSourceStaticSetNotReadyReq struct {
@@ -160,9 +161,10 @@ type pathPublisherRecordRes struct {
}
type pathPublisherRecordReq struct {
author publisher
tracks gortsplib.Tracks
res chan pathPublisherRecordRes
author publisher
tracks gortsplib.Tracks
generateRTPPackets bool
res chan pathPublisherRecordRes
}
type pathReaderPauseReq struct {
@@ -431,29 +433,32 @@ func (pa *path) run() {
}
case req := <-pa.chSourceStaticSetReady:
pa.sourceSetReady(req.tracks)
err := pa.sourceSetReady(req.tracks, req.generateRTPPackets)
if err != nil {
req.res <- pathSourceStaticSetReadyRes{err: err}
} else {
if pa.hasOnDemandStaticSource() {
pa.onDemandStaticSourceReadyTimer.Stop()
pa.onDemandStaticSourceReadyTimer = newEmptyTimer()
if pa.hasOnDemandStaticSource() {
pa.onDemandStaticSourceReadyTimer.Stop()
pa.onDemandStaticSourceReadyTimer = newEmptyTimer()
pa.onDemandStaticSourceScheduleClose()
pa.onDemandStaticSourceScheduleClose()
for _, req := range pa.describeRequestsOnHold {
req.res <- pathDescribeRes{
stream: pa.stream,
for _, req := range pa.describeRequestsOnHold {
req.res <- pathDescribeRes{
stream: pa.stream,
}
}
}
pa.describeRequestsOnHold = nil
pa.describeRequestsOnHold = nil
for _, req := range pa.setupPlayRequestsOnHold {
pa.handleReaderSetupPlayPost(req)
for _, req := range pa.setupPlayRequestsOnHold {
pa.handleReaderSetupPlayPost(req)
}
pa.setupPlayRequestsOnHold = nil
}
pa.setupPlayRequestsOnHold = nil
req.res <- pathSourceStaticSetReadyRes{stream: pa.stream}
}
req.res <- pathSourceStaticSetReadyRes{stream: pa.stream}
case req := <-pa.chSourceStaticSetNotReady:
pa.sourceSetNotReady()
@@ -657,9 +662,14 @@ func (pa *path) onDemandPublisherStop() {
}
}
func (pa *path) sourceSetReady(tracks gortsplib.Tracks) {
func (pa *path) sourceSetReady(tracks gortsplib.Tracks, generateRTPPackets bool) error {
stream, err := newStream(tracks, generateRTPPackets)
if err != nil {
return err
}
pa.stream = stream
pa.sourceReady = true
pa.stream = newStream(tracks)
if pa.conf.RunOnReady != "" {
pa.log(logger.Info, "runOnReady command started")
@@ -674,6 +684,8 @@ func (pa *path) sourceSetReady(tracks gortsplib.Tracks) {
}
pa.parent.pathSourceReady(pa)
return nil
}
func (pa *path) sourceSetNotReady() {
@@ -808,9 +820,13 @@ func (pa *path) handlePublisherRecord(req pathPublisherRecordReq) {
return
}
req.author.onPublisherAccepted(len(req.tracks))
err := pa.sourceSetReady(req.tracks, req.generateRTPPackets)
if err != nil {
req.res <- pathPublisherRecordRes{err: err}
return
}
pa.sourceSetReady(req.tracks)
req.author.onPublisherAccepted(len(req.tracks))
if pa.hasOnDemandPublisher() {
pa.onDemandPublisherReadyTimer.Stop()
internal/core/rtmp_conn.go
+24 -79
View File
@@ -14,7 +14,6 @@ import (
"github.com/aler9/gortsplib/pkg/h264"
"github.com/aler9/gortsplib/pkg/mpeg4audio"
"github.com/aler9/gortsplib/pkg/ringbuffer"
"github.com/aler9/gortsplib/pkg/rtph264"
"github.com/aler9/gortsplib/pkg/rtpmpeg4audio"
"github.com/notedit/rtmp/format/flv/flvio"
@@ -346,11 +345,11 @@ func (c *rtmpConn) runRead(ctx context.Context, u *url.URL) error {
// while audio is decoded in this routine:
// we have to sync their PTS.
