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优化代码

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This commit is contained in:
ziyue 2021-05-12 15:40:40 +08:00
parent cb280e1eeb
commit 12be56493e
2 changed files with 40 additions and 45 deletions
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72
webrtc/WebRtcTransport.cpp
View File

@ -414,27 +414,26 @@ 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];
ref.plan = &plan;
ref.media = m_with_ssrc;
ref.is_common_rtp = getCodecId(plan.codec) != CodecInvalid;
if (ref.is_common_rtp) {
auto info = std::make_shared<RtpPayloadInfo>();
_rtp_info_pt.emplace(plan.pt, info);
info->plan = &plan;
info->media = m_with_ssrc;
info->is_common_rtp = getCodecId(plan.codec) != CodecInvalid;
if (info->is_common_rtp) {
//rtp
_rtp_info_ssrc[m_with_ssrc->rtp_rtx_ssrc[0].ssrc] = &ref;
_rtp_info_ssrc[m_with_ssrc->rtp_rtx_ssrc[0].ssrc] = info;
} else {
//rtx
auto apt = atoi(plan.getFmtp("apt").data());
ref.plan_apt = m_with_ssrc->getPlan(apt);
info->plan_apt = m.getPlan(apt);
}
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) mutable{
onSortedRtp(ref, std::move(rtp));
}, [&ref, this](const RtpPacket::Ptr &rtp) mutable {
onBeforeSortedRtp(ref, rtp);
info->rtcp_context_recv = std::make_shared<RtcpContext>(info->plan->sample_rate, true);
info->rtcp_context_send = std::make_shared<RtcpContext>(info->plan->sample_rate, false);
info->receiver = std::make_shared<RtpReceiverImp>([info, this](RtpPacket::Ptr rtp) mutable {
onSortedRtp(*info, std::move(rtp));
});
ref.nack_ctx.setOnNack([&ref, this](const FCI_NACK &nack) mutable{
onNack(ref, nack);
info->nack_ctx.setOnNack([info, this](const FCI_NACK &nack) mutable {
onNack(*info, nack);
});
}
if (m.type != TrackApplication) {
@ -475,7 +474,7 @@ void WebRtcTransportImp::onStartWebRTC() {
auto it = _rtp_info_pt.find(m.plan[0].pt);
CHECK(it != _rtp_info_pt.end());
//记录发送rtp时约定的信息，届时发送rtp时需要修改pt和ssrc
_send_rtp_info[m.type] = &it->second;
_send_rtp_info[m.type] = it->second;
}
}
}
@ -564,9 +563,8 @@ SdpAttrCandidate::Ptr WebRtcTransportImp::getIceCandidate() const{
class RtpReceiverImp : public RtpReceiver {
public:
RtpReceiverImp( function<void(RtpPacket::Ptr rtp)> cb, function<void(const RtpPacket::Ptr &rtp)> cb_before = nullptr){
RtpReceiverImp( function<void(RtpPacket::Ptr rtp)> cb){
_on_sort = std::move(cb);
_on_before_sort = std::move(cb_before);
}
~RtpReceiverImp() override = default;
@ -580,15 +578,8 @@ protected:
_on_sort(std::move(rtp));
}
void onBeforeRtpSorted(const RtpPacket::Ptr &rtp, int track_index) override {
if (_on_before_sort) {
_on_before_sort(rtp);
}
}
private:
function<void(RtpPacket::Ptr rtp)> _on_sort;
function<void(const RtpPacket::Ptr &rtp)> _on_before_sort;
};
void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
@ -695,8 +686,11 @@ void WebRtcTransportImp::onRtp(const char *buf, size_t len) {
}
void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) {
if (!rtx) {
_bytes_usage += len;
_alive_ticker.resetTime();
}
RtpHeader *rtp = (RtpHeader *) buf;
//根据接收到的rtp的pt信息，找到该流的信息
auto it = _rtp_info_pt.find(rtp->pt);
@ -705,11 +699,11 @@ void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) {
return;
}
auto &info = it->second;
if (info.is_common_rtp) {
if (info->is_common_rtp) {
//这是普通的rtp数据
auto seq = ntohs(rtp->seq);
#if 0
if (!rtx && info.media->type == TrackVideo && seq % 100 == 0) {
if (!rtx && info->media->type == TrackVideo && seq % 100 == 0) {
//此处模拟接受丢包
DebugL << "recv dropped:" << seq;
return;
@ -717,10 +711,16 @@ void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) {
#endif
if (!rtx) {
//统计rtp接受情况，便于生成nack rtcp包
info.nack_ctx.received(seq);
info->nack_ctx.received(seq);
//修改ext id至统一
changeRtpExtId(rtp, _rtp_ext_id_to_type);
//时间戳转换成毫秒
auto stamp_ms = ntohl(rtp->stamp) * uint64_t(1000) / info->plan->sample_rate;
//统计rtp收到的情况，好做rr汇报
info->rtcp_context_recv->onRtp(seq, stamp_ms, len);
}
//解析并排序rtp
info.receiver->inputRtp(info.media->type, info.plan->sample_rate, (uint8_t *) buf, len);
info->receiver->inputRtp(info->media->type, info->plan->sample_rate, (uint8_t *) buf, len);
return;
}
@ -735,8 +735,8 @@ void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) {
auto origin_seq = payload[0] << 8 | payload[1];
InfoL << "received rtx rtp: " << origin_seq;
rtp->seq = htons(origin_seq);
rtp->ssrc = htonl(info.media->rtp_rtx_ssrc[0].ssrc);
rtp->pt = info.plan_apt->pt;
rtp->ssrc = htonl(info->media->rtp_rtx_ssrc[0].ssrc);
rtp->pt = info->plan_apt->pt;
memmove((uint8_t *) buf + 2, buf, payload - (uint8_t *) buf);
buf += 2;
len -= 2;
@ -752,12 +752,6 @@ void WebRtcTransportImp::onNack(RtpPayloadInfo &info, const FCI_NACK &nack) {
///////////////////////////////////////////////////////////////////
void WebRtcTransportImp::onBeforeSortedRtp(RtpPayloadInfo &info, const RtpPacket::Ptr &rtp) {
changeRtpExtId(rtp->getHeader(), _rtp_ext_id_to_type);
//统计rtp收到的情况，好做rr汇报
info.rtcp_context_recv->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
}
void WebRtcTransportImp::onSortedRtp(RtpPayloadInfo &info, 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
@ -779,7 +773,7 @@ void WebRtcTransportImp::onSortedRtp(RtpPayloadInfo &info, RtpPacket::Ptr rtp) {
///////////////////////////////////////////////////////////////////
void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool rtx){
auto info = _send_rtp_info[rtp->type];
auto &info = _send_rtp_info[rtp->type];
if (!info) {
//忽略，对方不支持该编码类型
return;
@ -798,7 +792,7 @@ void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool r
} else {
WarnL << "send rtx rtp:" << rtp->getSeq();
}
sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, info);
sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, info.get());
_bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize;
}

