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rtc播放支持nack重传

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This commit is contained in:
xiongziliang 2021-05-11 00:54:33 +08:00
parent e8d3dec050
commit 7ad361b22d
2 changed files with 122 additions and 19 deletions
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58
webrtc/WebRtcTransport.cpp
View File

@ -269,21 +269,22 @@ void WebRtcTransport::inputSockData(char *buf, size_t len, RTC::TransportTuple *
}
}
void WebRtcTransport::sendRtpPacket(char *buf, size_t len, bool flush, TrackType type) {
void WebRtcTransport::sendRtpPacket(char *buf, size_t len, bool flush, void *ctx) {
if (_srtp_session_send) {
CHECK(len + SRTP_MAX_TRAILER_LEN <= sizeof(_srtp_buf));
memcpy(_srtp_buf, buf, len);
onBeforeEncryptRtp((char *) _srtp_buf, len, type);
onBeforeEncryptRtp((char *) _srtp_buf, len, ctx);
if (_srtp_session_send->EncryptRtp(_srtp_buf, &len)) {
onSendSockData((char *) _srtp_buf, len, flush);
}
}
}
void WebRtcTransport::sendRtcpPacket(char *buf, size_t len, bool flush){
void WebRtcTransport::sendRtcpPacket(char *buf, size_t len, bool flush, void *ctx){
if (_srtp_session_send) {
CHECK(len + SRTP_MAX_TRAILER_LEN <= sizeof(_srtp_buf));
memcpy(_srtp_buf, buf, len);
onBeforeEncryptRtcp((char *) _srtp_buf, len, ctx);
if (_srtp_session_send->EncryptRtcp(_srtp_buf, &len)) {
onSendSockData((char *) _srtp_buf, len, flush);
}
@ -414,10 +415,13 @@ void WebRtcTransportImp::onStartWebRTC() {
auto m_with_ssrc = getSdpWithSSRC().getMedia(m.type);
//获取offer端rtp的ssrc和pt相关信息
auto &ref = _rtp_info_pt[plan.pt];
_rtp_info_ssrc[m_with_ssrc->rtp_rtx_ssrc[0].ssrc] = &ref;
ref.plan = &plan;
ref.media = m_with_ssrc;
ref.is_common_rtp = getCodecId(plan.codec) != CodecInvalid;
if (ref.is_common_rtp) {
//rtp
_rtp_info_ssrc[m_with_ssrc->rtp_rtx_ssrc[0].ssrc] = &ref;
}
ref.rtcp_context_recv = std::make_shared<RtcpContext>(ref.plan->sample_rate, true);
ref.rtcp_context_send = std::make_shared<RtcpContext>(ref.plan->sample_rate, false);
ref.receiver = std::make_shared<RtpReceiverImp>([&ref, this](RtpPacket::Ptr rtp) {
@ -628,7 +632,30 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
}
case RtcpType::RTCP_PSFB:
case RtcpType::RTCP_RTPFB: {
// DebugL << "\r\n" << rtcp->dumpString();
RtcpFB *fb = (RtcpFB *) rtcp;
auto it = _rtp_info_ssrc.find(fb->ssrc);
if (it == _rtp_info_ssrc.end()) {
WarnL << "未识别的 rtcp包:" << rtcp->dumpString();
return;
}
if ((RtcpType) rtcp->pt == RtcpType::RTCP_PSFB) {
// DebugL << "\r\n" << rtcp->dumpString();
break;
}
//RTPFB
switch ((RTPFBType) rtcp->report_count) {
case RTPFBType::RTCP_RTPFB_NACK : {
auto &fci = fb->getFci<FCI_NACK>();
it->second->nack_list.for_each_nack(fci, [&](const RtpPacket::Ptr &rtp) {
//rtp重传
onSendRtp(rtp, true, true);
});
break;
}
default:
// DebugL << "\r\n" << rtcp->dumpString();
break;
}
break;
}
default: break;
@ -703,27 +730,36 @@ void WebRtcTransportImp::onBeforeSortedRtp(const RtpPayloadInfo &info, const Rtp
info.rtcp_context_recv->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
}
void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){
void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool rtx){
auto info = _send_rtp_info[rtp->type];
if (!info) {
//忽略，对方不支持该编码类型
return;
}
if (!rtx) {
//统计rtp发送情况，好做sr汇报
info->rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
info->nack_list.push_back(rtp);
} else {
WarnL << "重传rtp:" << rtp->getSeq();
}
sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, info);
_bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize;
sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, rtp->type);
//统计rtp发送情况，好做sr汇报
info->rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
}
void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, size_t len, TrackType type) {
void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, size_t len, void *ctx) {
RtpPayloadInfo *info = reinterpret_cast<RtpPayloadInfo *>(ctx);
auto header = (RtpHeader *)buf;
auto info = _send_rtp_info[type];
//修改目标pt和ssrc
header->pt = info->plan->pt;
header->ssrc = htons(info->media->rtp_rtx_ssrc[0].ssrc);
changeRtpExtId(header, _rtp_ext_type_to_id);
}
void WebRtcTransportImp::onBeforeEncryptRtcp(const char *buf, size_t len, void *ctx) {
}
void WebRtcTransportImp::onShutdown(const SockException &ex){
InfoL << ex.what();
_self = nullptr;

