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ZLMediaKit
Commit 6c01cf33 by ziyue
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抽象MediaTrack与RtpChannel对象

parent ba672178
正在显示 包含 92 行增加 和 78 行删除
webrtc/WebRtcTransport.cpp
......@@ -148,7 +148,6 @@ void WebRtcTransport::sendRtcpRemb(uint32_t ssrc, size_t bit_rate) {
fb->ssrc = htonl(0);
fb->ssrc_media = htonl(ssrc);
sendRtcpPacket((char *) fb.get(), fb->getSize(), true);
TraceL << ssrc << " " << bit_rate;
}
void WebRtcTransport::sendRtcpPli(uint32_t ssrc) {
......@@ -400,7 +399,7 @@ void WebRtcTransportImp::onStartWebRTC() {
//获取ssrc和pt相关信息,届时收到rtp和rtcp时分别可以根据pt和ssrc找到相关的信息
for (auto &m_answer : getSdp(SdpType::answer).media) {
auto m_offer = getSdp(SdpType::offer).getMedia(m_answer.type);
auto info = std::make_shared<RtpPayloadInfo>();
auto info = std::make_shared<MediaTrack>();
info->media = &m_answer;
info->answer_ssrc_rtp = m_answer.getRtpSSRC();
......@@ -411,16 +410,16 @@ void WebRtcTransportImp::onStartWebRTC() {
info->plan_rtx = m_answer.getRelatedRtxPlan(info->plan_rtp->pt);
info->rtcp_context_send = std::make_shared<RtcpContext>(info->plan_rtp->sample_rate, false);
//send ssrc --> RtpPayloadInfo
//send ssrc --> MediaTrack
_rtp_info_ssrc[info->answer_ssrc_rtp] = info;
//recv ssrc --> RtpPayloadInfo
//recv ssrc --> MediaTrack
_rtp_info_ssrc[info->offer_ssrc_rtp] = info;
//rtp pt --> RtpPayloadInfo
//rtp pt --> MediaTrack
_rtp_info_pt.emplace(info->plan_rtp->pt, std::make_pair(false, info));
if (info->plan_rtx) {
//rtx pt --> RtpPayloadInfo
//rtx pt --> MediaTrack
_rtp_info_pt.emplace(info->plan_rtx->pt, std::make_pair(true, info));
}
if (m_offer->type != TrackApplication) {
......@@ -557,16 +556,29 @@ SdpAttrCandidate::Ptr WebRtcTransportImp::getIceCandidate() const{
///////////////////////////////////////////////////////////////////
class RtpReceiverImp : public RtpReceiver {
class RtpChannel : public RtpReceiver {
public:
RtpReceiverImp( function<void(RtpPacket::Ptr rtp)> cb){
_on_sort = std::move(cb);
uint32_t ssrc;
RtcpContext::Ptr rtcp_context;
public:
RtpChannel(function<void(RtpPacket::Ptr rtp)> on_rtp, function<void(const FCI_NACK &nack)> on_nack) {
_on_sort = std::move(on_rtp);
nack_ctx.setOnNack(std::move(on_nack));
}
~RtpReceiverImp() override = default;
~RtpChannel() override = default;
bool inputRtp(TrackType type, int samplerate, uint8_t *ptr, size_t len){
return handleOneRtp((int) type, type, samplerate, ptr, len);
bool inputRtp(TrackType type, int sample_rate, uint8_t *ptr, size_t len, bool is_rtx){
if (!is_rtx) {
RtpHeader *rtp = (RtpHeader *) ptr;
auto seq = ntohs(rtp->seq);
//统计rtp接受情况，便于生成nack rtcp包
nack_ctx.received(seq);
//统计rtp收到的情况，好做rr汇报
rtcp_context->onRtp(seq, ntohl(rtp->stamp) * uint64_t(1000) / sample_rate, len);
}
return handleOneRtp((int) type, type, sample_rate, ptr, len);
}
protected:
......@@ -575,9 +587,22 @@ protected:
}
private:
NackContext nack_ctx;
function<void(RtpPacket::Ptr rtp)> _on_sort;
};
std::shared_ptr<RtpChannel> WebRtcTransportImp::MediaTrack::getRtpChannel(uint32_t ssrc) const{
auto it_rid = ssrc_to_rid.find(ssrc);
if (it_rid == ssrc_to_rid.end()) {
return nullptr;
}
auto it_chn = rtp_channel.find(it_rid->second);
if (it_chn == rtp_channel.end()) {
return nullptr;
}
return it_chn->second;
}
void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
