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fix audio

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This commit is contained in:
Andrey Semochkin
2021-01-09 17:24:48 +03:00
parent 33b07c6a20
commit f16439f7ef
4 changed files with 51 additions and 16 deletions
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1
av/av.go
View File

@@ -238,6 +238,7 @@ type Packet struct {
Idx int8 // stream index in container format
CompositionTime time.Duration // packet presentation time minus decode time for H264 B-Frame
Time time.Duration // packet decode time
Duration time.Duration //packet duration
Data []byte // packet data
}

2
codec/codec.go
View File

@@ -26,7 +26,7 @@ func (self OpusCodecData) ChannelLayout() av.ChannelLayout {
}
func (self OpusCodecData) PacketDuration(data []byte) (time.Duration, error) {
return time.Duration(1000) * time.Second / time.Duration(self.SampleRate_), nil
return time.Duration(20) * time.Millisecond, nil
}
func (self OpusCodecData) SampleFormat() av.SampleFormat {

9
format/mp4f/muxer.go
View File

@@ -79,7 +79,7 @@ func (self *Muxer) newStream(codec av.CodecData) (err error) {
stream.sample.SyncSample = &mp4io.SyncSample{}
stream.timeScale = 90000
case av.AAC:
stream.timeScale = 8000
stream.timeScale = int64(codec.(av.AudioCodecData).SampleRate())
}
stream.muxer = self
@@ -179,10 +179,12 @@ func (self *Stream) fillTrackAtom() (err error) {
self.sample.SampleDesc.MP4ADesc = &mp4io.MP4ADesc{
DataRefIdx: 1,
NumberOfChannels: int16(codec.ChannelLayout().Count()),
SampleSize: 16,
SampleSize: int16(codec.SampleFormat().BytesPerSample() * 4),
SampleRate: float64(codec.SampleRate()),
Unknowns: []mp4io.Atom{self.buildEsds(codec.MPEG4AudioConfigBytes())},
}
//log.Fatalln(codec.MPEG4AudioConfigBytes())
//log.Fatalln(codec.SampleFormat().BytesPerSample())
self.trackAtom.Header.Volume = 1
self.trackAtom.Header.AlternateGroup = 1
self.trackAtom.Header.Duration = 0
@@ -270,6 +272,9 @@ func (element *Muxer) WritePacket(pkt av.Packet, GOP bool) (bool, []byte, error)
if stream.lastpkt != nil {
ts = pkt.Time - stream.lastpkt.Time
}
if stream.CodecData.Type().IsAudio() {
pkt.Data = pkt.Data[4:]
}
got, buf, err := stream.writePacketV2(pkt, ts, 5)
stream.lastpkt = &pkt
if err != nil {

