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* split data into specialized structs * move MPEG4-audio decoding into streamTrack * restore video/audio synchronization in HLS muxer and RTMP server * log decode errors * move H264 decoding and re-encoding here from gortsplib * add tests * update gortsplibpull/1212/head
Alessandro Ros
3 years ago
committed by
GitHub
19 changed files with 860 additions and 246 deletions
@ -1,19 +1,33 @@
@@ -1,19 +1,33 @@
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package core |
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import ( |
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"github.com/aler9/gortsplib" |
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"fmt" |
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) |
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type streamTrackGeneric struct { |
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writeDataInner func(*data) |
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const ( |
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// 1500 (UDP MTU) - 20 (IP header) - 8 (UDP header)
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maxPacketSize = 1472 |
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) |
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type streamTrackGeneric struct{} |
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func newStreamTrackGeneric() *streamTrackGeneric { |
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return &streamTrackGeneric{} |
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} |
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func newStreamTrackGeneric(track gortsplib.Track, writeDataInner func(*data)) *streamTrackGeneric { |
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return &streamTrackGeneric{ |
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writeDataInner: writeDataInner, |
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func (t *streamTrackGeneric) onData(dat data, hasNonRTSPReaders bool) error { |
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tdata := dat.(*dataGeneric) |
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pkt := tdata.rtpPackets[0] |
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// remove padding
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pkt.Header.Padding = false |
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pkt.PaddingSize = 0 |
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if pkt.MarshalSize() > maxPacketSize { |
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return fmt.Errorf("payload size (%d) is greater than maximum allowed (%d)", |
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pkt.MarshalSize(), maxPacketSize) |
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} |
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} |
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func (t *streamTrackGeneric) writeData(data *data) { |
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t.writeDataInner(data) |
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return nil |
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} |
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@ -1,58 +1,89 @@
@@ -1,58 +1,89 @@
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package core |
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import ( |
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"fmt" |
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"github.com/aler9/gortsplib" |
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"github.com/aler9/gortsplib/pkg/rtpmpeg4audio" |
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) |
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type streamTrackMPEG4Audio struct { |
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writeDataInner func(*data) |
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rtpEncoder *rtpmpeg4audio.Encoder |
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track *gortsplib.TrackMPEG4Audio |
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encoder *rtpmpeg4audio.Encoder |
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decoder *rtpmpeg4audio.Decoder |
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} |
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func newStreamTrackMPEG4Audio( |
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track *gortsplib.TrackMPEG4Audio, |
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generateRTPPackets bool, |
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writeDataInner func(*data), |
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) *streamTrackMPEG4Audio { |
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t := &streamTrackMPEG4Audio{ |
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writeDataInner: writeDataInner, |
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track: track, |
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} |
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if generateRTPPackets { |
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t.rtpEncoder = &rtpmpeg4audio.Encoder{ |
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t.encoder = &rtpmpeg4audio.Encoder{ |
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PayloadType: 96, |
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SampleRate: track.ClockRate(), |
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SizeLength: 13, |
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IndexLength: 3, |
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IndexDeltaLength: 3, |
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} |
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t.rtpEncoder.Init() |
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t.encoder.Init() |
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} |
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return t |
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} |
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func (t *streamTrackMPEG4Audio) generateRTPPackets(dat *data) { |
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pkts, err := t.rtpEncoder.Encode([][]byte{dat.mpeg4AudioAU}, dat.pts) |
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func (t *streamTrackMPEG4Audio) generateRTPPackets(tdata *dataMPEG4Audio) error { |
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pkts, err := t.encoder.Encode(tdata.aus, tdata.pts) |
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if err != nil { |
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return |
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return err |
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} |
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for _, pkt := range pkts { |
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t.writeDataInner(&data{ |
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trackID: dat.trackID, |
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rtpPacket: pkt, |
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ptsEqualsDTS: true, |
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}) |
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} |
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tdata.rtpPackets = pkts |
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return nil |
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} |
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func (t *streamTrackMPEG4Audio) writeData(dat *data) { |
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if dat.rtpPacket != nil { |
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t.writeDataInner(dat) |
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} else { |
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t.generateRTPPackets(dat) |
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func (t *streamTrackMPEG4Audio) onData(dat data, hasNonRTSPReaders bool) error { |
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tdata := dat.(*dataMPEG4Audio) |
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if tdata.rtpPackets != nil { |
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pkt := tdata.rtpPackets[0] |
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// remove padding
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pkt.Header.Padding = false |
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pkt.PaddingSize = 0 |
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if pkt.MarshalSize() > maxPacketSize { |
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return fmt.Errorf("payload size (%d) is greater than maximum allowed (%d)", |
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pkt.MarshalSize(), maxPacketSize) |
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} |
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// decode from RTP
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if hasNonRTSPReaders { |
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if t.decoder == nil { |
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t.decoder = &rtpmpeg4audio.Decoder{ |
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SampleRate: t.track.Config.SampleRate, |
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SizeLength: t.track.SizeLength, |
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IndexLength: t.track.IndexLength, |
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IndexDeltaLength: t.track.IndexDeltaLength, |
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} |
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t.decoder.Init() |
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} |
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aus, pts, err := t.decoder.Decode(pkt) |
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if err != nil { |
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return err |
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} |
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tdata.aus = aus |
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tdata.pts = pts |
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} |
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// route packet as is
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return nil |
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} |
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return t.generateRTPPackets(tdata) |
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} |
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