_bytes_usage += len;
auto rtcps = RtcpHeader::loadFromBytes((char *) buf, len);
......@@ -589,13 +614,13 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
auto it = _rtp_info_ssrc.find(sr->ssrc);
if (it != _rtp_info_ssrc.end()) {
auto &info = it->second;
auto it = info->rtcp_context_recv.find(sr->ssrc);
if (it != info->rtcp_context_recv.end()) {
it->second->onRtcp(sr);
auto rr = it->second->createRtcpRR(info->answer_ssrc_rtp, sr->ssrc);
sendRtcpPacket(rr->data(), rr->size(), true);
} else {
auto rtp_chn = info->getRtpChannel(sr->ssrc);
if(!rtp_chn){
WarnL << "未识别的sr rtcp包:" << rtcp->dumpString();
} else {
rtp_chn->rtcp_context->onRtcp(sr);
auto rr = rtp_chn->rtcp_context->createRtcpRR(info->answer_ssrc_rtp, sr->ssrc);
sendRtcpPacket(rr->data(), rr->size(), true);
}
} else {
WarnL << "未识别的sr rtcp包:" << rtcp->dumpString();
......@@ -665,7 +690,7 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
///////////////////////////////////////////////////////////////////
void WebRtcTransportImp::changeRtpExtId(RtpPayloadInfo &info, const RtpHeader *header, bool is_recv, string *rid_ptr) const{
void WebRtcTransportImp::changeRtpExtId(MediaTrack &info, const RtpHeader *header, bool is_recv, string *rid_ptr) const{
string rid, repaired_rid;
auto ext_map = RtpExt::getExtValue(header);
for (auto &pr : ext_map) {
......@@ -709,29 +734,30 @@ void WebRtcTransportImp::changeRtpExtId(RtpPayloadInfo &info, const RtpHeader *h
rid = info.ssrc_to_rid[ssrc];
} else {
//设置rid
info.ssrc_to_rid[ssrc] = rid;
if (info.ssrc_to_rid.emplace(ssrc, rid).second) {
InfoL << "rid of ssrc " << ssrc << " is:" << rid;
}
}
if (rid_ptr) {
*rid_ptr = rid;
}
}
std::shared_ptr<RtpReceiverImp> WebRtcTransportImp::createRtpReceiver(const string &rid, uint32_t ssrc, bool is_rtx, const RtpPayloadInfo::Ptr &info){
auto ref = std::make_shared<RtpReceiverImp>([info, this, rid](RtpPacket::Ptr rtp) mutable {
void WebRtcTransportImp::createRtpChannel(const string &rid, uint32_t ssrc, const MediaTrack::Ptr &info) {
//rid --> RtpReceiverImp
auto &ref = info->rtp_channel[rid];
ref = std::make_shared<RtpChannel>([info, this, rid](RtpPacket::Ptr rtp) mutable {
onSortedRtp(*info, rid, std::move(rtp));
}, [info, this, ssrc](const FCI_NACK &nack) mutable {
onSendNack(*info, nack, ssrc);
});
if (!is_rtx) {
//rtx没nack
info->nack_ctx[rid].setOnNack([info, this, ssrc](const FCI_NACK &nack) mutable {
onSendNack(*info, nack, ssrc);
});
//ssrc --> RtpPayloadInfo
_rtp_info_ssrc[ssrc] = info;
}
InfoL << "receive rtp of ssrc:" << ssrc << ", rid:" << rid << ", is rtx:" << is_rtx << ", codec:" << info->plan_rtp->codec;
return ref;
//rid --> rtp ssrc
ref->ssrc = ssrc;
//rtp ssrc --> RtcpContext
ref->rtcp_context = std::make_shared<RtcpContext>(info->plan_rtp->sample_rate, true);
//rtp ssrc --> MediaTrack
_rtp_info_ssrc[ssrc] = info;
InfoL << "create rtp receiver of ssrc:" << ssrc << ", rid:" << rid << ", codec:" << info->plan_rtp->codec;
}
void WebRtcTransportImp::onRtp(const char *buf, size_t len) {
......@@ -758,36 +784,27 @@ void WebRtcTransportImp::onRtp(const char *buf, size_t len) {
WarnL << "ssrc:" << ssrc << ", rtx:" << is_rtx << ",seq:" << ntohs((uint16_t) rtp->seq);
}
#endif
auto &ref = info->receiver[rid];
auto &ref = info->rtp_channel[rid];
if (!ref) {
ref = createRtpReceiver(rid, ssrc, is_rtx, info);
