|
|
// Copyright 2020, Chef. All rights reserved.
|
|
|
// https://github.com/q191201771/lal
|
|
|
//
|
|
|
// Use of this source code is governed by a MIT-style license
|
|
|
// that can be found in the License file.
|
|
|
//
|
|
|
// Author: Chef (191201771@qq.com)
|
|
|
|
|
|
package hls
|
|
|
|
|
|
import (
|
|
|
"github.com/q191201771/lal/pkg/aac"
|
|
|
"github.com/q191201771/lal/pkg/avc"
|
|
|
"github.com/q191201771/lal/pkg/base"
|
|
|
"github.com/q191201771/lal/pkg/mpegts"
|
|
|
"github.com/q191201771/naza/pkg/bele"
|
|
|
"github.com/q191201771/naza/pkg/nazalog"
|
|
|
)
|
|
|
|
|
|
type StreamerObserver interface {
|
|
|
// @param streamer: 供上层获取streamer内部的一些状态，比如spspps是否已缓存，音频缓存队列是否有数据等
|
|
|
//
|
|
|
// @param frame: 各字段含义见mpegts.Frame结构体定义
|
|
|
// frame.CC 注意，回调结束后，Streamer会保存frame.CC，上层在TS打包完成后，可通过frame.CC将cc值传递给Streamer
|
|
|
// frame.Raw 回调结束后，这块内存可能会被内部重复使用
|
|
|
//
|
|
|
OnFrame(streamer *Streamer, frame *mpegts.Frame)
|
|
|
}
|
|
|
|
|
|
type Streamer struct {
|
|
|
UniqueKey string
|
|
|
|
|
|
observer StreamerObserver
|
|
|
videoOut []byte // AnnexB TODO chef: 优化这块buff
|
|
|
spspps []byte // AnnexB
|
|
|
adts aac.ADTS
|
|
|
audioCacheFrames []byte // 缓存音频帧数据，注意，可能包含多个音频帧 TODO chef: 优化这块buff
|
|
|
audioCacheFirstFramePTS uint64 // audioCacheFrames中第一个音频帧的时间戳 TODO chef: rename to DTS
|
|
|
audioCC uint8
|
|
|
videoCC uint8
|
|
|
}
|
|
|
|
|
|
func NewStreamer(observer StreamerObserver) *Streamer {
|
|
|
uk := base.GenUniqueKey(base.UKPStreamer)
|
|
|
videoOut := make([]byte, 1024*1024)
|
|
|
videoOut = videoOut[0:0]
|
|
|
return &Streamer{
|
|
|
UniqueKey: uk,
|
|
|
observer: observer,
|
|
|
videoOut: videoOut,
|
|
|
}
|
|
|
}
|
|
|
|
|
|
// @param msg msg.Payload 调用结束后，函数内部不会持有这块内存
|
|
|
//
|
|
|
// TODO chef: 可以考虑数据有问题时，返回给上层，直接主动关闭输入流的连接
|
|
|
func (s *Streamer) FeedRTMPMessage(msg base.RTMPMsg) {
|
|
|
switch msg.Header.MsgTypeID {
|
|
|
case base.RTMPTypeIDAudio:
|
|
|
s.feedAudio(msg)
|
|
|
case base.RTMPTypeIDVideo:
|
|
|
s.feedVideo(msg)
|
|
|
}
|
|
|
}
|
|
|
|
|
|
func (s *Streamer) AudioSeqHeaderCached() bool {
|
|
|
return s.adts.HasInited()
|
|
|
}
|
|
|
|
|
|
func (s *Streamer) VideoSeqHeaderCached() bool {
|
|
|
return s.spspps != nil
|
|
|
}
|
|
|
|
|
|
func (s *Streamer) AudioCacheEmpty() bool {
|
|
|
return s.audioCacheFrames == nil
|
|
|
}
|
|
|
|
|
|
func (s *Streamer) feedVideo(msg base.RTMPMsg) {
|
|
|
if len(msg.Payload) < 5 {
|
|
|
nazalog.Errorf("[%s] invalid video message length. len=%d", s.UniqueKey, len(msg.Payload))
