fix audio
This commit is contained in:
Andrey Semochkin
parent
33b07c6a20
commit
f16439f7ef
1
av/av.go
1
av/av.go
@ -238,6 +238,7 @@ type Packet struct {
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Idx int8 // stream index in container format
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CompositionTime time.Duration // packet presentation time minus decode time for H264 B-Frame
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Time time.Duration // packet decode time
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Duration time.Duration //packet duration
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Data []byte // packet data
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}
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@ -26,7 +26,7 @@ func (self OpusCodecData) ChannelLayout() av.ChannelLayout {
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}
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func (self OpusCodecData) PacketDuration(data []byte) (time.Duration, error) {
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return time.Duration(1000) * time.Second / time.Duration(self.SampleRate_), nil
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return time.Duration(20) * time.Millisecond, nil
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}
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func (self OpusCodecData) SampleFormat() av.SampleFormat {
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@ -79,7 +79,7 @@ func (self *Muxer) newStream(codec av.CodecData) (err error) {
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stream.sample.SyncSample = &mp4io.SyncSample{}
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stream.timeScale = 90000
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case av.AAC:
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stream.timeScale = 8000
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stream.timeScale = int64(codec.(av.AudioCodecData).SampleRate())
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}
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stream.muxer = self
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@ -179,10 +179,12 @@ func (self *Stream) fillTrackAtom() (err error) {
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self.sample.SampleDesc.MP4ADesc = &mp4io.MP4ADesc{
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DataRefIdx: 1,
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NumberOfChannels: int16(codec.ChannelLayout().Count()),
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SampleSize: 16,
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SampleSize: int16(codec.SampleFormat().BytesPerSample() * 4),
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SampleRate: float64(codec.SampleRate()),
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Unknowns: []mp4io.Atom{self.buildEsds(codec.MPEG4AudioConfigBytes())},
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}
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//log.Fatalln(codec.MPEG4AudioConfigBytes())
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//log.Fatalln(codec.SampleFormat().BytesPerSample())
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self.trackAtom.Header.Volume = 1
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self.trackAtom.Header.AlternateGroup = 1
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self.trackAtom.Header.Duration = 0
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@ -270,6 +272,9 @@ func (element *Muxer) WritePacket(pkt av.Packet, GOP bool) (bool, []byte, error)
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if stream.lastpkt != nil {
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ts = pkt.Time - stream.lastpkt.Time
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}
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if stream.CodecData.Type().IsAudio() {
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pkt.Data = pkt.Data[4:]
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}
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got, buf, err := stream.writePacketV2(pkt, ts, 5)
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stream.lastpkt = &pkt
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if err != nil {
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@ -77,7 +77,9 @@ type RTSPClient struct {
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CodecData []av.CodecData
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AudioTimeLine time.Duration
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AudioTimeScale int64
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audioCodec string
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audioCodec av.CodecType
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PreAudioTS int64
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PreVideoTS int64
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}
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type RTSPClientOptions struct {
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@ -180,7 +182,7 @@ func Dial(options RTSPClientOptions) (*RTSPClient, error) {
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if CodecData != nil {
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client.CodecData = append(client.CodecData, CodecData)
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client.audioIDX = int8(len(client.CodecData) - 1)
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client.audioCodec = CodecData.Type().String()
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client.audioCodec = CodecData.Type()
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if i2.TimeScale != 0 {
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client.AudioTimeScale = int64(i2.TimeScale)
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}
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@ -227,7 +229,7 @@ func (client *RTSPClient) startStream() {
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timer = time.Now()
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}
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if !fixed {
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nb, err := io.ReadFull(client.conn, header)
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nb, err := io.ReadFull(client.connRW, header)
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if err != nil || nb != 4 {
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client.Println("RTSP Client RTP Read Header", err)
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return
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@ -246,7 +248,7 @@ func (client *RTSPClient) startStream() {
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content[1] = header[1]
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content[2] = header[2]
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content[3] = header[3]
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n, rerr := io.ReadFull(client.conn, content[4:length+4])
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n, rerr := io.ReadFull(client.connRW, content[4:length+4])
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if rerr != nil || n != int(length) {
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client.Println("RTSP Client RTP ReadFull", err)
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return
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@ -274,7 +276,7 @@ func (client *RTSPClient) startStream() {
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case 0x52:
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var responseTmp []byte
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for {
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n, rerr := io.ReadFull(client.conn, oneb)
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n, rerr := io.ReadFull(client.connRW, oneb)
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if rerr != nil || n != 1 {
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client.Println("RTSP Client RTP Read Keep-Alive Header", rerr)
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return
