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Merge branch 'master' of github.com:ZLMediaKit/ZLMediaKit

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This commit is contained in:
ziyue 2022-05-15 18:02:14 +08:00
commit e1d4417062
21 changed files with 359 additions and 89 deletions
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10
AUTHORS
View File

@ -46,3 +46,13 @@ Xinghua Zhao <(holychaossword@hotmail.com>
cgm <2958580318@qq.com>
hejilin <1724010622@qq.com>
alexliyu7352 <liyu7352@gmail.com>
cgm <2958580318@qq.com>
[haorui wang](https://github.com/HaoruiWang)
joshuafc <joshuafc@foxmail.com>
[JayChen0519](https://github.com/JayChen0519)
zx <zuoxue@qq.com>
wangcker <wangcker@163.com>
WuPeng <wp@zafu.edu.cn>
[starry](https://github.com/starry)
[mtdxc](https://github.com/mtdxc)
[胡刚风](https://github.com/hugangfeng333)

6
CMakeLists.txt
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@ -11,6 +11,12 @@ if(CCACHE_FOUND)
set_property(GLOBAL PROPERTY RULE_LAUNCH_LINK ccache)
endif(CCACHE_FOUND)
#add_compile_options(-D__STDC_FORMAT_MACROS)
if(CMAKE_COMPILER_IS_GNUCXX)
add_compile_options(-D__STDC_FORMAT_MACROS)
message(STATUS "-D__STDC_FORMAT_MACROS")
endif(CMAKE_COMPILER_IS_GNUCXX)
#set(CMAKE_BUILD_TYPE "Release")
if ("${CMAKE_BUILD_TYPE}" STREQUAL "")
set(CMAKE_BUILD_TYPE "Debug")

11
README.md
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@ -176,6 +176,7 @@ bash build_docker_images.sh
- 播放器
- [基于wasm支持H265的播放器](https://github.com/numberwolf/h265web.js)
- [基于MSE的websocket-fmp4播放器](https://github.com/v354412101/wsPlayer)
- [全国产webrtc sdk(metaRTC)](https://github.com/metartc/metaRTC)
## 授权协议
@ -263,6 +264,16 @@ bash build_docker_images.sh
[cgm](mailto:2958580318@qq.com)
[hejilin](mailto:1724010622@qq.com)
[alexliyu7352](mailto:liyu7352@gmail.com)
[cgm](mailto:2958580318@qq.com)
[haorui wang](https://github.com/HaoruiWang)
[joshuafc](mailto:joshuafc@foxmail.com)
[JayChen0519](https://github.com/JayChen0519)
[zx](mailto:zuoxue@qq.com)
[wangcker](mailto:wangcker@163.com)
[WuPeng](mailto:wp@zafu.edu.cn)
[starry](https://github.com/starry)
[mtdxc](https://github.com/mtdxc)
[胡刚风](https://github.com/hugangfeng333)
## 使用案例

2
conf/config.ini
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@ -266,7 +266,7 @@ port_range=30000-35000
[rtc]
#rtc播放推流、播放超时时间
timeoutSec=15
#本机对rtc客户端的可见ip，作为服务器时一般为公网ip，置空时，会自动获取网卡ip
#本机对rtc客户端的可见ip，作为服务器时一般为公网ip，可有多个，用','分开，当置空时，会自动获取网卡ip
externIP=
#rtc udp服务器监听端口号，所有rtc客户端将通过该端口传输stun/dtls/srtp/srtcp数据，
#该端口是多线程的，同时支持客户端网络切换导致的连接迁移