|
|
|
return
|
|
|
}
|
|
|
if msg.Payload[0]&0xF != base.RTMPCodecIDAVC {
|
|
|
return
|
|
|
}
|
|
|
|
|
|
ftype := msg.Payload[0] & 0xF0 >> 4
|
|
|
htype := msg.Payload[1]
|
|
|
|
|
|
// 将数据转换成AnnexB
|
|
|
|
|
|
// 如果是sps pps，缓存住，然后直接返回
|
|
|
if ftype == base.RTMPFrameTypeKey && htype == base.RTMPAVCPacketTypeSeqHeader {
|
|
|
if err := s.cacheSPSPPS(msg); err != nil {
|
|
|
nazalog.Errorf("[%s] cache spspps failed. err=%+v", s.UniqueKey, err)
|
|
|
}
|
|
|
return
|
|
|
}
|
|
|
|
|
|
cts := bele.BEUint24(msg.Payload[2:])
|
|
|
|
|
|
audSent := false
|
|
|
spsppsSent := false
|
|
|
// 优化这块buffer
|
|
|
out := s.videoOut[0:0]
|
|
|
|
|
|
// tag中可能有多个NALU，逐个获取
|
|
|
for i := 5; i != len(msg.Payload); {
|
|
|
if i+4 > len(msg.Payload) {
|
|
|
nazalog.Errorf("[%s] slice len not enough. i=%d, len=%d", s.UniqueKey, i, len(msg.Payload))
|
|
|
return
|
|
|
}
|
|
|
nalBytes := int(bele.BEUint32(msg.Payload[i:]))
|
|
|
i += 4
|
|
|
if i+nalBytes > len(msg.Payload) {
|
|
|
nazalog.Errorf("[%s] slice len not enough. i=%d, payload len=%d, nalBytes=%d", s.UniqueKey, i, len(msg.Payload), nalBytes)
|
|
|
return
|
|
|
}
|
|
|
|
|
|
nalType := avc.ParseNALUType(msg.Payload[i])
|
|
|
|
|
|
//nazalog.Debugf("[%s] hls: h264 NAL type=%d, len=%d(%d) cts=%d.", s.UniqueKey, nalType, nalBytes, len(msg.Payload), cts)
|
|
|
|
|
|
// sps pps前面已经缓存过了，这里就不用处理了
|
|
|
// aud有自己的生产逻辑，原流中的aud直接过滤掉
|
|
|
if nalType == avc.NALUTypeSPS || nalType == avc.NALUTypePPS || nalType == avc.NALUTypeAUD {
|
|
|
i += nalBytes
|
|
|
continue
|
|
|
}
|
|
|
|
|
|
if !audSent {
|
|
|
switch nalType {
|
|
|
case avc.NALUTypeSlice, avc.NALUTypeIDRSlice, avc.NALUTypeSEI:
|
|
|
// 在前面写入aud
|
|
|
out = append(out, audNal...)
|
|
|
audSent = true
|
|
|
//case avc.NALUTypeAUD:
|
|
|
// // 上面aud已经continue跳过了，应该进不到这个分支，可以考虑删除这个分支代码
|
|
|
// audSent = true
|
|
|
}
|
|
|
}
|
|
|
|
|
|
switch nalType {
|
|
|
case avc.NALUTypeSlice:
|
|
|
spsppsSent = false
|
|
|
case avc.NALUTypeIDRSlice:
|
|
|
// 如果是首个关键帧，在前面写入sps pps
|
|
|
if !spsppsSent {
|
|
|
var err error
|
|
|
out, err = s.appendSPSPPS(out)
|
|
|
if err != nil {
|
|
|
nazalog.Warnf("[%s] append spspps by not exist.", s.UniqueKey)
|
|
|
return
|
|
|
}
|
|
|
}
|
|
|
spsppsSent = true
|
|
|
|
|
|
}
|
|
|
|
|
|
// 这里不知为什么要区分写入两种类型的start code
|
|
|
if len(out) == 0 {
|
|
|
out = append(out, avc.NALUStartCode4...)
|
|
|
} else {
|
|
|
out = append(out, avc.NALUStartCode3...)
|
|
|
}
|
|
|
|
|
|
out = append(out, msg.Payload[i:i+nalBytes]...)