if (is_rtx) {
WarnL << "dropped rtx rtp, rid:" << rid << ", ssrc:" << ssrc << ", codec:" << info->plan_rtp->codec << ", seq:" << ntohs(rtp->seq);
return;
}
createRtpChannel(rid, ssrc, info);
}
if (!is_rtx) {
//这是普通的rtp数据
auto seq = ntohs(rtp->seq);
#if 0
auto seq = ntohs(rtp->seq);
if (info->media->type == TrackVideo && seq % 100 == 0) {
//此处模拟接受丢包
DebugL << "recv dropped:" << seq;
DebugL << "recv dropped:" << seq << ", rid:" << rid << ", ssrc:" << ssrc;
return;
}
#endif
//统计rtp接受情况，便于生成nack rtcp包
info->nack_ctx[rid].received(seq);
auto &cxt_ref = info->rtcp_context_recv[ssrc];
if (!cxt_ref) {
cxt_ref = std::make_shared<RtcpContext>(info->plan_rtp->sample_rate, true);
info->rid_to_ssrc[rid] = ssrc;
}
//时间戳转换成毫秒
auto stamp_ms = ntohl(rtp->stamp) * uint64_t(1000) / info->plan_rtp->sample_rate;
//统计rtp收到的情况，好做rr汇报
cxt_ref->onRtp(seq, stamp_ms, len);
//解析并排序rtp
ref->inputRtp(info->media->type, info->plan_rtp->sample_rate, (uint8_t *) buf, len);
ref->inputRtp(info->media->type, info->plan_rtp->sample_rate, (uint8_t *) buf, len, false);
return;
}
......@@ -801,19 +818,19 @@ void WebRtcTransportImp::onRtp(const char *buf, size_t len) {
//前两个字节是原始的rtp的seq
auto origin_seq = payload[0] << 8 | payload[1];
//rtx的seq转换为rtp的seq
rtp->seq = htons(origin_seq);
//rtx的ssrc转换为rtp的ssrc
rtp->ssrc = htonl(info->rid_to_ssrc[rid]);
//rtx的pt转换为rtp的pt
//InfoL << "rtx rtp, rid:" << rid << ", seq:" << origin_seq << ", ssrc:" << ssrc;
//rtx 转换为 rtp
rtp->pt = info->plan_rtp->pt;
rtp->seq = htons(origin_seq);
rtp->ssrc = htonl(ref->ssrc);
memmove((uint8_t *) buf + 2, buf, payload - (uint8_t *) buf);
buf += 2;
len -= 2;
ref->inputRtp(info->media->type, info->plan_rtp->sample_rate, (uint8_t *) buf, len);
ref->inputRtp(info->media->type, info->plan_rtp->sample_rate, (uint8_t *) buf, len, true);
}
void WebRtcTransportImp::onSendNack(RtpPayloadInfo &info, const FCI_NACK &nack, uint32_t ssrc) {
void WebRtcTransportImp::onSendNack(MediaTrack &info, const FCI_NACK &nack, uint32_t ssrc) {
auto rtcp = RtcpFB::create(RTPFBType::RTCP_RTPFB_NACK, &nack, FCI_NACK::kSize);
rtcp->ssrc = htons(info.answer_ssrc_rtp);
rtcp->ssrc_media = htonl(ssrc);
......@@ -823,7 +840,7 @@ void WebRtcTransportImp::onSendNack(RtpPayloadInfo &info, const FCI_NACK &nack,
///////////////////////////////////////////////////////////////////
void WebRtcTransportImp::onSortedRtp(RtpPayloadInfo &info, const string &rid, RtpPacket::Ptr rtp) {
void WebRtcTransportImp::onSortedRtp(MediaTrack &info, const string &rid, RtpPacket::Ptr rtp) {
if (info.media->type == TrackVideo && _pli_ticker.elapsedTime() > 2000) {
//定期发送pli请求关键帧，方便非rtc等协议
_pli_ticker.resetTime();
......@@ -880,13 +897,13 @@ void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool r
} else {
WarnL << "send rtx rtp:" << rtp->getSeq();
}
pair<bool/*rtx*/, RtpPayloadInfo *> ctx{rtx, info.get()};
pair<bool/*rtx*/, MediaTrack *> ctx{rtx, info.get()};
sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, &ctx);
_bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize;
}
void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, size_t &len, void *ctx) {
auto pr = (pair<bool/*rtx*/, RtpPayloadInfo *> *) ctx;
auto pr = (pair<bool/*rtx*/, MediaTrack *> *) ctx;
auto header = (RtpHeader *) buf;
if (!pr->first || !pr->second->plan_rtx) {
......