78
postman/ZLMediaKit.postman_collection.json
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@ -1050,7 +1050,7 @@
"method": "GET",
"header": [],
"url": {
"raw": "{{ZLMediaKit_URL}}/index/api/startRecord?secret={{ZLMediaKit_secret}}&vhost={{defaultVhost}}&app=live&stream=obs&stamp=1000",
"raw": "{{ZLMediaKit_URL}}/index/api/seekRecordStamp?secret={{ZLMediaKit_secret}}&vhost={{defaultVhost}}&app=live&stream=obs&stamp",
"host": [
"{{ZLMediaKit_URL}}"
],
@ -1517,6 +1517,82 @@
},
"response": []
},
{
"name": "开始tcp passive被动发送rtp(startSendRtpPassive)",
"request": {
"method": "GET",
"header": [],
"url": {
"raw": "{{ZLMediaKit_URL}}/index/api/startSendRtpPassive?secret={{ZLMediaKit_secret}}&vhost={{defaultVhost}}&app=live&stream=test&ssrc=1",
"host": [
"{{ZLMediaKit_URL}}"
],
"path": [
"index",
"api",
"startSendRtpPassive"
],
"query": [
{
"key": "secret",
"value": "{{ZLMediaKit_secret}}",
"description": "api操作密钥(配置文件配置)，如果操作ip是127.0.0.1，则不需要此参数"
},
{
"key": "vhost",
"value": "{{defaultVhost}}",
"description": "虚拟主机，例如__defaultVhost__"
},
{
"key": "app",
"value": "live",
"description": "应用名，例如 live"
},
{
"key": "stream",
"value": "test",
"description": "流id，例如 obs"
},
{
"key": "ssrc",
"value": "1",
"description": "rtp推流的ssrc，ssrc不同时，可以推流到多个上级服务器"
},
{
"key": "src_port",
"value": "0",
"description": "指定tcp/udp客户端使用的本地端口，0时为随机端口，该参数非必选参数，不传时为随机端口。",
"disabled": true
},
{
"key": "from_mp4",
"value": "0",
"description": "是否推送本地MP4录像，该参数非必选参数",
"disabled": true
},
{
"key": "use_ps",
"value": "1",
"description": "rtp打包采用ps还是es模式，默认采用ps模式，该参数非必选参数",
"disabled": true
},
{
"key": "pt",
"value": "96",
"description": "rtp payload type，默认96，该参数非必选参数",
"disabled": true
},
{
"key": "only_audio",
"value": "1",
"description": "rtp es方式打包时，是否只打包音频，该参数非必选参数",
"disabled": true
}
]
}
},
"response": []
},
{
"name": "停止 发送rtp(stopSendRtp)",
"request": {

31
server/WebApi.cpp
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@ -1107,6 +1107,7 @@ void installWebApi() {
}
MediaSourceEvent::SendRtpArgs args;
args.passive = false;
args.dst_url = allArgs["dst_url"];
args.dst_port = allArgs["dst_port"];
args.ssrc = allArgs["ssrc"];
@ -1115,7 +1116,7 @@ void installWebApi() {
args.pt = allArgs["pt"].empty() ? 96 : allArgs["pt"].as<int>();
args.use_ps = allArgs["use_ps"].empty() ? true : allArgs["use_ps"].as<bool>();
args.only_audio = allArgs["only_audio"].empty() ? false : allArgs["only_audio"].as<bool>();
TraceL << "pt " << int(args.pt) << " ps " << args.use_ps << " audio " << args.only_audio;
TraceL << "startSendRtp, pt " << int(args.pt) << " ps " << args.use_ps << " audio " << args.only_audio;
src->startSendRtp(args, [val, headerOut, invoker](uint16_t local_port, const SockException &ex) mutable {
if (ex) {
@ -1127,6 +1128,34 @@ void installWebApi() {
});
});
api_regist("/index/api/startSendRtpPassive",[](API_ARGS_MAP_ASYNC){
CHECK_SECRET();
CHECK_ARGS("vhost", "app", "stream", "ssrc");
auto src = MediaSource::find(allArgs["vhost"], allArgs["app"], allArgs["stream"], allArgs["from_mp4"].as<int>());
if (!src) {
throw ApiRetException("该媒体流不存在", API::OtherFailed);
}
MediaSourceEvent::SendRtpArgs args;
args.passive = true;
args.ssrc = allArgs["ssrc"];
args.is_udp = false;
args.src_port = allArgs["src_port"];
args.pt = allArgs["pt"].empty() ? 96 : allArgs["pt"].as<int>();
args.use_ps = allArgs["use_ps"].empty() ? true : allArgs["use_ps"].as<bool>();
args.only_audio = allArgs["only_audio"].empty() ? false : allArgs["only_audio"].as<bool>();
TraceL << "startSendRtpPassive, pt " << int(args.pt) << " ps " << args.use_ps << " audio " << args.only_audio;
src->startSendRtp(args, [val, headerOut, invoker](uint16_t local_port, const SockException &ex) mutable {
if (ex) {
val["code"] = API::OtherFailed;
val["msg"] = ex.what();
}
val["local_port"] = local_port;
invoker(200, headerOut, val.toStyledString());
});
});
api_regist("/index/api/stopSendRtp",[](API_ARGS_MAP){
CHECK_SECRET();
CHECK_ARGS("vhost", "app", "stream");