55
format/rtspv2/client.go
View File

@@ -77,7 +77,9 @@ type RTSPClient struct {
CodecData []av.CodecData
AudioTimeLine time.Duration
AudioTimeScale int64
audioCodec string
audioCodec av.CodecType
PreAudioTS int64
PreVideoTS int64
}
type RTSPClientOptions struct {
@@ -180,7 +182,7 @@ func Dial(options RTSPClientOptions) (*RTSPClient, error) {
if CodecData != nil {
client.CodecData = append(client.CodecData, CodecData)
client.audioIDX = int8(len(client.CodecData) - 1)
client.audioCodec = CodecData.Type().String()
client.audioCodec = CodecData.Type()
if i2.TimeScale != 0 {
client.AudioTimeScale = int64(i2.TimeScale)
}
@@ -227,7 +229,7 @@ func (client *RTSPClient) startStream() {
timer = time.Now()
}
if !fixed {
nb, err := io.ReadFull(client.conn, header)
nb, err := io.ReadFull(client.connRW, header)
if err != nil || nb != 4 {
client.Println("RTSP Client RTP Read Header", err)
return
@@ -246,7 +248,7 @@ func (client *RTSPClient) startStream() {
content[1] = header[1]
content[2] = header[2]
content[3] = header[3]
n, rerr := io.ReadFull(client.conn, content[4:length+4])
n, rerr := io.ReadFull(client.connRW, content[4:length+4])
if rerr != nil || n != int(length) {
client.Println("RTSP Client RTP ReadFull", err)
return
@@ -274,7 +276,7 @@ func (client *RTSPClient) startStream() {
case 0x52:
var responseTmp []byte
for {
n, rerr := io.ReadFull(client.conn, oneb)
n, rerr := io.ReadFull(client.connRW, oneb)
if rerr != nil || n != 1 {
client.Println("RTSP Client RTP Read Keep-Alive Header", rerr)
return
@@ -288,7 +290,7 @@ func (client *RTSPClient) startStream() {
return
}
cont := make([]byte, si)
_, err = io.ReadFull(client.conn, cont)
_, err = io.ReadFull(client.connRW, cont)
if err != nil {
client.Println("RTSP Client RTP Read Keep-Alive ReadFull", err)
return
@@ -512,6 +514,9 @@ func (client *RTSPClient) RTPDemuxer(payloadRAW *[]byte) ([]*av.Packet, bool) {
offset += 4
switch int(content[1]) {
case client.videoID:
if client.PreVideoTS == 0 {
client.PreVideoTS = timestamp
}
if client.BufferRtpPacket.Len() > 4048576 {
client.Println("Big Buffer Flush")
client.BufferRtpPacket.Truncate(0)
@@ -529,6 +534,7 @@ func (client *RTSPClient) RTPDemuxer(payloadRAW *[]byte) ([]*av.Packet, bool) {
CompositionTime: time.Duration(1) * time.Millisecond,
Idx: client.videoIDX,
IsKeyFrame: naluType == 5,
Duration: time.Duration(float32(timestamp-client.PreVideoTS)/90) * time.Millisecond,
Time: time.Duration(timestamp/90) * time.Millisecond,
})
case naluType == 7:
@@ -556,40 +562,63 @@ func (client *RTSPClient) RTPDemuxer(payloadRAW *[]byte) ([]*av.Packet, bool) {
retmap = append(retmap, &av.Packet{
Data: append(binSize(client.BufferRtpPacket.Len()), client.BufferRtpPacket.Bytes()...),
CompositionTime: time.Duration(1) * time.Millisecond,
Duration: time.Duration(float32(timestamp-client.PreVideoTS)/90) * time.Millisecond,
Idx: client.videoIDX,
IsKeyFrame: naluTypef == 5,
Time: time.Duration(timestamp/90) * time.Millisecond,
})
}
}
default:
client.Println("Unsupported NAL Type", naluType)
}
}
if len(retmap) > 0 {
client.PreVideoTS = timestamp
return retmap, true
}
case client.audioID:
if client.PreAudioTS == 0 {
client.PreAudioTS = timestamp
}
nalRaw, _ := h264parser.SplitNALUs(content[offset:end])
var retmap []*av.Packet
for _, nal := range nalRaw {
if client.audioCodec == av.PCM_MULAW.String() || client.audioCodec == av.PCM_ALAW.String() || client.audioCodec == av.PCM.String() {
client.AudioTimeLine += time.Duration(len(nal)) * time.Second / time.Duration(client.AudioTimeScale)
} else if client.audioCodec == av.OPUS.String() {
client.AudioTimeLine += time.Duration(20) * time.Millisecond
} else {
client.AudioTimeLine = time.Duration(float32(timestamp)/float32(float32(client.AudioTimeScale)/float32(1000))) * time.Millisecond
var duration time.Duration
switch client.audioCodec {
case av.PCM_MULAW:
duration = time.Duration(len(nal)) * time.Second / time.Duration(client.AudioTimeScale)
client.AudioTimeLine += duration
case av.PCM_ALAW:
duration = time.Duration(len(nal)) * time.Second / time.Duration(client.AudioTimeScale)
client.AudioTimeLine += duration
case av.OPUS:
duration = time.Duration(20) * time.Millisecond
client.AudioTimeLine += duration
case av.AAC:
if nal[1] == 32 {
return nil, false
}
nal = nal[4:]
if _, _, _, _, err := aacparser.ParseADTSHeader(nal); err == nil {
nal = nal[7:]
}
duration = time.Duration((float32(1024)/float32(client.AudioTimeScale))*1000) * time.Millisecond
client.AudioTimeLine += duration
}
retmap = append(retmap, &av.Packet{
Data: append(binSize(len(nal)), nal...),
CompositionTime: time.Duration(1) * time.Millisecond,
Duration: duration,
Idx: client.audioIDX,
IsKeyFrame: false,
Time: client.AudioTimeLine,
})
}
if len(retmap) > 0 {
client.PreAudioTS = timestamp
return retmap, true
}
default: