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ZLMediaKit
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优化代码

This commit is contained in:
xiongziliang 2024-03-23 22:43:09 +08:00
parent f81f62c8ff
commit 73678a6e69
8 changed files with 86 additions and 96 deletions
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3
api/source/mk_common.cpp
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@ -305,8 +305,7 @@ API_EXPORT void API_CALL mk_webrtc_get_answer_sdp2(void *user_data, on_user_data
std::shared_ptr<void> ptr(user_data, user_data_free ? user_data_free : [](void *) {});
auto args = std::make_shared<WebRtcArgsUrl>(url);
WebRtcPluginManager::Instance().negotiateSdp(*session, type, *args, [offer_str, session, ptr, cb, args](const WebRtcInterface &exchanger) mutable {
setWebRtcArgs(*args, exchanger);
auto& handler = const_cast<WebRtcInterface &>(exchanger);
auto &handler = const_cast<WebRtcInterface &>(exchanger);
try {
auto sdp_answer = handler.getAnswerSdp(offer_str);
cb(ptr.get(), sdp_answer.data(), nullptr);

13
server/WebApi.cpp
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@ -1771,13 +1771,7 @@ void installWebApi() {
auto &session = static_cast<Session&>(sender);
auto args = std::make_shared<WebRtcArgsImp>(allArgs, sender.getIdentifier());
WebRtcPluginManager::Instance().negotiateSdp(session, type, *args, [invoker, val, offer, headerOut, args](const WebRtcInterface &exchanger) mutable {
//设置返回类型
headerOut["Content-Type"] = HttpFileManager::getContentType(".json");
//设置跨域
headerOut["Access-Control-Allow-Origin"] = "*";
setWebRtcArgs(*args, exchanger);
auto& handler = const_cast<WebRtcInterface &>(exchanger);
auto &handler = const_cast<WebRtcInterface &>(exchanger);
try {
val["sdp"] = handler.getAnswerSdp(offer);
val["id"] = exchanger.getIdentifier();
@ -1800,10 +1794,7 @@ void installWebApi() {
auto location = std::string(session.overSsl() ? "https://" : "http://") + allArgs["host"] + delete_webrtc_url;
auto args = std::make_shared<WebRtcArgsImp>(allArgs, sender.getIdentifier());
WebRtcPluginManager::Instance().negotiateSdp(session, type, *args, [invoker, offer, headerOut, location, args](const WebRtcInterface &exchanger) mutable {
// 设置跨域
headerOut["Access-Control-Allow-Origin"] = "*";
setWebRtcArgs(*args, exchanger);
auto& handler = const_cast<WebRtcInterface &>(exchanger);
auto &handler = const_cast<WebRtcInterface &>(exchanger);
try {
// 设置返回类型
headerOut["Content-Type"] = "application/sdp";

8
webrtc/WebRtcPlayer.cpp
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@ -17,9 +17,8 @@ namespace mediakit {
WebRtcPlayer::Ptr WebRtcPlayer::create(const EventPoller::Ptr &poller,
const RtspMediaSource::Ptr &src,
const MediaInfo &info,
bool preferred_tcp) {
WebRtcPlayer::Ptr ret(new WebRtcPlayer(poller, src, info, preferred_tcp), [](WebRtcPlayer *ptr) {
const MediaInfo &info) {
WebRtcPlayer::Ptr ret(new WebRtcPlayer(poller, src, info), [](WebRtcPlayer *ptr) {
ptr->onDestory();
delete ptr;
});
@ -29,8 +28,7 @@ WebRtcPlayer::Ptr WebRtcPlayer::create(const EventPoller::Ptr &poller,
WebRtcPlayer::WebRtcPlayer(const EventPoller::Ptr &poller,
const RtspMediaSource::Ptr &src,
const MediaInfo &info,
bool preferred_tcp) : WebRtcTransportImp(poller,preferred_tcp) {
const MediaInfo &info) : WebRtcTransportImp(poller) {
_media_info = info;
_play_src = src;
CHECK(src);

4
webrtc/WebRtcPlayer.h
View File

@ -19,7 +19,7 @@ namespace mediakit {
class WebRtcPlayer : public WebRtcTransportImp {
public:
using Ptr = std::shared_ptr<WebRtcPlayer>;
static Ptr create(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src, const MediaInfo &info, bool preferred_tcp = false);
static Ptr create(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src, const MediaInfo &info);
MediaInfo getMediaInfo() { return _media_info; }
protected:
@ -29,7 +29,7 @@ protected:
void onRtcConfigure(RtcConfigure &configure) const override;
private:
WebRtcPlayer(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src, const MediaInfo &info, bool preferred_tcp);
WebRtcPlayer(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src, const MediaInfo &info);
private:
//媒体相关元数据