2
src/Common/MediaSource.h
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@ -94,6 +94,8 @@ public:
bool use_ps = true;
//发送es流时指定是否只发送纯音频流
bool only_audio = true;
//tcp被动方式
bool passive = false;
// rtp payload type
uint8_t pt = 96;
// 指定rtp ssrc

2
src/Common/Parser.cpp
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@ -8,7 +8,7 @@
* may be found in the AUTHORS file in the root of the source tree.
*/
#include <inttypes.h>
#include <cinttypes>
#include "Parser.h"
#include "macros.h"
#include "Network/sockutil.h"

26
src/Http/HlsPlayer.cpp
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@ -275,6 +275,14 @@ void HlsDemuxer::start(const EventPoller::Ptr &poller, TrackListener *listener)
}, poller);
}
void HlsDemuxer::pushTask(std::function<void()> task) {
int64_t stamp = 0;
if (!_frame_cache.empty()) {
stamp = _frame_cache.back().first;
}
_frame_cache.emplace_back(std::make_pair(stamp, std::move(task)));
}
bool HlsDemuxer::inputFrame(const Frame::Ptr &frame) {
//为了避免track准备时间过长, 因此在没准备好之前, 直接消费掉所有的帧
if (!_delegate.isAllTrackReady()) {
@ -287,12 +295,15 @@ bool HlsDemuxer::inputFrame(const Frame::Ptr &frame) {
setPlayPosition(frame->dts());
}
//根据时间戳缓存frame
_frame_cache.emplace(frame->dts(), Frame::getCacheAbleFrame(frame));
auto cached_frame = Frame::getCacheAbleFrame(frame);
_frame_cache.emplace_back(std::make_pair(frame->dts(), [cached_frame, this]() {
_delegate.inputFrame(cached_frame);
}));
if (getBufferMS() > 30 * 1000) {
//缓存超过30秒，强制消费至15秒(减少延时或内存占用)
while (getBufferMS() > 15 * 1000) {
_delegate.inputFrame(_frame_cache.begin()->second);
_frame_cache.begin()->second();
_frame_cache.erase(_frame_cache.begin());
}
//接着播放缓存中最早的帧
@ -332,7 +343,7 @@ void HlsDemuxer::onTick() {
}
//消费掉已经到期的帧
_delegate.inputFrame(it->second);
it->second();
it = _frame_cache.erase(it);
}
}
@ -368,8 +379,15 @@ void HlsPlayerImp::onPlayResult(const SockException &ex) {
void HlsPlayerImp::onShutdown(const SockException &ex) {
if (_demuxer) {
std::weak_ptr<HlsPlayerImp> weak_self = static_pointer_cast<HlsPlayerImp>(shared_from_this());
static_pointer_cast<HlsDemuxer>(_demuxer)->pushTask([weak_self, ex]() {
auto strong_self = weak_self.lock();
if (strong_self) {
strong_self->PlayerImp<HlsPlayer, PlayerBase>::onShutdown(ex);
}
});
} else {
PlayerImp<HlsPlayer, PlayerBase>::onShutdown(ex);
_demuxer = nullptr;
}
}