webrtc/WebRtcTransport.h
......@@ -125,7 +125,7 @@ private:
RtcSession::Ptr _answer_sdp;
};
class RtpReceiverImp;
class RtpChannel;
class NackList {
public:
......@@ -337,9 +337,9 @@ private:
bool canSendRtp() const;
bool canRecvRtp() const;
class RtpPayloadInfo {
class MediaTrack {
public:
using Ptr = std::shared_ptr<RtpPayloadInfo>;
using Ptr = std::shared_ptr<MediaTrack>;
const RtcCodecPlan *plan_rtp;
const RtcCodecPlan *plan_rtx;
uint32_t offer_ssrc_rtp = 0;
......@@ -349,19 +349,16 @@ private:
const RtcMedia *media;
NackList nack_list;
RtcpContext::Ptr rtcp_context_send;
//只生成rtp的nack，rtx的不生成
unordered_map<string/*rid*/, NackContext> nack_ctx;
unordered_map<string/*rid*/, uint32_t/*rtp ssrc*/> rid_to_ssrc;
unordered_map<string/*rid*/, std::shared_ptr<RtpReceiverImp> > receiver;
unordered_map<string/*rid*/, std::shared_ptr<RtpChannel> > rtp_channel;
unordered_map<uint32_t/*simulcast ssrc*/, string/*rid*/> ssrc_to_rid;
//只统计rtp的接收情况，rtx的不统计
unordered_map<uint32_t/*simulcast ssrc*/, RtcpContext::Ptr> rtcp_context_recv;
std::shared_ptr<RtpChannel> getRtpChannel(uint32_t ssrc) const;
};
void onSortedRtp(RtpPayloadInfo &info, const string &rid, RtpPacket::Ptr rtp);
void onSendNack(RtpPayloadInfo &info, const FCI_NACK &nack, uint32_t ssrc);
void changeRtpExtId(RtpPayloadInfo &info, const RtpHeader *header, bool is_recv, string *rid_ptr = nullptr) const;
std::shared_ptr<RtpReceiverImp> createRtpReceiver(const string &rid, uint32_t ssrc, bool is_rtx, const RtpPayloadInfo::Ptr &info);
void onSortedRtp(MediaTrack &track, const string &rid, RtpPacket::Ptr rtp);
void onSendNack(MediaTrack &track, const FCI_NACK &nack, uint32_t ssrc);
void changeRtpExtId(MediaTrack &track, const RtpHeader *header, bool is_recv, string *rid_ptr = nullptr) const;
void createRtpChannel(const string &rid, uint32_t ssrc, const MediaTrack::Ptr &info);
private:
uint16_t _rtx_seq[2] = {0, 0};
......@@ -387,11 +384,11 @@ private:
//播放rtsp源的reader对象
RtspMediaSource::RingType::RingReader::Ptr _reader;
//根据发送rtp的track类型获取相关信息
RtpPayloadInfo::Ptr _send_rtp_info[2];
MediaTrack::Ptr _send_rtp_info[2];
//根据接收rtp的pt获取相关信息
unordered_map<uint8_t/*pt*/, std::pair<bool/*is rtx*/,RtpPayloadInfo::Ptr> > _rtp_info_pt;
unordered_map<uint8_t/*pt*/, std::pair<bool/*is rtx*/,MediaTrack::Ptr> > _rtp_info_pt;
//根据rtcp的ssrc获取相关信息，只记录rtp的ssrc，rtx的ssrc不记录
unordered_map<uint32_t/*ssrc*/, RtpPayloadInfo::Ptr> _rtp_info_ssrc;
unordered_map<uint32_t/*ssrc*/, MediaTrack::Ptr> _rtp_info_ssrc;
//发送rtp时需要修改rtp ext id
map<RtpExtType, uint8_t> _rtp_ext_type_to_id;
//接收rtp时需要修改rtp ext id
......
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