8
webrtc/WebRtcPusher.cpp
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@ -20,9 +20,8 @@ WebRtcPusher::Ptr WebRtcPusher::create(const EventPoller::Ptr &poller,
const RtspMediaSource::Ptr &src,
const std::shared_ptr<void> &ownership,
const MediaInfo &info,
const ProtocolOption &option,
bool preferred_tcp) {
WebRtcPusher::Ptr ret(new WebRtcPusher(poller, src, ownership, info, option,preferred_tcp), [](WebRtcPusher *ptr) {
const ProtocolOption &option) {
WebRtcPusher::Ptr ret(new WebRtcPusher(poller, src, ownership, info, option), [](WebRtcPusher *ptr) {
ptr->onDestory();
delete ptr;
});
@ -34,8 +33,7 @@ WebRtcPusher::WebRtcPusher(const EventPoller::Ptr &poller,
const RtspMediaSource::Ptr &src,
const std::shared_ptr<void> &ownership,
const MediaInfo &info,
const ProtocolOption &option,
bool preferred_tcp) : WebRtcTransportImp(poller,preferred_tcp) {
const ProtocolOption &option) : WebRtcTransportImp(poller) {
_media_info = info;
_push_src = src;
_push_src_ownership = ownership;

5
webrtc/WebRtcPusher.h
View File

@ -20,8 +20,7 @@ class WebRtcPusher : public WebRtcTransportImp, public MediaSourceEvent {
public:
using Ptr = std::shared_ptr<WebRtcPusher>;
static Ptr create(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src,
const std::shared_ptr<void> &ownership, const MediaInfo &info, const ProtocolOption &option, bool preferred_tcp = false);
const std::shared_ptr<void> &ownership, const MediaInfo &info, const ProtocolOption &option);
protected:
///////WebRtcTransportImp override///////
@ -53,7 +52,7 @@ protected:
private:
WebRtcPusher(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src,
const std::shared_ptr<void> &ownership, const MediaInfo &info, const ProtocolOption &option, bool preferred_tcp);
const std::shared_ptr<void> &ownership, const MediaInfo &info, const ProtocolOption &option);
private:
bool _simulcast = false;