3
src/Http/HlsPlayer.h
View File

@ -34,6 +34,7 @@ public:
void addTrackCompleted() override { _delegate.addTrackCompleted(); }
void resetTracks() override { ((MediaSink &)_delegate).resetTracks(); }
std::vector<Track::Ptr> getTracks(bool ready = true) const override { return _delegate.getTracks(ready); }
void pushTask(std::function<void()> task);
private:
void onTick();
@ -46,7 +47,7 @@ private:
toolkit::Ticker _ticker;
toolkit::Timer::Ptr _timer;
MediaSinkDelegate _delegate;
std::multimap<int64_t, Frame::Ptr> _frame_cache;
std::deque<std::pair<int64_t, std::function<void()> > > _frame_cache;
};
class HlsPlayer : public HttpClientImp , public PlayerBase , public HlsParser{

2
src/Http/HttpClientImp.cpp
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@ -16,6 +16,8 @@ namespace mediakit {
void HttpClientImp::onConnect(const SockException &ex) {
if (!isHttps()) {
//https 302跳转 http时，需要关闭ssl
setDoNotUseSSL();
HttpClient::onConnect(ex);
} else {
TcpClientWithSSL<HttpClient>::onConnect(ex);

13
src/Http/TsplayerImp.cpp
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@ -45,8 +45,17 @@ void TsPlayerImp::onPlayResult(const SockException &ex) {
}
void TsPlayerImp::onShutdown(const SockException &ex) {
PlayerImp<TsPlayer, PlayerBase>::onShutdown(ex);
_demuxer = nullptr;
if (_demuxer) {
std::weak_ptr<TsPlayerImp> weak_self = static_pointer_cast<TsPlayerImp>(shared_from_this());
static_pointer_cast<HlsDemuxer>(_demuxer)->pushTask([weak_self, ex]() {
auto strong_self = weak_self.lock();
if (strong_self) {
strong_self->PlayerImp<TsPlayer, PlayerBase>::onShutdown(ex);
}
});
} else {
PlayerImp<TsPlayer, PlayerBase>::onShutdown(ex);
}
}
vector<Track::Ptr> TsPlayerImp::getTracks(bool ready) const {