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@ -288,7 +290,7 @@ func (client *RTSPClient) startStream() {
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return
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}
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cont := make([]byte, si)
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_, err = io.ReadFull(client.conn, cont)
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_, err = io.ReadFull(client.connRW, cont)
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if err != nil {
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client.Println("RTSP Client RTP Read Keep-Alive ReadFull", err)
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return
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@ -512,6 +514,9 @@ func (client *RTSPClient) RTPDemuxer(payloadRAW *[]byte) ([]*av.Packet, bool) {
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offset += 4
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switch int(content[1]) {
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case client.videoID:
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if client.PreVideoTS == 0 {
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client.PreVideoTS = timestamp
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}
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if client.BufferRtpPacket.Len() > 4048576 {
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client.Println("Big Buffer Flush")
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client.BufferRtpPacket.Truncate(0)
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@ -529,6 +534,7 @@ func (client *RTSPClient) RTPDemuxer(payloadRAW *[]byte) ([]*av.Packet, bool) {
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CompositionTime: time.Duration(1) * time.Millisecond,
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Idx: client.videoIDX,
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IsKeyFrame: naluType == 5,
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Duration: time.Duration(float32(timestamp-client.PreVideoTS)/90) * time.Millisecond,
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Time: time.Duration(timestamp/90) * time.Millisecond,
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})
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case naluType == 7:
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@ -556,40 +562,63 @@ func (client *RTSPClient) RTPDemuxer(payloadRAW *[]byte) ([]*av.Packet, bool) {
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retmap = append(retmap, &av.Packet{
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Data: append(binSize(client.BufferRtpPacket.Len()), client.BufferRtpPacket.Bytes()...),
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CompositionTime: time.Duration(1) * time.Millisecond,
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Duration: time.Duration(float32(timestamp-client.PreVideoTS)/90) * time.Millisecond,
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Idx: client.videoIDX,
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IsKeyFrame: naluTypef == 5,
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Time: time.Duration(timestamp/90) * time.Millisecond,
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})
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}
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}
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default:
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client.Println("Unsupported NAL Type", naluType)
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}
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}
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if len(retmap) > 0 {
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client.PreVideoTS = timestamp
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return retmap, true
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}
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case client.audioID:
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if client.PreAudioTS == 0 {
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client.PreAudioTS = timestamp
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}
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nalRaw, _ := h264parser.SplitNALUs(content[offset:end])
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var retmap []*av.Packet
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for _, nal := range nalRaw {
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if client.audioCodec == av.PCM_MULAW.String() || client.audioCodec == av.PCM_ALAW.String() || client.audioCodec == av.PCM.String() {
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client.AudioTimeLine += time.Duration(len(nal)) * time.Second / time.Duration(client.AudioTimeScale)
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} else if client.audioCodec == av.OPUS.String() {
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client.AudioTimeLine += time.Duration(20) * time.Millisecond
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} else {
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client.AudioTimeLine = time.Duration(float32(timestamp)/float32(float32(client.AudioTimeScale)/float32(1000))) * time.Millisecond
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var duration time.Duration
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switch client.audioCodec {
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case av.PCM_MULAW:
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duration = time.Duration(len(nal)) * time.Second / time.Duration(client.AudioTimeScale)
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client.AudioTimeLine += duration
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case av.PCM_ALAW:
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duration = time.Duration(len(nal)) * time.Second / time.Duration(client.AudioTimeScale)
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client.AudioTimeLine += duration
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case av.OPUS:
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duration = time.Duration(20) * time.Millisecond
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client.AudioTimeLine += duration
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case av.AAC:
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if nal[1] == 32 {
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return nil, false
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}
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nal = nal[4:]
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if _, _, _, _, err := aacparser.ParseADTSHeader(nal); err == nil {
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nal = nal[7:]
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}
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duration = time.Duration((float32(1024)/float32(client.AudioTimeScale))*1000) * time.Millisecond
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client.AudioTimeLine += duration
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}
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retmap = append(retmap, &av.Packet{
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Data: append(binSize(len(nal)), nal...),
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CompositionTime: time.Duration(1) * time.Millisecond,
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Duration: duration,
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Idx: client.audioIDX,
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IsKeyFrame: false,
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Time: client.AudioTimeLine,
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})
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}
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if len(retmap) > 0 {
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client.PreAudioTS = timestamp
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return retmap, true
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}
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default:
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