Merge pull request #1 from ZLMediaKit/master

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ljx0305 2024-03-25 18:59:03 +08:00 committed by GitHub
commit 5428ba1d31
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40 changed files with 693 additions and 859 deletions

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@ -44,7 +44,6 @@ Xinghua Zhao <(holychaossword@hotmail.com>
[Dw9](https://github.com/Dw9)
明月惊鹊 <mingyuejingque@gmail.com>
cgm <2958580318@qq.com>
hejilin <1724010622@qq.com>
alexliyu7352 <liyu7352@gmail.com>
cgm <2958580318@qq.com>
[haorui wang](https://github.com/HaoruiWang)
@ -104,3 +103,8 @@ WuPeng <wp@zafu.edu.cn>
[sandro-qiang](https://github.com/sandro-qiang)
[Paul Philippov](https://github.com/themactep)
[张传峰](https://github.com/zhang-chuanfeng)
[lidaofu-hub](https://github.com/lidaofu-hub)
[huangcaichun](https://github.com/huangcaichun)
[jamesZHANG500](https://github.com/jamesZHANG500)
[weidelong](https://github.com/wdl1697454803)
[小强先生](https://github.com/linshangqiang)

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@ -141,8 +141,8 @@ if(GIT_FOUND)
endif()
configure_file(
${CMAKE_CURRENT_SOURCE_DIR}/version.h.ini
${CMAKE_CURRENT_BINARY_DIR}/version.h
${CMAKE_CURRENT_SOURCE_DIR}/ZLMVersion.h.ini
${CMAKE_CURRENT_BINARY_DIR}/ZLMVersion.h
@ONLY)
message(STATUS "Git version is ${BRANCH_NAME} ${COMMIT_HASH}/${COMMIT_TIME} ${BUILD_TIME}")

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@ -358,6 +358,11 @@ bash build_docker_images.sh
[sandro-qiang](https://github.com/sandro-qiang)
[Paul Philippov](https://github.com/themactep)
[张传峰](https://github.com/zhang-chuanfeng)
[lidaofu-hub](https://github.com/lidaofu-hub)
[huangcaichun](https://github.com/huangcaichun)
[jamesZHANG500](https://github.com/jamesZHANG500)
[weidelong](https://github.com/wdl1697454803)
[小强先生](https://github.com/linshangqiang)
同时感谢JetBrains对开源项目的支持本项目使用CLion开发与调试

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@ -516,6 +516,11 @@ Thanks to all those who have supported this project in various ways, including b
[sandro-qiang](https://github.com/sandro-qiang)
[Paul Philippov](https://github.com/themactep)
[张传峰](https://github.com/zhang-chuanfeng)
[lidaofu-hub](https://github.com/lidaofu-hub)
[huangcaichun](https://github.com/huangcaichun)
[jamesZHANG500](https://github.com/jamesZHANG500)
[weidelong](https://github.com/wdl1697454803)
[小强先生](https://github.com/linshangqiang)
Also thank to JetBrains for their support for open source project, we developed and debugged zlmediakit with CLion:

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@ -304,10 +304,10 @@ API_EXPORT void API_CALL mk_webrtc_get_answer_sdp2(void *user_data, on_user_data
std::string offer_str = offer;
std::shared_ptr<void> ptr(user_data, user_data_free ? user_data_free : [](void *) {});
auto args = std::make_shared<WebRtcArgsUrl>(url);
WebRtcPluginManager::Instance().getAnswerSdp(*session, type, *args,
[offer_str, session, ptr, cb](const WebRtcInterface &exchanger) mutable {
WebRtcPluginManager::Instance().negotiateSdp(*session, type, *args, [offer_str, session, ptr, cb](const WebRtcInterface &exchanger) mutable {
auto &handler = const_cast<WebRtcInterface &>(exchanger);
try {
auto sdp_answer = exchangeSdp(exchanger, offer_str);
auto sdp_answer = handler.getAnswerSdp(offer_str);
cb(ptr.get(), sdp_answer.data(), nullptr);
} catch (std::exception &ex) {
cb(ptr.get(), nullptr, ex.what());