123
src/Rtp/RtpSender.cpp
View File

@ -12,6 +12,7 @@
#include "RtpSender.h"
#include "Rtsp/RtspSession.h"
#include "Thread/WorkThreadPool.h"
#include "Util/uv_errno.h"
#include "RtpCache.h"
using namespace std;
@ -19,49 +20,105 @@ using namespace toolkit;
namespace mediakit{
RtpSender::RtpSender() {
_poller = EventPollerPool::Instance().getPoller();
_socket = Socket::createSocket(_poller, false);
}
void RtpSender::startSend(const MediaSourceEvent::SendRtpArgs &args, const function<void(uint16_t local_port, const SockException &ex)> &cb){
_args = args;
_poller = EventPollerPool::Instance().getPoller();
auto lam = [this](std::shared_ptr<List<Buffer::Ptr>> list) { onFlushRtpList(std::move(list)); };
if (args.use_ps) {
_interface = std::make_shared<RtpCachePS>(lam, atoi(args.ssrc.data()), args.pt);
} else {
_interface = std::make_shared<RtpCacheRaw>(lam, atoi(args.ssrc.data()), args.pt, args.only_audio);
}
_socket = Socket::createSocket(_poller, false);
weak_ptr<RtpSender> weak_self = shared_from_this();
if (args.is_udp) {
if (args.src_port) {
//指定端口
_socket->bindUdpSock(args.src_port);
if (!_interface) {
//重连时不重新创建对象
auto lam = [this](std::shared_ptr<List<Buffer::Ptr>> list) { onFlushRtpList(std::move(list)); };
if (args.use_ps) {
_interface = std::make_shared<RtpCachePS>(lam, atoi(args.ssrc.data()), args.pt);
} else {
auto pr = std::make_pair(std::move(_socket), Socket::createSocket(_poller, false));
//从端口池获取随机端口
makeSockPair(pr, "::", true);
_socket = std::move(pr.first);
_interface = std::make_shared<RtpCacheRaw>(lam, atoi(args.ssrc.data()), args.pt, args.only_audio);
}
}
weak_ptr<RtpSender> weak_self = shared_from_this();
if (args.passive) {
// tcp被动发流模式
_args.is_udp = false;
try {
auto tcp_listener = Socket::createSocket(_poller, false);
if (args.src_port) {
//指定端口
if (!tcp_listener->listen(args.src_port)) {
throw std::invalid_argument(StrPrinter << "open tcp passive server failed on port:" << args.src_port
<< ", err:" << get_uv_errmsg(true));
}
} else {
auto pr = std::make_pair(tcp_listener, Socket::createSocket(_poller, false));
//从端口池获取随机端口
makeSockPair(pr, "::", false, false);
}
// tcp服务器默认开启5秒
auto delay_task = _poller->doDelayTask(5 * 1000, [tcp_listener, cb]() mutable {
cb(0, SockException(Err_timeout, "wait tcp connection timeout"));
tcp_listener = nullptr;
return 0;
});
tcp_listener->setOnAccept([weak_self, cb, delay_task](Socket::Ptr &sock, std::shared_ptr<void> &complete) {
auto strong_self = weak_self.lock();
if (!strong_self) {
return;
}
//立即关闭tcp服务器
delay_task->cancel();
strong_self->_socket = sock;
strong_self->onConnect();
cb(sock->get_local_port(), SockException());
InfoL << "accept connection from:" << sock->get_peer_ip() << ":" << sock->get_peer_port();
});
InfoL << "start tcp passive server on:" << tcp_listener->get_local_port();