|
|
|
|
|
|
i += nalBytes
|
|
|
}
|
|
|
|
|
|
key := ftype == base.RTMPFrameTypeKey && htype == base.RTMPAVCPacketTypeNALU
|
|
|
dts := uint64(msg.Header.TimestampAbs) * 90
|
|
|
|
|
|
if s.audioCacheFrames != nil && s.audioCacheFirstFramePTS+maxAudioCacheDelayByVideo < dts {
|
|
|
s.FlushAudio()
|
|
|
}
|
|
|
|
|
|
var frame mpegts.Frame
|
|
|
frame.CC = s.videoCC
|
|
|
frame.DTS = dts
|
|
|
frame.PTS = frame.DTS + uint64(cts)*90
|
|
|
frame.Key = key
|
|
|
frame.Raw = out
|
|
|
frame.Pid = mpegts.PidVideo
|
|
|
frame.Sid = mpegts.StreamIDVideo
|
|
|
|
|
|
s.observer.OnFrame(s, &frame)
|
|
|
s.videoCC = frame.CC
|
|
|
}
|
|
|
|
|
|
func (s *Streamer) feedAudio(msg base.RTMPMsg) {
|
|
|
if len(msg.Payload) < 3 {
|
|
|
nazalog.Errorf("[%s] invalid audio message length. len=%d", s.UniqueKey, len(msg.Payload))
|
|
|
return
|
|
|
}
|
|
|
if msg.Payload[0]>>4 != base.RTMPSoundFormatAAC {
|
|
|
return
|
|
|
}
|
|
|
|
|
|
//nazalog.Debugf("[%s] hls: feedAudio. dts=%d len=%d", s.UniqueKey, msg.Header.TimestampAbs, len(msg.Payload))
|
|
|
|
|
|
if msg.Payload[1] == base.RTMPAACPacketTypeSeqHeader {
|
|
|
if err := s.cacheAACSeqHeader(msg); err != nil {
|
|
|
nazalog.Errorf("[%s] cache aac seq header failed. err=%+v", s.UniqueKey, err)
|
|
|
}
|
|
|
return
|
|
|
}
|
|
|
|
|
|
if !s.adts.HasInited() {
|
|
|
nazalog.Warnf("[%s] feed audio message but aac seq header not exist.", s.UniqueKey)
|
|
|
return
|
|
|
}
|
|
|
|
|
|
pts := uint64(msg.Header.TimestampAbs) * 90
|
|
|
|
|
|
if s.audioCacheFrames != nil && s.audioCacheFirstFramePTS+maxAudioCacheDelayByAudio < pts {
|
|
|
s.FlushAudio()
|
|
|
}
|
|
|
|
|
|
if s.audioCacheFrames == nil {
|
|
|
s.audioCacheFirstFramePTS = pts
|
|
|
}
|
|
|
|
|
|
adtsHeader, _ := s.adts.CalcADTSHeader(uint16(msg.Header.MsgLen - 2))
|
|
|
s.audioCacheFrames = append(s.audioCacheFrames, adtsHeader...)
|
|
|
s.audioCacheFrames = append(s.audioCacheFrames, msg.Payload[2:]...)
|
|
|
}
|
|
|
|
|
|
// 吐出音频数据的三种情况：
|
|
|
// 1. 收到音频或视频时，音频缓存队列已达到一定长度
|
|
|
// 2. 打开一个新的TS文件切片时
|
|
|
// 3. 输入流关闭时
|
|
|
func (s *Streamer) FlushAudio() {
|
|
|
if s.audioCacheFrames == nil {
|
|
|
return
|
|
|
}
|
|
|
|
|
|
var frame mpegts.Frame
|
|
|
frame.CC = s.audioCC
|
|
|
frame.DTS = s.audioCacheFirstFramePTS
|
|
|
frame.PTS = s.audioCacheFirstFramePTS
|
|
|
frame.Key = false
|
|
|
frame.Raw = s.audioCacheFrames
|
|
|
frame.Pid = mpegts.PidAudio
|
|
|
frame.Sid = mpegts.StreamIDAudio
|
|
|
s.observer.OnFrame(s, &frame)
|
|
|
s.audioCC = frame.CC
|
|
|
|
|
|
s.audioCacheFrames = nil
|
|
|
}
|
|
|
|
|
|
func (s *Streamer) cacheAACSeqHeader(msg base.RTMPMsg) error {
|
|
|
return s.adts.InitWithAACAudioSpecificConfig(msg.Payload[2:])
|
|
|
}
|
|
|
|
|
|
func (s *Streamer) cacheSPSPPS(msg base.RTMPMsg) error {
|
|
|
var err error
|
|
|
s.spspps, err = avc.SPSPPSSeqHeader2AnnexB(msg.Payload)
|
|
|
return err
|
|
|
}
|
|
|
|
|
|
func (s *Streamer) appendSPSPPS(out []byte) ([]byte, error) {
|
|
|
if s.spspps == nil {
|
|
|
return out, ErrHLS
|
|
|
}
|
|
|
|
|
|
out = append(out, s.spspps...)
|
|
|
return out, nil
|
|
|
}
|