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@ -14,7 +14,10 @@
#include "Http/HttpSession.h"
#include "Rtsp/RtspSession.h"
#include "Record/MP4Recorder.h"
#ifdef ENABLE_WEBRTC
#include "webrtc/WebRtcTransport.h"
#endif
using namespace toolkit;
using namespace mediakit;
@ -168,7 +171,7 @@ API_EXPORT void API_CALL mk_events_listen(const mk_events *events){
sender.getMediaTuple().stream.c_str(), ssrc.c_str(), ex.getErrCode(), ex.what());
}
});
#ifdef ENABLE_WEBRTC
NoticeCenter::Instance().addListener(&s_tag, Broadcast::kBroadcastRtcSctpConnecting,[](BroadcastRtcSctpConnectArgs){
if (s_events.on_mk_rtc_sctp_connecting) {
s_events.on_mk_rtc_sctp_connecting((mk_rtc_transport)&sender);
@ -204,6 +207,7 @@ API_EXPORT void API_CALL mk_events_listen(const mk_events *events){
s_events.on_mk_rtc_sctp_received((mk_rtc_transport)&sender, streamId, ppid, msg, len);
}
});
#endif
});
}

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@ -17,7 +17,10 @@
#include "Http/HttpClient.h"
#include "Rtsp/RtspSession.h"
#ifdef ENABLE_WEBRTC
#include "webrtc/WebRtcTransport.h"
#endif
using namespace toolkit;
using namespace mediakit;

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@ -357,7 +357,7 @@ tcpPort = 8000
rembBitRate=0
#rtc支持的音频codec类型,在前面的优先级更高
#以下范例为所有支持的音频codec
preferredCodecA=PCMU,PCMA,opus,mpeg4-generic
preferredCodecA=PCMA,PCMU,opus,mpeg4-generic
#rtc支持的视频codec类型,在前面的优先级更高
#以下范例为所有支持的视频codec
preferredCodecV=H264,H265,AV1,VP9,VP8

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@ -17,7 +17,7 @@ using namespace toolkit;
namespace mediakit {
void AACRtmpDecoder::inputRtmp(const RtmpPacket::Ptr &pkt) {
CHECK(pkt->size() > 2);
CHECK_RET(pkt->size() > 2);
if (pkt->isConfigFrame()) {
getTrack()->setExtraData((uint8_t *)pkt->data() + 2, pkt->size() - 2);
return;

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@ -14,14 +14,6 @@
using namespace std;
using namespace toolkit;
#define CHECK_RET(...) \
try { \
CHECK(__VA_ARGS__); \
} catch (AssertFailedException & ex) { \
WarnL << ex.what(); \
return; \
}
namespace mediakit {
void H264RtmpDecoder::inputRtmp(const RtmpPacket::Ptr &pkt) {

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@ -18,14 +18,6 @@
using namespace std;
using namespace toolkit;
#define CHECK_RET(...) \
try { \
CHECK(__VA_ARGS__); \
} catch (AssertFailedException & ex) { \
WarnL << ex.what(); \
return; \
}
namespace mediakit {
void H265RtmpDecoder::inputRtmp(const RtmpPacket::Ptr &pkt) {