if videoInitialPTS == nil {
v := data.h264PTS
v := data.pts
videoInitialPTS = &v
}
pts := data.h264PTS - *videoInitialPTS
pts := data.pts - *videoInitialPTS
idrPresent := false
nonIDRPresent := false
@@ -442,7 +441,7 @@ func (c *rtmpConn) runRead(ctx context.Context, u *url.URL) error {
return err
}
} else if audioTrack != nil && data.trackID == audioTrackID {
aus, pts, err := aacDecoder.Decode(data.rtp)
aus, pts, err := aacDecoder.Decode(data.rtpPacket)
if err != nil {
if err != rtpmpeg4audio.ErrMorePacketsNeeded {
c.log(logger.Warn, "unable to decode audio track: %v", err)
@@ -491,24 +490,12 @@ func (c *rtmpConn) runPublish(ctx context.Context, u *url.URL) error {
videoTrackID := -1
audioTrackID := -1
var h264Encoder *rtph264.Encoder
if videoTrack != nil {
h264Encoder = &rtph264.Encoder{PayloadType: 96}
h264Encoder.Init()
videoTrackID = len(tracks)
tracks = append(tracks, videoTrack)
}
var aacEncoder *rtpmpeg4audio.Encoder
if audioTrack != nil {
aacEncoder = &rtpmpeg4audio.Encoder{
PayloadType: 96,
SampleRate: audioTrack.ClockRate(),
SizeLength: 13,
IndexLength: 3,
IndexDeltaLength: 3,
}
aacEncoder.Init()
audioTrackID = len(tracks)
tracks = append(tracks, audioTrack)
}
@@ -550,8 +537,9 @@ func (c *rtmpConn) runPublish(ctx context.Context, u *url.URL) error {
c.nconn.SetWriteDeadline(time.Time{})
rres := c.path.publisherRecord(pathPublisherRecordReq{
author: c,
tracks: tracks,
author: c,
tracks: tracks,
generateRTPPackets: true,
})
if rres.err != nil {
return rres.err
@@ -573,35 +561,17 @@ func (c *rtmpConn) runPublish(ctx context.Context, u *url.URL) error {
return fmt.Errorf("unable to parse H264 config: %v", err)
}
pts := tmsg.DTS + tmsg.PTSDelta
nalus := [][]byte{
conf.SPS,
conf.PPS,
}
pkts, err := h264Encoder.Encode(nalus, pts)
if err != nil {
return fmt.Errorf("error while encoding H264: %v", err)
}
lastPkt := len(pkts) - 1
for i, pkt := range pkts {
if i != lastPkt {
rres.stream.writeData(&data{
trackID: videoTrackID,
rtp: pkt,
ptsEqualsDTS: false,
})
} else {
rres.stream.writeData(&data{
trackID: videoTrackID,
rtp: pkt,
ptsEqualsDTS: false,
h264NALUs: nalus,
h264PTS: pts,
})
}
}
rres.stream.writeData(&data{
trackID: videoTrackID,
ptsEqualsDTS: false,
pts: tmsg.DTS + tmsg.PTSDelta,
h264NALUs: nalus,
})
} else if tmsg.H264Type == flvio.AVC_NALU {
if videoTrack == nil {
return fmt.Errorf("received an H264 packet, but track is not set up")
@@ -632,31 +602,12 @@ func (c *rtmpConn) runPublish(ctx context.Context, u *url.URL) error {
}
}
pts := tmsg.DTS + tmsg.PTSDelta
pkts, err := h264Encoder.Encode(validNALUs, pts)
if err != nil {
return fmt.Errorf("error while encoding H264: %v", err)
}
lastPkt := len(pkts) - 1
for i, pkt := range pkts {
if i != lastPkt {
rres.stream.writeData(&data{
trackID: videoTrackID,
rtp: pkt,
ptsEqualsDTS: false,
})
} else {
rres.stream.writeData(&data{
trackID: videoTrackID,
rtp: pkt,
ptsEqualsDTS: h264.IDRPresent(validNALUs),
h264NALUs: validNALUs,
h264PTS: pts,
})
}
}
rres.stream.writeData(&data{
trackID: videoTrackID,
ptsEqualsDTS: h264.IDRPresent(validNALUs),
pts: tmsg.DTS + tmsg.PTSDelta,
h264NALUs: validNALUs,
})
}
case *message.MsgAudio:
@@ -665,18 +616,12 @@ func (c *rtmpConn) runPublish(ctx context.Context, u *url.URL) error {
return fmt.Errorf("received an AAC packet, but track is not set up")
}
pkts, err := aacEncoder.Encode([][]byte{tmsg.Payload}, tmsg.DTS)
if err != nil {
return fmt.Errorf("error while encoding AAC: %v", err)
}
for _, pkt := range pkts {
rres.stream.writeData(&data{
trackID: audioTrackID,
rtp: pkt,
ptsEqualsDTS: true,
})
}
rres.stream.writeData(&data{
trackID: audioTrackID,
ptsEqualsDTS: true,
pts: tmsg.DTS,
mpeg4AudioAU: tmsg.Payload,
})
}
}
}
internal/core/rtmp_source.go
+16 -52
View File
@@ -9,8 +9,6 @@ import (
"github.com/aler9/gortsplib"
"github.com/aler9/gortsplib/pkg/h264"
"github.com/aler9/gortsplib/pkg/rtph264"
"github.com/aler9/gortsplib/pkg/rtpmpeg4audio"
"github.com/notedit/rtmp/format/flv/flvio"
"github.com/aler9/rtsp-simple-server/internal/conf"
@@ -97,29 +95,20 @@ func (s *rtmpSource) run(ctx context.Context) error {
videoTrackID := -1
audioTrackID := -1
var h264Encoder *rtph264.Encoder
if videoTrack != nil {
h264Encoder = &rtph264.Encoder{PayloadType: 96}
h264Encoder.Init()
videoTrackID = len(tracks)
tracks = append(tracks, videoTrack)
}
var aacEncoder *rtpmpeg4audio.Encoder
if audioTrack != nil {
aacEncoder = &rtpmpeg4audio.Encoder{
PayloadType: 96,
SampleRate: audioTrack.ClockRate(),
SizeLength: 13,
IndexLength: 3,
IndexDeltaLength: 3,
}
aacEncoder.Init()
audioTrackID = len(tracks)
tracks = append(tracks, audioTrack)
}
res := s.parent.sourceStaticImplSetReady(pathSourceStaticSetReadyReq{tracks: tracks})
res := s.parent.sourceStaticImplSetReady(pathSourceStaticSetReadyReq{
tracks: tracks,
generateRTPPackets: true,
})
if res.err != nil {
return res.err
}
@@ -149,31 +138,12 @@ func (s *rtmpSource) run(ctx context.Context) error {
return fmt.Errorf("unable to decode AVCC: %v", err)
}
pts := tmsg.DTS + tmsg.PTSDelta
pkts, err := h264Encoder.Encode(nalus, pts)
if err != nil {
return fmt.Errorf("error while encoding H264: %v", err)
}
lastPkt := len(pkts) - 1
for i, pkt := range pkts {
if i != lastPkt {
res.stream.writeData(&data{
trackID: videoTrackID,
rtp: pkt,
ptsEqualsDTS: false,
})
} else {
res.stream.writeData(&data{
trackID: videoTrackID,
rtp: pkt,
ptsEqualsDTS: h264.IDRPresent(nalus),
h264NALUs: nalus,
h264PTS: pts,
})
}
}
res.stream.writeData(&data{
trackID: videoTrackID,
ptsEqualsDTS: h264.IDRPresent(nalus),
pts: tmsg.DTS + tmsg.PTSDelta,
h264NALUs: nalus,
})
}
case *message.MsgAudio:
@@ -182,18 +152,12 @@ func (s *rtmpSource) run(ctx context.Context) error {
return fmt.Errorf("received an AAC packet, but track is not set up")
}
pkts, err := aacEncoder.Encode([][]byte{tmsg.Payload}, tmsg.DTS)
if err != nil {
return fmt.Errorf("error while encoding AAC: %v", err)
}
for _, pkt := range pkts {
res.stream.writeData(&data{
trackID: audioTrackID,
rtp: pkt,
ptsEqualsDTS: true,
})
}
res.stream.writeData(&data{
trackID: audioTrackID,
ptsEqualsDTS: true,
pts: tmsg.DTS,
mpeg4AudioAU: tmsg.Payload,
})
}
}
}
internal/core/rtsp_session.go
+6 -5
View File
@@ -273,8 +273,9 @@ func (s *rtspSession) onPlay(ctx *gortsplib.ServerHandlerOnPlayCtx) (*base.Respo
// onRecord is called by rtspServer.