} catch (std::exception &ex) {
cb(0, SockException(Err_other, ex.what()));
return;
}
return;
}
if (args.is_udp) {
auto poller = _poller;
auto local_port = _socket->get_local_port();
WorkThreadPool::Instance().getPoller()->async([cb, args, weak_self, poller, local_port]() {
WorkThreadPool::Instance().getPoller()->async([cb, args, weak_self, poller]() {
struct sockaddr_storage addr;
//切换线程目的是为了dns解析放在后台线程执行
if (!SockUtil::getDomainIP(args.dst_url.data(), args.dst_port, addr, AF_INET, SOCK_DGRAM, IPPROTO_UDP)) {
poller->async([args, cb, local_port]() {
poller->async([args, cb]() {
//切回自己的线程
cb(local_port, SockException(Err_dns, StrPrinter << "dns解析域名失败:" << args.dst_url));
cb(0, SockException(Err_dns, StrPrinter << "dns解析域名失败:" << args.dst_url));
});
return;
}
//dns解析成功
poller->async([addr, weak_self, cb, local_port]() {
poller->async([args, addr, weak_self, cb]() {
//切回自己的线程
cb(local_port, SockException());
auto strong_self = weak_self.lock();
if (strong_self) {
strong_self->_socket->bindPeerAddr((struct sockaddr *)&addr);
strong_self->onConnect();
if (!strong_self) {
return;
}
string ifr_ip = addr.ss_family == AF_INET ? "0.0.0.0" : "::";
try {
if (args.src_port) {
//指定端口
if (!strong_self->_socket->bindUdpSock(args.src_port, ifr_ip)) {
throw std::invalid_argument(StrPrinter << "bindUdpSock failed on port:" << args.src_port
<< ", err:" << get_uv_errmsg(true));
}
} else {
auto pr = std::make_pair(strong_self->_socket, Socket::createSocket(strong_self->_poller, false));
//从端口池获取随机端口
makeSockPair(pr, ifr_ip, true);
}
} catch (std::exception &ex) {
cb(0, SockException(Err_other, ex.what()));
return;
}
strong_self->_socket->bindPeerAddr((struct sockaddr *)&addr);
strong_self->onConnect();
cb(strong_self->_socket->get_local_port(), SockException());
});
});
} else {
@ -76,7 +133,7 @@ void RtpSender::startSend(const MediaSourceEvent::SendRtpArgs &args, const funct
} else {
cb(0, err);
}
}, 5.0F, "::", args.src_port);
}, 5.0F, "::", args.src_port);
}
}
@ -147,11 +204,15 @@ void RtpSender::onFlushRtpList(shared_ptr<List<Buffer::Ptr> > rtp_list) {
void RtpSender::onErr(const SockException &ex, bool is_connect) {
_is_connect = false;
//监听socket断开事件，方便重连
if (is_connect) {
WarnL << "重连" << _args.dst_url << ":" << _args.dst_port << "失败, 原因为:" << ex.what();
if (_args.passive) {
WarnL << "tcp passive connection lost: " << ex.what();
} else {
WarnL << "停止发送 rtp:" << _args.dst_url << ":" << _args.dst_port << ", 原因为:" << ex.what();
//监听socket断开事件，方便重连
if (is_connect) {
WarnL << "重连" << _args.dst_url << ":" << _args.dst_port << "失败, 原因为:" << ex.what();
} else {
WarnL << "停止发送 rtp:" << _args.dst_url << ":" << _args.dst_port << ", 原因为:" << ex.what();
}
}
weak_ptr<RtpSender> weak_self = shared_from_this();