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@ -62,7 +62,8 @@ void Channel::addParam(const std::weak_ptr<Param>& p)
void Channel::onFrame(const mediakit::FFmpegFrame::Ptr& frame)
{
std::weak_ptr<Channel> weakSelf = shared_from_this();
// toolkit::WorkThreadPool::Instance().getFirstPoller()->async([weakSelf, frame]() {
_poller = _poller ? _poller : toolkit::WorkThreadPool::Instance().getPoller();
_poller->async([weakSelf, frame]() {
auto self = weakSelf.lock();
if (!self) {
return;
@ -70,7 +71,7 @@ void Channel::onFrame(const mediakit::FFmpegFrame::Ptr& frame)
self->_tmp = self->_sws->inputFrame(frame);
self->forEachParam([self](const Param::Ptr& p) { self->fillBuffer(p); });
// });
});
}
void Channel::forEachParam(const std::function<void(const Param::Ptr&)>& func)
@ -440,6 +441,7 @@ int VideoStackManager::stopVideoStack(const std::string& id)
auto it = _stackMap.find(id);
if (it != _stackMap.end()) {
_stackMap.erase(it);
InfoL << "VideoStack stop: " << id;
return 0;
}
return -1;

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@ -80,6 +80,7 @@ class Channel : public std::enable_shared_from_this<Channel> {
std::vector<std::weak_ptr<Param>> _params;
mediakit::FFmpegSws::Ptr _sws;
toolkit::EventPoller::Ptr _poller;
};
class StackPlayer : public std::enable_shared_from_this<StackPlayer> {