func (s *rtspSession) onRecord(ctx *gortsplib.ServerHandlerOnRecordCtx) (*base.Response, error) {
res := s.path.publisherRecord(pathPublisherRecordReq{
author: s,
tracks: s.announcedTracks,
author: s,
tracks: s.announcedTracks,
generateRTPPackets: false,
})
if res.err != nil {
return &base.Response{
@@ -391,15 +392,15 @@ func (s *rtspSession) onPacketRTP(ctx *gortsplib.ServerHandlerOnPacketRTPCtx) {
if ctx.H264NALUs != nil {
s.stream.writeData(&data{
trackID: ctx.TrackID,
rtp: ctx.Packet,
rtpPacket: ctx.Packet,
ptsEqualsDTS: ctx.PTSEqualsDTS,
pts: ctx.H264PTS,
h264NALUs: append([][]byte(nil), ctx.H264NALUs...),
h264PTS: ctx.H264PTS,
})
} else {
s.stream.writeData(&data{
trackID: ctx.TrackID,
rtp: ctx.Packet,
rtpPacket: ctx.Packet,
ptsEqualsDTS: ctx.PTSEqualsDTS,
})
}
internal/core/rtsp_source.go
+7 -4
View File
@@ -125,7 +125,10 @@ func (s *rtspSource) run(ctx context.Context) error {
}
}
res := s.parent.sourceStaticImplSetReady(pathSourceStaticSetReadyReq{tracks: tracks})
res := s.parent.sourceStaticImplSetReady(pathSourceStaticSetReadyReq{
tracks: tracks,
generateRTPPackets: false,
})
if res.err != nil {
return res.err
}
@@ -140,15 +143,15 @@ func (s *rtspSource) run(ctx context.Context) error {
if ctx.H264NALUs != nil {
res.stream.writeData(&data{
trackID: ctx.TrackID,
rtp: ctx.Packet,
rtpPacket: ctx.Packet,
ptsEqualsDTS: ctx.PTSEqualsDTS,
pts: ctx.H264PTS,
h264NALUs: append([][]byte(nil), ctx.H264NALUs...),
h264PTS: ctx.H264PTS,
})
} else {
res.stream.writeData(&data{
trackID: ctx.TrackID,
rtp: ctx.Packet,
rtpPacket: ctx.Packet,
ptsEqualsDTS: ctx.PTSEqualsDTS,
})
}
internal/core/stream.go
+20 -58
View File
@@ -1,11 +1,9 @@
package core
import (
"bytes"
"sync"
"github.com/aler9/gortsplib"
"github.com/aler9/gortsplib/pkg/h264"
)
type streamNonRTSPReadersMap struct {
@@ -37,7 +35,7 @@ func (m *streamNonRTSPReadersMap) remove(r reader) {
delete(m.ma, r)
}
func (m *streamNonRTSPReadersMap) forwardPacketRTP(data *data) {
func (m *streamNonRTSPReadersMap) writeData(data *data) {
m.mutex.RLock()
defer m.mutex.RUnlock()
@@ -49,14 +47,26 @@ func (m *streamNonRTSPReadersMap) forwardPacketRTP(data *data) {
type stream struct {
nonRTSPReaders *streamNonRTSPReadersMap
rtspStream *gortsplib.ServerStream
streamTracks []streamTrack
}
func newStream(tracks gortsplib.Tracks) *stream {
func newStream(tracks gortsplib.Tracks, generateRTPPackets bool) (*stream, error) {
s := &stream{
nonRTSPReaders: newStreamNonRTSPReadersMap(),
rtspStream: gortsplib.NewServerStream(tracks),
}
return s
s.streamTracks = make([]streamTrack, len(s.rtspStream.Tracks()))
for i, track := range s.rtspStream.Tracks() {
var err error
s.streamTracks[i], err = newStreamTrack(track, generateRTPPackets, s.writeDataInner)
if err != nil {
return nil, err
}
}
return s, nil
}
func (s *stream) close() {
@@ -80,62 +90,14 @@ func (s *stream) readerRemove(r reader) {
}
}
func (s *stream) updateH264TrackParameters(h264track *gortsplib.TrackH264, nalus [][]byte) {
for _, nalu := range nalus {
typ := h264.NALUType(nalu[0] & 0x1F)
switch typ {
case h264.NALUTypeSPS:
if !bytes.Equal(nalu, h264track.SafeSPS()) {
h264track.SafeSetSPS(append([]byte(nil), nalu...))