2
src/Rtp/RtpSender.h
View File

@ -21,7 +21,7 @@ class RtpSender : public MediaSinkInterface, public std::enable_shared_from_this
public:
typedef std::shared_ptr<RtpSender> Ptr;
RtpSender() = default;
RtpSender();
~RtpSender() override = default;
/**

3
src/Rtp/RtpServer.cpp
View File

@ -99,9 +99,6 @@ void RtpServer::start(uint16_t local_port, const string &stream_id, bool enable_
//随机端口，rtp端口采用偶数
auto pair = std::make_pair(rtp_socket, rtcp_socket);
makeSockPair(pair, local_ip, re_use_port);
//取偶数端口
rtp_socket = pair.first;
rtcp_socket = pair.second;
} else if (!rtp_socket->bindUdpSock(local_port, local_ip, re_use_port)) {
//用户指定端口
throw std::runtime_error(StrPrinter << "创建rtp端口 " << local_ip << ":" << local_port << " 失败:" << get_uv_errmsg(true));

65
src/Rtsp/Rtsp.cpp
View File

@ -312,7 +312,8 @@ string SdpParser::toString() const {
return title + video + audio;
}
class PortManager : public std::enable_shared_from_this<PortManager> {
template<int type>
class PortManager : public std::enable_shared_from_this<PortManager<type> > {
public:
PortManager() {
static auto func = [](const string &str, int index) {
@ -331,28 +332,44 @@ public:
return *instance;
}
void bindUdpSock(std::pair<Socket::Ptr, Socket::Ptr> &pair, const string &local_ip, bool re_use_port) {
void makeSockPair(std::pair<Socket::Ptr, Socket::Ptr> &pair, const string &local_ip, bool re_use_port, bool is_udp) {
auto &sock0 = pair.first;
auto &sock1 = pair.second;
auto sock_pair = getPortPair();
if (!sock_pair) {
throw runtime_error("none reserved udp port in pool");
throw runtime_error("none reserved port in pool");
}
if (is_udp) {
if (!sock0->bindUdpSock(2 * *sock_pair, local_ip.data(), re_use_port)) {
//分配端口失败
throw runtime_error("open udp socket[0] failed");
}
if (!sock0->bindUdpSock(2 * *sock_pair, local_ip.data(), re_use_port)) {
//分配端口失败
throw runtime_error("open udp socket[0] failed");
if (!sock1->bindUdpSock(2 * *sock_pair + 1, local_ip.data(), re_use_port)) {
//分配端口失败
throw runtime_error("open udp socket[1] failed");
}
auto on_cycle = [sock_pair](Socket::Ptr &, std::shared_ptr<void> &) {};
// udp socket没onAccept事件，设置该回调，目的是为了在销毁socket时，回收对象
sock0->setOnAccept(on_cycle);
sock1->setOnAccept(on_cycle);
} else {
if (!sock0->listen(2 * *sock_pair, local_ip.data())) {
//分配端口失败
throw runtime_error("listen tcp socket[0] failed");
}
if (!sock1->listen(2 * *sock_pair + 1, local_ip.data())) {
//分配端口失败
throw runtime_error("listen tcp socket[1] failed");
}
auto on_cycle = [sock_pair](const Buffer::Ptr &buf, struct sockaddr *addr, int addr_len) {};
// udp socket没onAccept事件，设置该回调，目的是为了在销毁socket时，回收对象
sock0->setOnRead(on_cycle);
sock1->setOnRead(on_cycle);
}
if (!sock1->bindUdpSock(2 * *sock_pair + 1, local_ip.data(), re_use_port)) {
//分配端口失败
throw runtime_error("open udp socket[1] failed");
}
auto on_cycle = [sock_pair](Socket::Ptr &, std::shared_ptr<void> &) {};
// udp socket没onAccept事件，设置该回调，目的是为了在销毁socket时，回收对象
sock0->setOnAccept(on_cycle);
sock1->setOnAccept(on_cycle);
}
private:
@ -372,7 +389,7 @@ private:
_port_pair_pool.pop_front();
InfoL << "got port from pool:" << 2 * pos << "-" << 2 * pos + 1;
weak_ptr<PortManager> weak_self = shared_from_this();
weak_ptr<PortManager> weak_self = this->shared_from_this();
std::shared_ptr<uint16_t> ret(new uint16_t(pos), [weak_self, pos](uint16_t *ptr) {
delete ptr;
auto strong_self = weak_self.lock();
@ -392,20 +409,22 @@ private:
deque<uint16_t> _port_pair_pool;
};
void makeSockPair(std::pair<Socket::Ptr, Socket::Ptr> &pair, const string &local_ip, bool re_use_port) {
//全局互斥锁保护，防止端口重复分配
static recursive_mutex s_mtx;
lock_guard<recursive_mutex> lck(s_mtx);
void makeSockPair(std::pair<Socket::Ptr, Socket::Ptr> &pair, const string &local_ip, bool re_use_port, bool is_udp) {
int try_count = 0;
while (true) {
try {
PortManager::Instance().bindUdpSock(pair, local_ip, re_use_port);
//udp和tcp端口池使用相同算法和范围分配，但是互不相干
if (is_udp) {
PortManager<0>::Instance().makeSockPair(pair, local_ip, re_use_port, is_udp);
} else {
PortManager<1>::Instance().makeSockPair(pair, local_ip, re_use_port, is_udp);
}
break;
} catch (exception &ex) {
if (++try_count == 3) {
throw;
}
WarnL << "open udp socket failed:" << ex.what() << ", retry: " << try_count;
WarnL << "open socket failed:" << ex.what() << ", retry: " << try_count;
}
}
}

2
src/Rtsp/Rtsp.h
View File

@ -357,7 +357,7 @@ private:
//创建rtp over tcp4个字节的头
toolkit::Buffer::Ptr makeRtpOverTcpPrefix(uint16_t size, uint8_t interleaved);
//创建rtp-rtcp端口对
void makeSockPair(std::pair<toolkit::Socket::Ptr, toolkit::Socket::Ptr> &pair, const std::string &local_ip, bool re_use_port = false);
void makeSockPair(std::pair<toolkit::Socket::Ptr, toolkit::Socket::Ptr> &pair, const std::string &local_ip, bool re_use_port = false, bool is_udp = true);
//十六进制方式打印ssrc
std::string printSSRC(uint32_t ui32Ssrc);

2
webrtc/RtpExt.cpp
View File

@ -189,7 +189,7 @@ map<uint8_t/*id*/, RtpExt/*data*/> RtpExt::getExtValue(const RtpHeader *header)
appendExt<RtpExtOneByte>(ret, ptr, end);
return ret;
}
if ((reserved & 0xFFF0) >> 4 == kTwoByteHeader) {
if ((reserved & 0xFFF0) == kTwoByteHeader) {
appendExt<RtpExtTwoByte>(ret, ptr, end);
return ret;
}