9
webrtc/WebRtcTransport.h
View File

@ -342,6 +342,8 @@ private:
class RtpPayloadInfo {
public:
using Ptr = std::shared_ptr<RtpPayloadInfo>;
bool is_common_rtp;
const RtcCodecPlan *plan;
const RtcCodecPlan *plan_apt;
@ -354,7 +356,6 @@ private:
};
void onSortedRtp(RtpPayloadInfo &info, RtpPacket::Ptr rtp);
void onBeforeSortedRtp(RtpPayloadInfo &info, const RtpPacket::Ptr &rtp);
void onNack(RtpPayloadInfo &info, const FCI_NACK &nack);
private:
@ -371,7 +372,7 @@ private:
//pli rtcp计时器
Ticker _pli_ticker;
//记录协商的发送rtp的pt和ssrc
RtpPayloadInfo* _send_rtp_info[2] = {nullptr, nullptr};
RtpPayloadInfo::Ptr _send_rtp_info[2];
//复合udp端口，接收一切rtp与rtcp
Socket::Ptr _socket;
//推流的rtsp源
@ -381,9 +382,9 @@ private:
//播放rtsp源的reader对象
RtspMediaSource::RingType::RingReader::Ptr _reader;
//根据rtp的pt获取相关信息
unordered_map<uint8_t/*pt*/, RtpPayloadInfo> _rtp_info_pt;
unordered_map<uint8_t/*pt*/, RtpPayloadInfo::Ptr> _rtp_info_pt;
//根据rtcp的ssrc获取相关信息
unordered_map<uint32_t/*ssrc*/, RtpPayloadInfo*> _rtp_info_ssrc;
unordered_map<uint32_t/*ssrc*/, RtpPayloadInfo::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