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@ -59,7 +59,7 @@
#endif
#if defined(ENABLE_VERSION)
#include "version.h"
#include "ZLMVersion.h"
#endif
#if defined(ENABLE_X264) && defined (ENABLE_FFMPEG)
@ -119,7 +119,7 @@ static HttpApi toApi(const function<void(API_ARGS_MAP_ASYNC)> &cb) {
//参数解析成map
auto args = getAllArgs(parser);
cb(sender, headerOut, HttpAllArgs<decltype(args)>(parser, args), val, invoker);
cb(sender, headerOut, ArgsMap(parser, args), val, invoker);
};
}
@ -147,7 +147,7 @@ static HttpApi toApi(const function<void(API_ARGS_JSON_ASYNC)> &cb) {
Json::Reader reader;
reader.parse(parser.content(), args);
cb(sender, headerOut, HttpAllArgs<decltype(args)>(parser, args), val, invoker);
cb(sender, headerOut, ArgsJson(parser, args), val, invoker);
};
}
@ -167,7 +167,7 @@ static HttpApi toApi(const function<void(API_ARGS_STRING_ASYNC)> &cb) {
Json::Value val;
val["code"] = API::Success;
cb(sender, headerOut, HttpAllArgs<string>(parser, (string &)parser.content()), val, invoker);
cb(sender, headerOut, ArgsString(parser, (string &)parser.content()), val, invoker);
};
}
@ -584,8 +584,10 @@ void addStreamProxy(const string &vhost, const string &app, const string &stream
//添加拉流代理
auto player = s_player_proxy.make(key, vhost, app, stream, option, retry_count);
// 先透传参数
player->mINI::operator=(args);
// 先透传拷贝参数
for (auto &pr : args) {
(*player)[pr.first] = pr.second;
}
//指定RTP over TCP(播放rtsp时有效)
(*player)[Client::kRtpType] = rtp_type;
@ -660,13 +662,6 @@ void addStreamPusherProxy(const string &schema,
pusher->publish(url);
}
template <typename Type>
static void getArgsValue(const HttpAllArgs<ApiArgsType> &allArgs, const string &key, Type &value) {
auto val = allArgs[key];
if (!val.empty()) {
value = (Type)val;
}
}
/**
* api接口
@ -733,7 +728,7 @@ void installWebApi() {
CHECK_SECRET();
auto &ini = mINI::Instance();
int changed = API::Success;
for (auto &pr : allArgs.getArgs()) {
for (auto &pr : allArgs.args) {
if (ini.find(pr.first) == ini.end()) {
#if 1
//没有这个key
@ -1091,7 +1086,7 @@ void installWebApi() {
CHECK_ARGS("vhost","app","stream","url");
mINI args;
for (auto &pr : allArgs.getArgs()) {
for (auto &pr : allArgs.args) {
args.emplace(pr.first, pr.second);
}
@ -1188,7 +1183,7 @@ void installWebApi() {
//测试url http://127.0.0.1/index/api/downloadBin
api_regist("/index/api/downloadBin",[](API_ARGS_MAP_ASYNC){
CHECK_SECRET();
invoker.responseFile(allArgs.getParser().getHeader(),StrCaseMap(),exePath());
invoker.responseFile(allArgs.parser.getHeader(), StrCaseMap(), exePath());
});
#if defined(ENABLE_RTPPROXY)
@ -1695,7 +1690,7 @@ void installWebApi() {
//截图存在,且未过期,那么返回之
res_old_snap = true;
responseSnap(path, allArgs.getParser().getHeader(), invoker);
responseSnap(path, allArgs.parser.getHeader(), invoker);
//中断遍历
return false;
});
@ -1726,7 +1721,7 @@ void installWebApi() {
File::delete_file(new_snap);
rename(new_snap_tmp.data(), new_snap.data());
}
responseSnap(new_snap, allArgs.getParser().getHeader(), invoker, err_msg);
responseSnap(new_snap, allArgs.parser.getHeader(), invoker, err_msg);
});
});
@ -1741,7 +1736,7 @@ void installWebApi() {
#ifdef ENABLE_WEBRTC
class WebRtcArgsImp : public WebRtcArgs {
public:
WebRtcArgsImp(const HttpAllArgs<string> &args, std::string session_id)
WebRtcArgsImp(const ArgsString &args, std::string session_id)
: _args(args)
, _session_id(std::move(session_id)) {}
~WebRtcArgsImp() override = default;
@ -1759,40 +1754,26 @@ void installWebApi() {
CHECK_ARGS("app", "stream");
return StrPrinter << "rtc://" << _args["Host"] << "/" << _args["app"] << "/"
<< _args["stream"] << "?" << _args.getParser().params() + "&session=" + _session_id;
<< _args["stream"] << "?" << _args.parser.params() + "&session=" + _session_id;
}
private:
HttpAllArgs<string> _args;
ArgsString _args;
std::string _session_id;
};
api_regist("/index/api/webrtc",[](API_ARGS_STRING_ASYNC){
CHECK_ARGS("type");
auto type = allArgs["type"];
auto offer = allArgs.getArgs();
auto offer = allArgs.args;
CHECK(!offer.empty(), "http body(webrtc offer sdp) is empty");
std::string host = allArgs.getParser()["Host"];
std::string localIp = host.substr(0, host.find(':'));
auto isVaildIP = [](std::string ip)-> bool {
int a,b,c,d;