121
webrtc/WebRtcTransport.cpp
View File

@ -530,8 +530,7 @@ void WebRtcTransportImp::OnDtlsTransportApplicationDataReceived(const RTC::DtlsT
#endif
}
WebRtcTransportImp::WebRtcTransportImp(const EventPoller::Ptr &poller,bool preferred_tcp)
: WebRtcTransport(poller), _preferred_tcp(preferred_tcp) {
WebRtcTransportImp::WebRtcTransportImp(const EventPoller::Ptr &poller) : WebRtcTransport(poller) {
InfoL << getIdentifier();
}
@ -671,7 +670,7 @@ void WebRtcTransportImp::onCheckAnswer(RtcSession &sdp) {
});
for (auto &m : sdp.media) {
m.addr.reset();
m.addr.address = extern_ips.empty() ? _localIp.empty() ? SockUtil::get_local_ip() : _localIp : extern_ips[0];
m.addr.address = extern_ips.empty() ? _local_ip.empty() ? SockUtil::get_local_ip() : _local_ip : extern_ips[0];
m.rtcp_addr.reset();
m.rtcp_addr.address = m.addr.address;
@ -766,7 +765,7 @@ void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
return ret;
});
if (extern_ips.empty()) {
std::string local_ip = _localIp.empty() ? SockUtil::get_local_ip() : _localIp;
std::string local_ip = _local_ip.empty() ? SockUtil::get_local_ip() : _local_ip;
if (local_udp_port) { configure.addCandidate(*makeIceCandidate(local_ip, local_udp_port, 120, "udp")); }
if (local_tcp_port) { configure.addCandidate(*makeIceCandidate(local_ip, local_tcp_port, _preferred_tcp ? 125 : 115, "tcp")); }
} else {
@ -780,12 +779,16 @@ void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
}
}
void WebRtcTransportImp::setIceCandidate(vector<SdpAttrCandidate> cands) {
_cands = std::move(cands);
void WebRtcTransportImp::setPreferredTcp(bool flag) {
_preferred_tcp = flag;
}
void WebRtcTransportImp::setLocalIp(const std::string &localIp) {
_localIp = localIp;
void WebRtcTransportImp::setLocalIp(std::string local_ip) {
_local_ip = std::move(local_ip);
}
void WebRtcTransportImp::setIceCandidate(vector<SdpAttrCandidate> cands) {
_cands = std::move(cands);
}
///////////////////////////////////////////////////////////////////
@ -1310,9 +1313,7 @@ void echo_plugin(Session &sender, const WebRtcArgs &args, const onCreateWebRtc &
void push_plugin(Session &sender, const WebRtcArgs &args, const onCreateWebRtc &cb) {
MediaInfo info(args["url"]);
bool preferred_tcp = args["preferred_tcp"];
Broadcast::PublishAuthInvoker invoker = [cb, info, preferred_tcp](const string &err, const ProtocolOption &option) mutable {
Broadcast::PublishAuthInvoker invoker = [cb, info](const string &err, const ProtocolOption &option) mutable {
if (!err.empty()) {
cb(WebRtcException(SockException(Err_other, err)));
return;
@ -1351,7 +1352,7 @@ void push_plugin(Session &sender, const WebRtcArgs &args, const onCreateWebRtc &
push_src_ownership = push_src->getOwnership();
push_src->setProtocolOption(option);
}
auto rtc = WebRtcPusher::create(EventPollerPool::Instance().getPoller(), push_src, push_src_ownership, info, option, preferred_tcp);
auto rtc = WebRtcPusher::create(EventPollerPool::Instance().getPoller(), push_src, push_src_ownership, info, option);
push_src->setListener(rtc);
cb(*rtc);
};
@ -1366,10 +1367,8 @@ void push_plugin(Session &sender, const WebRtcArgs &args, const onCreateWebRtc &
void play_plugin(Session &sender, const WebRtcArgs &args, const onCreateWebRtc &cb) {
MediaInfo info(args["url"]);
bool preferred_tcp = args["preferred_tcp"];
auto session_ptr = static_pointer_cast<Session>(sender.shared_from_this());
Broadcast::AuthInvoker invoker = [cb, info, session_ptr, preferred_tcp](const string &err) mutable {
Broadcast::AuthInvoker invoker = [cb, info, session_ptr](const string &err) mutable {
if (!err.empty()) {
cb(WebRtcException(SockException(Err_other, err)));
return;
@ -1385,7 +1384,7 @@ void play_plugin(Session &sender, const WebRtcArgs &args, const onCreateWebRtc &
}
// 还原成rtc，目的是为了hook时识别哪种播放协议
info.schema = "rtc";
auto rtc = WebRtcPlayer::create(EventPollerPool::Instance().getPoller(), src, info, preferred_tcp);
auto rtc = WebRtcPlayer::create(EventPollerPool::Instance().getPoller(), src, info);
cb(*rtc);
});
};
@ -1398,58 +1397,64 @@ void play_plugin(Session &sender, const WebRtcArgs &args, const onCreateWebRtc &
}
}
bool isVaildIP(const std::string& ip) {
int a,b,c,d;
return sscanf(ip.c_str(),"%d.%d.%d.%d", &a, &b, &c, &d) == 4;
}
static void set_local_ip(const WebRtcArgs &args, const WebRtcInterface &rtc) {
std::string host = args["Host"];
if (!host.empty()) {
std::string localIp = host.substr(0, host.find(':'));
if (!isVaildIP(localIp) || localIp == "127.0.0.1") {
localIp = "";
}
const_cast<WebRtcInterface &>(rtc).setLocalIp(std::move(localIp));
}
}
static void set_webrtc_cands(const WebRtcArgs &args, const WebRtcInterface &rtc) {
vector<SdpAttrCandidate> cands;
static void setWebRtcArgs(const WebRtcArgs &args, WebRtcInterface &rtc) {
{
auto cand_str = trim(args["cand_udp"]);
auto ip_port = toolkit::split(cand_str, ":");
if (ip_port.size() == 2) {
static auto is_vaild_ip = [](const std::string &ip) -> bool {
int a, b, c, d;
return sscanf(ip.c_str(), "%d.%d.%d.%d", &a, &b, &c, &d) == 4;
};
std::string host = args["Host"];
if (!host.empty()) {
auto local_ip = host.substr(0, host.find(':'));
if (!is_vaild_ip(local_ip) || local_ip == "127.0.0.1") {
local_ip = "";
}
rtc.setLocalIp(std::move(local_ip));
}
}
bool preferred_tcp = args["preferred_tcp"];
{
rtc.setPreferredTcp(preferred_tcp);
}
{
vector<SdpAttrCandidate> cands;
{
auto cand_str = trim(args["cand_udp"]);
auto ip_port = toolkit::split(cand_str, ":");
if (ip_port.size() == 2) {
// udp优先
auto ice_cand = makeIceCandidate(ip_port[0], atoi(ip_port[1].data()), preferred_tcp ? 100 : 120, "udp");
cands.emplace_back(std::move(*ice_cand));
}
}
{
auto cand_str = trim(args["cand_tcp"]);
auto ip_port = toolkit::split(cand_str, ":");
if (ip_port.size() == 2) {
// tcp模式
auto ice_cand = makeIceCandidate(ip_port[0], atoi(ip_port[1].data()), preferred_tcp ? 120 : 100, "tcp");
cands.emplace_back(std::move(*ice_cand));
}
}
if (!cands.empty()) {
// udp优先
auto ice_cand = makeIceCandidate(ip_port[0], atoi(ip_port[1].data()), 120, "udp");
cands.emplace_back(std::move(*ice_cand));
rtc.setIceCandidate(std::move(cands));
}
}
{
auto cand_str = trim(args["cand_tcp"]);
auto ip_port = toolkit::split(cand_str, ":");
if (ip_port.size() == 2) {
// tcp模式
auto ice_cand = makeIceCandidate(ip_port[0], atoi(ip_port[1].data()), 100, "tcp");
cands.emplace_back(std::move(*ice_cand));
}
}
if (!cands.empty()) {
// udp优先
const_cast<WebRtcInterface &>(rtc).setIceCandidate(std::move(cands));
}
}
void setWebRtcArgs(const WebRtcArgs &args, const WebRtcInterface &rtc) {
set_local_ip(args, rtc);
set_webrtc_cands(args, rtc);
}
static onceToken s_rtc_auto_register([]() {
#if !defined (NDEBUG)
// debug模式才开启echo插件
WebRtcPluginManager::Instance().registerPlugin("echo", echo_plugin);
#endif
WebRtcPluginManager::Instance().registerPlugin("push", push_plugin);
WebRtcPluginManager::Instance().registerPlugin("play", play_plugin);
WebRtcPluginManager::Instance().setListener([](Session &sender, const std::string &type, const WebRtcArgs &args, const WebRtcInterface &rtc) {
setWebRtcArgs(args, const_cast<WebRtcInterface&>(rtc));
});
});
}// namespace mediakit