83
webrtc/WebRtcTransport.h
View File

@ -21,6 +21,7 @@
#include "Network/Socket.h"
#include "Rtsp/RtspMediaSourceImp.h"
#include "Rtcp/RtcpContext.h"
#include "Rtcp/RtcpFCI.h"
using namespace toolkit;
using namespace mediakit;
@ -60,10 +61,10 @@ public:
* @param buf rtcp内容
* @param len rtcp长度
* @param flush 是否flush socket
* @param type rtp类型
* @param ctx 用户指针
*/
void sendRtpPacket(char *buf, size_t len, bool flush, TrackType type);
void sendRtcpPacket(char *buf, size_t len, bool flush);
void sendRtpPacket(char *buf, size_t len, bool flush, void *ctx = nullptr);
void sendRtcpPacket(char *buf, size_t len, bool flush, void *ctx = nullptr);
const EventPoller::Ptr& getPoller() const;
@ -100,7 +101,8 @@ protected:
virtual void onRtp(const char *buf, size_t len) = 0;
virtual void onRtcp(const char *buf, size_t len) = 0;
virtual void onShutdown(const SockException &ex) = 0;
virtual void onBeforeEncryptRtp(const char *buf, size_t len, TrackType type) = 0;
virtual void onBeforeEncryptRtp(const char *buf, size_t len, void *ctx) = 0;
virtual void onBeforeEncryptRtcp(const char *buf, size_t len, void *ctx) = 0;
protected:
const RtcSession& getSdp(SdpType type) const;
@ -125,6 +127,69 @@ private:
class RtpReceiverImp;
class NackList {
public:
void push_back(RtpPacket::Ptr rtp) {
auto seq = rtp->getSeq();
nack_cache_seq.emplace_back(seq);
nack_cache_pkt.emplace(seq, std::move(rtp));
while (get_cache_ms() > kMaxNackMS) {
//需要清除部分nack缓存
pop_front();
}
}
template<typename FUNC>
void for_each_nack(const FCI_NACK &nack, const FUNC &func) {
auto seq = nack.getPid();
for (auto bit : nack.getBitArray()) {
if (!bit) {
//丢包
RtpPacket::Ptr *ptr = get_rtp(seq);
if (ptr) {
func(*ptr);
}
}
++seq;
}
}
private:
void pop_front() {
if (nack_cache_seq.empty()) {
return;
}
nack_cache_pkt.erase(nack_cache_seq.front());
nack_cache_seq.pop_front();
}
RtpPacket::Ptr *get_rtp(uint16_t seq) {
auto it = nack_cache_pkt.find(seq);
if (it == nack_cache_pkt.end()) {
return nullptr;
}
return &it->second;
}
uint32_t get_cache_ms() {
if (nack_cache_seq.size() < 2) {
return 0;
}
uint32_t back = nack_cache_pkt[nack_cache_seq.back()]->getStampMS();
uint32_t front = nack_cache_pkt[nack_cache_seq.front()]->getStampMS();
if (back > front) {
return back - front;
}
//很有可能回环了
return back + (UINT32_MAX - front);
}
private:
static constexpr uint32_t kMaxNackMS = 10 * 1000;
deque<uint16_t> nack_cache_seq;
unordered_map<uint16_t, RtpPacket::Ptr > nack_cache_pkt;
};
class WebRtcTransportImp : public WebRtcTransport, public MediaSourceEvent, public SockInfo, public std::enable_shared_from_this<WebRtcTransportImp>{
public:
using Ptr = std::shared_ptr<WebRtcTransportImp>;
@ -152,7 +217,8 @@ protected:
void onRtp(const char *buf, size_t len) override;
void onRtcp(const char *buf, size_t len) override;
void onBeforeEncryptRtp(const char *buf, size_t len, TrackType type) override;
void onBeforeEncryptRtp(const char *buf, size_t len, void *ctx) override;
void onBeforeEncryptRtcp(const char *buf, size_t len, void *ctx) override;
void onShutdown(const SockException &ex) override;
@ -184,7 +250,7 @@ private:
WebRtcTransportImp(const EventPoller::Ptr &poller);
void onCreate() override;
void onDestory() override;
void onSendRtp(const RtpPacket::Ptr &rtp, bool flush);
void onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool rtx = false);
SdpAttrCandidate::Ptr getIceCandidate() const;
bool canSendRtp() const;
bool canRecvRtp() const;
@ -198,10 +264,11 @@ private:
std::shared_ptr<RtpReceiverImp> receiver;
RtcpContext::Ptr rtcp_context_recv;
RtcpContext::Ptr rtcp_context_send;
NackList nack_list;
};
void onSortedRtp(const RtpPayloadInfo &info,RtpPacket::Ptr rtp);
void onBeforeSortedRtp(const RtpPayloadInfo &info,const RtpPacket::Ptr &rtp);
void onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr rtp);
void onBeforeSortedRtp(const RtpPayloadInfo &info, const RtpPacket::Ptr &rtp);
private:
//用掉的总流量