mirror of
https://github.com/Monibuca/plugin-rtsp.git
synced 2025-09-26 19:51:14 +08:00
78 lines
2.0 KiB
Go
78 lines
2.0 KiB
Go
package rtsp
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import (
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"github.com/aler9/gortsplib/v2/pkg/codecs/mpeg4audio"
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"github.com/aler9/gortsplib/v2/pkg/format"
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"github.com/aler9/gortsplib/v2/pkg/media"
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"github.com/pion/rtp"
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"go.uber.org/zap"
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. "m7s.live/engine/v4"
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"m7s.live/engine/v4/common"
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. "m7s.live/engine/v4/track"
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)
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type RTSPPublisher struct {
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Publisher
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Tracks map[*media.Media]common.AVTrack `json:"-"`
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RTSPIO
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}
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func (p *RTSPPublisher) SetTracks() error {
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p.Tracks = make(map[*media.Media]common.AVTrack, len(p.tracks))
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defer func() {
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for _, track := range p.Tracks {
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p.Info("set track", zap.String("name", track.GetBase().Name))
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}
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}()
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for _, track := range p.tracks {
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for _, forma := range track.Formats {
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switch f := forma.(type) {
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case *format.H264:
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vt := NewH264(p.Stream, f.PayloadType())
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p.Tracks[track] = vt
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if len(f.SPS) > 0 {
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vt.WriteSliceBytes(f.SPS)
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}
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if len(f.PPS) > 0 {
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vt.WriteSliceBytes(f.PPS)
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}
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case *format.H265:
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vt := NewH265(p.Stream, f.PayloadType())
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p.Tracks[track] = vt
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if len(f.VPS) > 0 {
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vt.WriteSliceBytes(f.VPS)
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}
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if len(f.SPS) > 0 {
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vt.WriteSliceBytes(f.SPS)
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}
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if len(f.PPS) > 0 {
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vt.WriteSliceBytes(f.PPS)
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}
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case *format.MPEG4Audio:
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at := NewAAC(p.Stream, f.PayloadType(), uint32(f.ClockRate()))
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p.Tracks[track] = at
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at.SizeLength = f.SizeLength
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if f.Config.Type == mpeg4audio.ObjectTypeAACLC {
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at.Mode = 1
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}
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at.SampleRate = uint32(f.Config.SampleRate)
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at.Channels = uint8(f.Config.ChannelCount)
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asc, _ := f.Config.Marshal()
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// 复用AVCC写入逻辑，解析出AAC的配置信息
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at.WriteSequenceHead(append([]byte{0xAF, 0x00}, asc...))
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case *format.G711:
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at := NewG711(p.Stream, !f.MULaw, f.PayloadType(), uint32(f.ClockRate()))
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p.Tracks[track] = at
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at.AVCCHead = []byte{(byte(at.CodecID) << 4) | (1 << 1)}
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}
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}
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}
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return nil
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}
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func (p *RTSPPublisher) OnPacket(m *media.Media, f format.Format, pack *rtp.Packet) {
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if t, ok := p.Tracks[m]; ok {
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t.WriteRTPPack(pack)
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}
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}
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