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This commit is contained in:
3503207480@qq.com 2019-08-30 11:17:27 +08:00
parent e3139134e8
commit 036b36f012
12 changed files with 37 additions and 12 deletions
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2
src/Common/MediaSink.cpp
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@ -70,7 +70,7 @@ void MediaSink::inputFrame(const Frame::Ptr &frame) {
it->second->inputFrame(frame);
if(!_allTrackReady && !_trackReadyCallback.empty() && it->second->ready()){
//Track由未就绪状态装换成就绪状态，我们就触发onTrackReady回调
//Track由未就绪状态转换成就绪状态，我们就触发onTrackReady回调
auto it_callback = _trackReadyCallback.find(codec_id);
if(it_callback != _trackReadyCallback.end()){
it_callback->second();

3
src/Common/MediaSource.cpp
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@ -123,6 +123,7 @@ MediaSource::Ptr MediaSource::find(
lock_guard<recursive_mutex> lock(g_mtxMediaSrc);
MediaSource::Ptr ret;
//查找某一媒体源，找到后返回
searchMedia(schema, vhost, app, id,
[&](SchemaVhostAppStreamMap::iterator &it0 ,
VhostAppStreamMap::iterator &it1,
@ -138,7 +139,7 @@ MediaSource::Ptr MediaSource::find(
return true;
});
if(!ret && bMake){
//查找某一媒体源，找到后返回
//未查找媒体源，则创建一个
ret = MediaReader::onMakeMediaSource(schema, vhost,app,id);
}
return ret;

2
src/Extension/AAC.h
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@ -48,7 +48,7 @@ string makeAdtsConfig(const uint8_t *pcAdts);
void getAACInfo(const AACFrame &adts,int &iSampleRate,int &iChannel);
/**
/**
* aac帧，包含adts头
*/
class AACFrame : public Frame {

1
src/Extension/H264Rtmp.h
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@ -36,6 +36,7 @@ using namespace toolkit;
namespace mediakit{
/**
* h264 Rtmp解码类
* 将 h264 over rtmp 解复用出 h264-Frame
*/
class H264RtmpDecoder : public RtmpCodec ,public ResourcePoolHelper<H264Frame> {
public:

23
src/Extension/H264Rtp.cpp
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@ -90,7 +90,24 @@ bool H264RtpDecoder::decodeRtp(const RtpPacket::Ptr &rtppack) {
* Type==7:SPS frame
* Type==8:PPS frame
*/
/*
RTF3984 5.2节 Common Structure of the RTP Payload Format
Table 1. Summary of NAL unit types and their payload structures
Type Packet Type name Section
---------------------------------------------------------
0 undefined -
1-23 NAL unit Single NAL unit packet per H.264 5.6
24 STAP-A Single-time aggregation packet 5.7.1
25 STAP-B Single-time aggregation packet 5.7.1
26 MTAP16 Multi-time aggregation packet 5.7.2
27 MTAP24 Multi-time aggregation packet 5.7.2
28 FU-A Fragmentation unit 5.8
29 FU-B Fragmentation unit 5.8
30-31 undefined -
*/
const uint8_t *frame = (uint8_t *) rtppack->data() + rtppack->offset;
int length = rtppack->size() - rtppack->offset;
NALU nal;
@ -145,7 +162,7 @@ bool H264RtpDecoder::decodeRtp(const RtpPacket::Ptr &rtppack) {
FU fu;
MakeFU(frame[1], fu);
if (fu.S) {
//该帧的第一个rtp包
//该帧的第一个rtp包 FU-A start
char tmp = (nal.forbidden_zero_bit << 7 | nal.nal_ref_idc << 5 | fu.type);
_h264frame->buffer.assign("\x0\x0\x0\x1", 4);
_h264frame->buffer.push_back(tmp);
@ -164,14 +181,14 @@ bool H264RtpDecoder::decodeRtp(const RtpPacket::Ptr &rtppack) {
}
if (!fu.E) {
//该帧的中间rtp包
//该帧的中间rtp包 FU-A mid
_h264frame->buffer.append((char *)frame + 2, length - 2);
//该函数return时，保存下当前sequence,以便下次对比seq是否连续
_lastSeq = rtppack->sequence;
return false;
}
//该帧最后一个rtp包
//该帧最后一个rtp包 FU-A end
_h264frame->buffer.append((char *)frame + 2, length - 2);
_h264frame->timeStamp = rtppack->timeStamp;
auto key = _h264frame->keyFrame();

