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rtsp支持h265拉流

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This commit is contained in:
yangjiechina
2024-04-17 17:21:14 +08:00
parent 1fd0a2f063
commit 0be53fbf4b
3 changed files with 13 additions and 9 deletions
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11
rtmp/rtmp_publisher.go
View File

@@ -75,13 +75,7 @@ func (p *publisher) OnDiscardPacket(pkt interface{}) {
} }
func (p *publisher) OnDeMuxStream(stream_ utils.AVStream) { func (p *publisher) OnDeMuxStream(stream_ utils.AVStream) {
//AVStream的Data单独拷贝出来
//释放掉内存池中最新分配的内存 //释放掉内存池中最新分配的内存
tmp := stream_.Extra()
bytes := make([]byte, len(tmp))
copy(bytes, tmp)
stream_.SetExtraData(bytes)
if utils.AVMediaTypeAudio == stream_.Type() { if utils.AVMediaTypeAudio == stream_.Type() {
p.audioMemoryPool.FreeTail() p.audioMemoryPool.FreeTail()
} else if utils.AVMediaTypeVideo == stream_.Type() { } else if utils.AVMediaTypeVideo == stream_.Type() {
@@ -100,10 +94,11 @@ func (p *publisher) OnDeMuxPacket(packet utils.AVPacket) {
return return
} }
//未开启GOP缓存，释放掉内存
if utils.AVMediaTypeAudio == packet.MediaType() { if utils.AVMediaTypeAudio == packet.MediaType() {
p.audioMemoryPool.FreeHead() p.audioMemoryPool.FreeTail()
} else if utils.AVMediaTypeVideo == packet.MediaType() { } else if utils.AVMediaTypeVideo == packet.MediaType() {
p.videoMemoryPool.FreeHead() p.videoMemoryPool.FreeTail()
} }
} }

1
rtsp/rtsp_session.go
View File

@@ -271,6 +271,7 @@ func (s *session) Input(method string, url_ *url.URL, headers textproto.MIMEHead
var err error var err error
split := strings.Split(url_.Path, "/") split := strings.Split(url_.Path, "/")
source := split[len(split)-1] source := split[len(split)-1]
if MethodOptions == method { if MethodOptions == method {
err = s.onOptions(source, headers) err = s.onOptions(source, headers)
} else if MethodDescribe == method { } else if MethodDescribe == method {

10
rtsp/rtsp_stream.go
View File

@@ -49,10 +49,13 @@ func (t *tranStream) onAllocBuffer(params interface{}) []byte {
return t.rtpTracks[params.(int)].buffer[OverTcpHeaderSize:] return t.rtpTracks[params.(int)].buffer[OverTcpHeaderSize:]
} }
// onRtpPacket 所有封装后的RTP流都将回调于此
func (t *tranStream) onRtpPacket(data []byte, timestamp uint32, params interface{}) { func (t *tranStream) onRtpPacket(data []byte, timestamp uint32, params interface{}) {
//params传递track索引
index := params.(int) index := params.(int)
track := t.rtpTracks[index] track := t.rtpTracks[index]
//保存sps和ssp等数据
if track.cache && track.header == nil { if track.cache && track.header == nil {
bytes := make([]byte, OverTcpHeaderSize+len(data)) bytes := make([]byte, OverTcpHeaderSize+len(data))
copy(bytes[OverTcpHeaderSize:], data) copy(bytes[OverTcpHeaderSize:], data)
@@ -155,11 +158,16 @@ func (t *tranStream) AddTrack(stream utils.AVStream) error {
//创建RTP封装 //创建RTP封装
var muxer librtp.Muxer var muxer librtp.Muxer
if utils.AVCodecIdH264 == stream.CodecId() || utils.AVCodecIdH265 == stream.CodecId() { if utils.AVCodecIdH264 == stream.CodecId() {
muxer = librtp.NewH264Muxer(payloadType.Pt, 0, 0xFFFFFFFF) muxer = librtp.NewH264Muxer(payloadType.Pt, 0, 0xFFFFFFFF)
} else if utils.AVCodecIdH265 == stream.CodecId() {
muxer = librtp.NewH265Muxer(payloadType.Pt, 0, 0xFFFFFFFF)
} else if utils.AVCodecIdAAC == stream.CodecId() { } else if utils.AVCodecIdAAC == stream.CodecId() {
muxer = librtp.NewAACMuxer(payloadType.Pt, 0, 0xFFFFFFFF) muxer = librtp.NewAACMuxer(payloadType.Pt, 0, 0xFFFFFFFF)
} else if utils.AVCodecIdPCMALAW == stream.CodecId() || utils.AVCodecIdPCMMULAW == stream.CodecId() {
muxer = librtp.NewMuxer(payloadType.Pt, 0, 0xFFFFFFFF)
} }
muxer.SetAllocHandler(t.onAllocBuffer) muxer.SetAllocHandler(t.onAllocBuffer)
muxer.SetWriteHandler(t.onRtpPacket) muxer.SetWriteHandler(t.onRtpPacket)