20
webrtc/WebRtcTransport.h
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@ -44,11 +44,10 @@ public:
virtual const std::string& getIdentifier() const = 0;
virtual const std::string& deleteRandStr() const { static std::string s_null; return s_null; }
virtual void setIceCandidate(std::vector<SdpAttrCandidate> cands) {}
virtual void setLocalIp(const std::string &localIp) {}
virtual void setLocalIp(std::string localIp) {}
virtual void setPreferredTcp(bool flag) {}
};
void setWebRtcArgs(const WebRtcArgs &args, const WebRtcInterface &rtc);
class WebRtcException : public WebRtcInterface {
public:
WebRtcException(const SockException &ex) : _ex(ex) {};
@ -252,14 +251,16 @@ public:
void onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool rtx = false);
void createRtpChannel(const std::string &rid, uint32_t ssrc, MediaTrack &track);
void setIceCandidate(std::vector<SdpAttrCandidate> cands) override;
void removeTuple(RTC::TransportTuple* tuple);
void safeShutdown(const SockException &ex);
void setLocalIp(const std::string &localIp) override;
void setPreferredTcp(bool flag) override;
void setLocalIp(std::string local_ip) override;
void setIceCandidate(std::vector<SdpAttrCandidate> cands) override;
protected:
void OnIceServerSelectedTuple(const RTC::IceServer *iceServer, RTC::TransportTuple *tuple) override;
WebRtcTransportImp(const EventPoller::Ptr &poller,bool preferred_tcp = false);
WebRtcTransportImp(const EventPoller::Ptr &poller);
void OnDtlsTransportApplicationDataReceived(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) override;
void onStartWebRTC() override;
void onSendSockData(Buffer::Ptr buf, bool flush = true, RTC::TransportTuple *tuple = nullptr) override;
@ -289,7 +290,7 @@ private:
void onCheckAnswer(RtcSession &sdp);
private:
bool _preferred_tcp;
bool _preferred_tcp = false;
uint16_t _rtx_seq[2] = {0, 0};
//用掉的总流量
uint64_t _bytes_usage = 0;
@ -310,8 +311,8 @@ private:
//根据接收rtp的pt获取相关信息
std::unordered_map<uint8_t/*pt*/, std::unique_ptr<WrappedMediaTrack>> _pt_to_track;
std::vector<SdpAttrCandidate> _cands;
//源访问的hostip
std::string _localIp;
//http访问时的host ip
std::string _local_ip;
};
class WebRtcTransportManager {
@ -333,7 +334,6 @@ private:
class WebRtcArgs : public std::enable_shared_from_this<WebRtcArgs> {
public:
virtual ~WebRtcArgs() = default;
virtual variant operator[](const std::string &key) const = 0;
};