2
src/Extension/H264Rtp.h
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@ -36,6 +36,8 @@ namespace mediakit{
/**
* h264 rtp解码类
* 将 h264 over rtsp-rtp 解复用出 h264-Frame
* rfc3984
*/
class H264RtpDecoder : public RtpCodec , public ResourcePoolHelper<H264Frame> {
public:

2
src/Extension/H265Rtp.h
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@ -37,6 +37,8 @@ namespace mediakit{
/**
* h265 rtp解码类
* 将 h265 over rtsp-rtp 解复用出 h265-Frame
* 《草案（H265-over-RTP）draft-ietf-payload-rtp-h265-07.pdf》
*/
class H265RtpDecoder : public RtpCodec , public ResourcePoolHelper<H265Frame> {
public:

1
src/Http/HttpSession.cpp
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@ -490,6 +490,7 @@ inline void HttpSession::Handle_Req_GET(int64_t &content_len) {
//再看看是否为http-flv直播请求
if(checkLiveFlvStream()){
//若是，return！
return;
}

6
src/Rtmp/RtmpPlayer.h
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@ -44,7 +44,7 @@ using namespace toolkit;
using namespace mediakit::Client;
namespace mediakit {
//实现了rtmp播放器协议部分的功能，及数据接收功能
class RtmpPlayer:public PlayerBase, public TcpClient, public RtmpProtocol{
public:
typedef std::shared_ptr<RtmpPlayer> Ptr;
@ -63,11 +63,11 @@ protected:
void onMediaData_l(const RtmpPacket::Ptr &chunkData);
void onPlayResult_l(const SockException &ex);
//for Tcpclient
//form Tcpclient
void onRecv(const Buffer::Ptr &pBuf) override;
void onConnect(const SockException &err) override;
void onErr(const SockException &ex) override;
//fro RtmpProtocol
//from RtmpProtocol
void onRtmpChunk(RtmpPacket &chunkData) override;
void onStreamDry(uint32_t ui32StreamId) override;
void onSendRawData(const Buffer::Ptr &buffer) override{

2
src/Rtsp/RtspMuxer.cpp
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@ -39,7 +39,7 @@ RtspMuxer::RtspMuxer(const TitleSdp::Ptr &title){
}
void RtspMuxer::onTrackReady(const Track::Ptr &track) {
//根据track生产sdp
//根据track生成sdp
Sdp::Ptr sdp = track->getSdp();
if (!sdp) {
return;

2
src/Rtsp/RtspPlayer.h
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@ -46,7 +46,7 @@ using namespace toolkit;
namespace mediakit {
//实现了rtsp播放器协议部分的功能
//实现了rtsp播放器协议部分的功能，及数据接收功能
class RtspPlayer: public PlayerBase,public TcpClient, public RtspSplitter, public RtpReceiver {
public:
typedef std::shared_ptr<RtspPlayer> Ptr;

3
src/Rtsp/RtspSession.cpp
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@ -361,10 +361,11 @@ void RtspSession::onAuthSuccess() {
strongSelf->shutdown(SockException(Err_shutdown,err));
return;
}
//找到了响应的rtsp流
//找到了相应的rtsp流
strongSelf->_aTrackInfo = SdpParser(rtsp_src->getSdp()).getAvailableTrack();
if (strongSelf->_aTrackInfo.empty()) {
//该流无效
DebugL << "无trackInfo，该流无效";
strongSelf->send_StreamNotFound();
strongSelf->shutdown(SockException(Err_shutdown,"can not find any availabe track in sdp"));
return;