5
webrtc/Sdp.cpp
View File

@ -1591,8 +1591,13 @@ RETRY:
if (type == TrackApplication) {
RtcMedia answer_media = offer_media;
answer_media.role = mathDtlsRole(offer_media.role);
#ifdef ENABLE_SCTP
answer_media.direction = matchDirection(offer_media.direction, configure.direction);
answer_media.candidate = configure.candidate;
#else
answer_media.direction = RtpDirection::inactive;
#endif
ret->media.emplace_back(answer_media);
return;
}

57
webrtc/WebRtcTransport.cpp
View File

@ -471,10 +471,15 @@ void WebRtcTransportImp::onStartWebRTC() {
void WebRtcTransportImp::onCheckAnswer(RtcSession &sdp) {
//修改answer sdp的ip、端口信息
GET_CONFIG(string, extern_ip, RTC::kExternIP);
GET_CONFIG_FUNC(std::vector<std::string>, extern_ips, RTC::kExternIP, [](string str){
std::vector<std::string> ret;
if (str.length())
ret = split(str, ",");
return ret;
});
for (auto &m : sdp.media) {
m.addr.reset();
m.addr.address = extern_ip.empty() ? SockUtil::get_local_ip() : extern_ip;
m.addr.address = extern_ips.empty() ? SockUtil::get_local_ip() : extern_ips[0];
m.rtcp_addr.reset();
m.rtcp_addr.address = m.addr.address;
@ -522,28 +527,48 @@ void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp) {
}
}
void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
WebRtcTransport::onRtcConfigure(configure);
//添加接收端口candidate信息
configure.addCandidate(*getIceCandidate());
}
SdpAttrCandidate::Ptr WebRtcTransportImp::getIceCandidate() const{
SdpAttrCandidate::Ptr makeIceCandidate(std::string ip, uint16_t port,
uint32_t priority = 100, std::string proto = "udp") {
auto candidate = std::make_shared<SdpAttrCandidate>();
candidate->foundation = "udpcandidate";
//rtp端口
candidate->component = 1;
candidate->transport = "udp";
candidate->transport = proto;
candidate->foundation = proto + "candidate";
//优先级，单candidate时随便
candidate->priority = 100;
GET_CONFIG(string, extern_ip, RTC::kExternIP);
candidate->address = extern_ip.empty() ? SockUtil::get_local_ip() : extern_ip;
GET_CONFIG(uint16_t, local_port, RTC::kPort);
candidate->port = local_port;
candidate->priority = priority;
candidate->address = ip;
candidate->port = port;
candidate->type = "host";
return candidate;
}
void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
WebRtcTransport::onRtcConfigure(configure);
GET_CONFIG(uint16_t, local_port, RTC::kPort);
//添加接收端口candidate信息
GET_CONFIG_FUNC(std::vector<std::string>, extern_ips, RTC::kExternIP, [](string str){
std::vector<std::string> ret;
if (str.length())
ret = split(str, ",");
return ret;
});
if (extern_ips.empty()) {
std::string localIp = SockUtil::get_local_ip();
configure.addCandidate(*makeIceCandidate(localIp, local_port, 120, "udp"));
}
else {
const uint32_t delta = 10;
uint32_t priority = 100 + delta * extern_ips.size();
for (auto ip : extern_ips) {
configure.addCandidate(*makeIceCandidate(ip, local_port, priority, "udp"));
priority -= delta;
}
}
}
///////////////////////////////////////////////////////////////////
class RtpChannel : public RtpTrackImp, public std::enable_shared_from_this<RtpChannel> {

1
webrtc/WebRtcTransport.h
View File

@ -264,7 +264,6 @@ protected:
void updateTicker();
private:
SdpAttrCandidate::Ptr getIceCandidate() const;
void onSortedRtp(MediaTrack &track, const std::string &rid, mediakit::RtpPacket::Ptr rtp);
void onSendNack(MediaTrack &track, const mediakit::FCI_NACK &nack, uint32_t ssrc);
void onSendTwcc(uint32_t ssrc, const std::string &twcc_fci);