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// Copyright 2021, Chef. All rights reserved.
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// https://github.com/q191201771/lal
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//
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// Use of this source code is governed by a MIT-style license
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// that can be found in the License file.
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//
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// Author: Chef (191201771@qq.com)
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package remux
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import (
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"math/rand"
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"time"
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"github.com/q191201771/lal/pkg/aac"
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"github.com/q191201771/lal/pkg/avc"
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"github.com/q191201771/lal/pkg/base"
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"github.com/q191201771/lal/pkg/hevc"
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"github.com/q191201771/lal/pkg/rtprtcp"
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"github.com/q191201771/lal/pkg/sdp"
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"github.com/q191201771/naza/pkg/nazalog"
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)
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// TODO(chef): refactor 将analyze部分独立出来作为一个filter
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var (
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// config
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// TODO(chef): 提供option，另外还有ssrc和pt都支持自定义
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maxAnalyzeAvMsgSize = 16
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)
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// 提供rtmp数据向sdp+rtp数据的转换
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type Rtmp2RtspRemuxer struct {
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onSdp OnSdp
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onRtpPacket OnRtpPacket
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analyzeDone bool
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msgCache []base.RtmpMsg
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vps, sps, pps, asc []byte
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audioPt base.AvPacketPt
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videoPt base.AvPacketPt
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audioSsrc uint32
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videoSsrc uint32
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audioPacker *rtprtcp.RtpPacker
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videoPacker *rtprtcp.RtpPacker
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}
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type OnSdp func(rawSdp []byte, sdpCtx sdp.LogicContext)
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type OnRtpPacket func(pkt rtprtcp.RtpPacket)
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func NewRtmp2RtspRemuxer(onSdp OnSdp, onRtpPacket OnRtpPacket) *Rtmp2RtspRemuxer {
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return &Rtmp2RtspRemuxer{
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onSdp: onSdp,
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onRtpPacket: onRtpPacket,
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audioPt: base.AvPacketPtUnknown,
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videoPt: base.AvPacketPtUnknown,
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}
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}
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// @param msg: 函数调用结束后，内部不持有`msg`内存块
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//
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func (r *Rtmp2RtspRemuxer) FeedRtmpMsg(msg base.RtmpMsg) {
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var err error
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if msg.Header.MsgTypeId == base.RtmpTypeIdMetadata {
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// noop
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return
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}
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// 我们需要先接收一部分rtmp数据，得到音频头、视频头
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// 并且考虑，流中只有音频或只有视频的情况
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// 我们把前面这个阶段叫做Analyze分析阶段
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if !r.analyzeDone {
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if msg.IsAvcKeySeqHeader() || msg.IsHevcKeySeqHeader() {
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if msg.IsAvcKeySeqHeader() {
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r.sps, r.pps, err = avc.ParseSpsPpsFromSeqHeader(msg.Payload)
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nazalog.Assert(nil, err)
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} else if msg.IsHevcKeySeqHeader() {
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r.vps, r.sps, r.pps, err = hevc.ParseVpsSpsPpsFromSeqHeader(msg.Payload)
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nazalog.Assert(nil, err)
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}
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r.doAnalyze()
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return
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}
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if msg.IsAacSeqHeader() {
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r.asc = msg.Clone().Payload[2:]
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r.doAnalyze()
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return
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}
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r.msgCache = append(r.msgCache, msg.Clone())
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r.doAnalyze()
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return
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}
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// 正常阶段
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// 音视频头已通过sdp回调，rtp数据中不再包含音视频头
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if msg.IsAvcKeySeqHeader() || msg.IsHevcKeySeqHeader() || msg.IsAacSeqHeader() {
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return
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}
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r.remux(msg)
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}
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func (r *Rtmp2RtspRemuxer) doAnalyze() {
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nazalog.Assert(false, r.analyzeDone)
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if r.isAnalyzeEnough() {
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if r.sps != nil && r.pps != nil {
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if r.vps != nil {
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r.videoPt = base.AvPacketPtHevc
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} else {
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r.videoPt = base.AvPacketPtAvc
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}
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}
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if r.asc != nil {
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r.audioPt = base.AvPacketPtAac
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}
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// 回调sdp
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ctx, rawSdp, err := sdp.Pack(r.vps, r.sps, r.pps, r.asc)
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nazalog.Assert(nil, err)
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r.onSdp(rawSdp, ctx)
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// 分析阶段缓存的数据
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for i := range r.msgCache {
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r.remux(r.msgCache[i])
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}
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r.msgCache = nil
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r.analyzeDone = true
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}
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}
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// 是否应该退出Analyze阶段
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func (r *Rtmp2RtspRemuxer) isAnalyzeEnough() bool {
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// 音视频头都收集好了
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if r.sps != nil && r.pps != nil && r.asc != nil {
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return true
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}
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// 达到分析包数阈值了
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if len(r.msgCache) >= maxAnalyzeAvMsgSize {
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return true
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}
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return false
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}
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func (r *Rtmp2RtspRemuxer) remux(msg base.RtmpMsg) {
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var rtppkts []rtprtcp.RtpPacket
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switch msg.Header.MsgTypeId {
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case base.RtmpTypeIdAudio:
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rtppkts = r.getAudioPacker().Pack(base.AvPacket{
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Timestamp: msg.Header.TimestampAbs,
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PayloadType: r.audioPt,
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Payload: msg.Payload[2:],
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})
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case base.RtmpTypeIdVideo:
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rtppkts = r.getVideoPacker().Pack(base.AvPacket{
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Timestamp: msg.Header.TimestampAbs,
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PayloadType: r.videoPt,
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Payload: msg.Payload[5:],
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})
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}
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for i := range rtppkts {
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r.onRtpPacket(rtppkts[i])
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}
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}
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func (r *Rtmp2RtspRemuxer) getAudioPacker() *rtprtcp.RtpPacker {
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if r.audioPacker == nil {
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// TODO(chef): ssrc随机产生，并且整个lal没有在setup信令中传递ssrc
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r.audioSsrc = rand.Uint32()
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// TODO(chef): 如果rtmp不是以音视频头开始，也可能收到了帧数据，但是头不存在，目前该remux没有做过多容错判断，后续要加上，或者在输入层保证
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ascCtx, err := aac.NewAscContext(r.asc)
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if err != nil {
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nazalog.Errorf("parse asc failed. err=%+v", err)
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}
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clockRate, err := ascCtx.GetSamplingFrequency()
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if err != nil {
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nazalog.Errorf("get sampling frequency failed. err=%+v", err)
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}
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pp := rtprtcp.NewRtpPackerPayloadAac()
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r.audioPacker = rtprtcp.NewRtpPacker(pp, clockRate, r.audioSsrc)
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}
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return r.audioPacker
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}
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func (r *Rtmp2RtspRemuxer) getVideoPacker() *rtprtcp.RtpPacker {
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if r.videoPacker == nil {
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r.videoSsrc = rand.Uint32()
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pp := rtprtcp.NewRtpPackerPayloadAvcHevc(r.videoPt, func(option *rtprtcp.RtpPackerPayloadAvcHevcOption) {
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option.Typ = rtprtcp.RtpPackerPayloadAvcHevcTypeAvcc
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})
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r.videoPacker = rtprtcp.NewRtpPacker(pp, 90000, r.videoSsrc)
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}
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return r.videoPacker
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}
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func init() {
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rand.Seed(time.Now().UnixNano())
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}
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joestarzxh
4 years ago
committed by
GitHub