}
case h264.NALUTypePPS:
if !bytes.Equal(nalu, h264track.SafePPS()) {
h264track.SafeSetPPS(append([]byte(nil), nalu...))
}
}
}
}
// remux is needed to
// - fix corrupted streams
// - make streams compatible with all protocols
func (s *stream) remuxH264NALUs(h264track *gortsplib.TrackH264, data *data) {
var filteredNALUs [][]byte //nolint:prealloc
for _, nalu := range data.h264NALUs {
typ := h264.NALUType(nalu[0] & 0x1F)
switch typ {
case h264.NALUTypeSPS, h264.NALUTypePPS:
// remove since they're automatically added before every IDR
continue
case h264.NALUTypeAccessUnitDelimiter:
// remove since it is not needed
continue
case h264.NALUTypeIDR:
// add SPS and PPS before every IDR
filteredNALUs = append(filteredNALUs, h264track.SafeSPS(), h264track.SafePPS())
}
filteredNALUs = append(filteredNALUs, nalu)
}
data.h264NALUs = filteredNALUs
}
func (s *stream) writeData(data *data) {
track := s.rtspStream.Tracks()[data.trackID]
if h264track, ok := track.(*gortsplib.TrackH264); ok {
s.updateH264TrackParameters(h264track, data.h264NALUs)
s.remuxH264NALUs(h264track, data)
}
s.streamTracks[data.trackID].writeData(data)
}
func (s *stream) writeDataInner(data *data) {
// forward to RTSP readers
s.rtspStream.WritePacketRTP(data.trackID, data.rtp, data.ptsEqualsDTS)
s.rtspStream.WritePacketRTP(data.trackID, data.rtpPacket, data.ptsEqualsDTS)
// forward to non-RTSP readers
s.nonRTSPReaders.forwardPacketRTP(data)
s.nonRTSPReaders.writeData(data)
}
internal/core/streamtrack.go
+27
View File
@@ -0,0 +1,27 @@
package core
import (
"fmt"
"github.com/aler9/gortsplib"
)
type streamTrack interface {
writeData(*data)
}
func newStreamTrack(track gortsplib.Track, generateRTPPackets bool, writeDataInner func(*data)) (streamTrack, error) {
switch ttrack := track.(type) {
case *gortsplib.TrackH264:
return newStreamTrackH264(ttrack, generateRTPPackets, writeDataInner), nil
case *gortsplib.TrackMPEG4Audio:
return newStreamTrackMPEG4Audio(ttrack, generateRTPPackets, writeDataInner), nil
default:
if generateRTPPackets {
return nil, fmt.Errorf("we don't know how to generate RTP packets of track %+v", track)
}
return newStreamTrackGeneric(track, writeDataInner), nil
}
}
internal/core/streamtrack_generic.go
+19
View File
@@ -0,0 +1,19 @@
package core
import (
"github.com/aler9/gortsplib"
)
type streamTrackGeneric struct {
writeDataInner func(*data)
}
func newStreamTrackGeneric(track gortsplib.Track, writeDataInner func(*data)) *streamTrackGeneric {
return &streamTrackGeneric{
writeDataInner: writeDataInner,
}
}
func (t *streamTrackGeneric) writeData(data *data) {
t.writeDataInner(data)
}
internal/core/streamtrack_h264.go
+140
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@@ -0,0 +1,140 @@
package core
import (
"bytes"
"github.com/aler9/gortsplib"
"github.com/aler9/gortsplib/pkg/h264"
"github.com/aler9/gortsplib/pkg/rtph264"
)
type streamTrackH264 struct {
track *gortsplib.TrackH264
writeDataInner func(*data)
rtpEncoder *rtph264.Encoder
}
func newStreamTrackH264(
track *gortsplib.TrackH264,
generateRTPPackets bool,
writeDataInner func(*data),
) *streamTrackH264 {
t := &streamTrackH264{
track: track,
writeDataInner: writeDataInner,
}
if generateRTPPackets {
t.rtpEncoder = &rtph264.Encoder{PayloadType: 96}
t.rtpEncoder.Init()
}
return t
}
func (t *streamTrackH264) updateTrackParameters(nalus [][]byte) {
for _, nalu := range nalus {
typ := h264.NALUType(nalu[0] & 0x1F)
switch typ {
case h264.NALUTypeSPS:
if !bytes.Equal(nalu, t.track.SafeSPS()) {
t.track.SafeSetSPS(append([]byte(nil), nalu...))
}
case h264.NALUTypePPS:
if !bytes.Equal(nalu, t.track.SafePPS()) {
t.track.SafeSetPPS(append([]byte(nil), nalu...))
}
}
}
}
// remux is needed to
// - fix corrupted streams
// - make streams compatible with all protocols
func (t *streamTrackH264) remuxNALUs(dat *data) {
n := 0
for _, nalu := range dat.h264NALUs {
typ := h264.NALUType(nalu[0] & 0x1F)
switch typ {
case h264.NALUTypeSPS, h264.NALUTypePPS:
continue
case h264.NALUTypeAccessUnitDelimiter:
continue
case h264.NALUTypeIDR:
n += 2
}
n++
}
filteredNALUs := make([][]byte, n)
i := 0
for _, nalu := range dat.h264NALUs {
typ := h264.NALUType(nalu[0] & 0x1F)
switch typ {
case h264.NALUTypeSPS, h264.NALUTypePPS:
// remove since they're automatically added before every IDR
continue
case h264.NALUTypeAccessUnitDelimiter:
// remove since it is not needed
continue
case h264.NALUTypeIDR:
// add SPS and PPS before every IDR
filteredNALUs[i] = t.track.SafeSPS()
i++
filteredNALUs[i] = t.track.SafePPS()
i++
}
filteredNALUs[i] = nalu
i++
}
dat.h264NALUs = filteredNALUs
}
func (t *streamTrackH264) generateRTPPackets(dat *data) {
// if remuxNALUs() returned nil, do not write any data
if dat.h264NALUs == nil {
return
}
pkts, err := t.rtpEncoder.Encode(dat.h264NALUs, dat.pts)
if err != nil {
return
}
lastPkt := len(pkts) - 1
for i, pkt := range pkts {
if i != lastPkt {
t.writeDataInner(&data{
trackID: dat.trackID,
rtpPacket: pkt,
})
} else {
t.writeDataInner(&data{
trackID: dat.trackID,
rtpPacket: pkt,
ptsEqualsDTS: dat.ptsEqualsDTS,
pts: dat.pts,
h264NALUs: dat.h264NALUs,
})
}
}
}
func (t *streamTrackH264) writeData(dat *data) {
t.updateTrackParameters(dat.h264NALUs)
t.remuxNALUs(dat)
if dat.rtpPacket != nil {
t.writeDataInner(dat)
} else {
t.generateRTPPackets(dat)
}
}
internal/core/streamtrack_mpeg4audio.go
+58
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@@ -0,0 +1,58 @@
package core
import (
"github.com/aler9/gortsplib"
"github.com/aler9/gortsplib/pkg/rtpmpeg4audio"
)
type streamTrackMPEG4Audio struct {
writeDataInner func(*data)
rtpEncoder *rtpmpeg4audio.Encoder
}
func newStreamTrackMPEG4Audio(
track *gortsplib.TrackMPEG4Audio,
generateRTPPackets bool,
writeDataInner func(*data),
) *streamTrackMPEG4Audio {
t := &streamTrackMPEG4Audio{
writeDataInner: writeDataInner,
}
if generateRTPPackets {
t.rtpEncoder = &rtpmpeg4audio.Encoder{
PayloadType: 96,
SampleRate: track.ClockRate(),
SizeLength: 13,
IndexLength: 3,
IndexDeltaLength: 3,
}
t.rtpEncoder.Init()
}
return t
}
func (t *streamTrackMPEG4Audio) generateRTPPackets(dat *data) {
pkts, err := t.rtpEncoder.Encode([][]byte{dat.mpeg4AudioAU}, dat.pts)
if err != nil {
return
}
for _, pkt := range pkts {
t.writeDataInner(&data{
trackID: dat.trackID,
rtpPacket: pkt,
ptsEqualsDTS: true,
})
}
}
func (t *streamTrackMPEG4Audio) writeData(dat *data) {
if dat.rtpPacket != nil {
t.writeDataInner(dat)
} else {
t.generateRTPPackets(dat)
}
}