return sscanf(ip.c_str(),"%d.%d.%d.%d", &a, &b, &c, &d) == 4;
};
if (!isVaildIP(localIp) || localIp=="127.0.0.1") {
localIp = "";
}
auto &session = static_cast<Session&>(sender);
auto args = std::make_shared<WebRtcArgsImp>(allArgs, sender.getIdentifier());
WebRtcPluginManager::Instance().getAnswerSdp(static_cast<Session&>(sender), type, *args, [invoker, val, offer, headerOut, localIp](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"] = "*";
WebRtcPluginManager::Instance().negotiateSdp(session, type, *args, [invoker, val, offer, headerOut](const WebRtcInterface &exchanger) mutable {
auto &handler = const_cast<WebRtcInterface &>(exchanger);
try {
setLocalIp(exchanger,localIp);
val["sdp"] = exchangeSdp(exchanger, offer);
val["sdp"] = handler.getAnswerSdp(offer);
val["id"] = exchanger.getIdentifier();
val["type"] = "answer";
invoker(200, headerOut, val.toStyledString());
@ -1806,21 +1787,19 @@ void installWebApi() {
static constexpr char delete_webrtc_url [] = "/index/api/delete_webrtc";
static auto whip_whep_func = [](const char *type, API_ARGS_STRING_ASYNC) {
auto offer = allArgs.getArgs();
auto offer = allArgs.args;
CHECK(!offer.empty(), "http body(webrtc offer sdp) is empty");
auto &session = static_cast<Session&>(sender);
auto location = std::string("http") + (session.overSsl() ? "s" : "") + "://" + allArgs["host"] + delete_webrtc_url;
auto location = std::string(session.overSsl() ? "https://" : "http://") + allArgs["host"] + delete_webrtc_url;
auto args = std::make_shared<WebRtcArgsImp>(allArgs, sender.getIdentifier());
WebRtcPluginManager::Instance().getAnswerSdp(session, type, *args,
[invoker, offer, headerOut, location](const WebRtcInterface &exchanger) mutable {
// 设置跨域
headerOut["Access-Control-Allow-Origin"] = "*";
WebRtcPluginManager::Instance().negotiateSdp(session, type, *args, [invoker, offer, headerOut, location](const WebRtcInterface &exchanger) mutable {
auto &handler = const_cast<WebRtcInterface &>(exchanger);
try {
// 设置返回类型
headerOut["Content-Type"] = "application/sdp";
headerOut["Location"] = location + "?id=" + exchanger.getIdentifier() + "&token=" + exchanger.deleteRandStr();
invoker(201, headerOut, exchangeSdp(exchanger, offer));
invoker(201, headerOut, handler.getAnswerSdp(offer));
} catch (std::exception &ex) {
headerOut["Content-Type"] = "text/plain";
invoker(406, headerOut, ex.what());
@ -1833,7 +1812,7 @@ void installWebApi() {
api_regist(delete_webrtc_url, [](API_ARGS_MAP_ASYNC) {
CHECK_ARGS("id", "token");
CHECK(allArgs.getParser().method() == "DELETE", "http method is not DELETE: " + allArgs.getParser().method());
CHECK(allArgs.parser.method() == "DELETE", "http method is not DELETE: " + allArgs.parser.method());
auto obj = WebRtcTransportManager::Instance().getItem(allArgs["id"]);
if (!obj) {
invoker(404, headerOut, "id not found");
@ -1919,11 +1898,11 @@ void installWebApi() {
if (!save_name.empty()) {
res_header.emplace("Content-Disposition", "attachment;filename=\"" + save_name + "\"");
}
invoker.responseFile(allArgs.getParser().getHeader(), res_header, allArgs["file_path"]);
invoker.responseFile(allArgs.parser.getHeader(), res_header, allArgs["file_path"]);
}
};
bool flag = NOTICE_EMIT(BroadcastHttpAccessArgs, Broadcast::kBroadcastHttpAccess, allArgs.getParser(), file_path, false, file_invoker, sender);
bool flag = NOTICE_EMIT(BroadcastHttpAccessArgs, Broadcast::kBroadcastHttpAccess, allArgs.parser, file_path, false, file_invoker, sender);
if (!flag) {
// 文件下载鉴权事件无人监听,不允许下载
invoker(401, StrCaseMap {}, "None http access event listener");
@ -1935,28 +1914,23 @@ void installWebApi() {
NoticeCenter::Instance().addListener(nullptr, Broadcast::kBroadcastStreamNoneReader, [](BroadcastStreamNoneReaderArgs) {
auto id = sender.getMediaTuple().stream;
VideoStackManager::Instance().stopVideoStack(id);
InfoL << "VideoStack: " << id <<" stop";
});
api_regist("/index/api/stack/start", [](API_ARGS_JSON_ASYNC) {
CHECK_SECRET();
auto ret = VideoStackManager::Instance().startVideoStack(allArgs.getArgs());
if (!ret) {
invoker(200, headerOut, "success");
} else {
invoker(200, headerOut, "failed");
}
val["code"] = ret;
val["msg"] = ret ? "failed" : "success";
invoker(200, headerOut, val.toStyledString());
});
api_regist("/index/api/stack/stop", [](API_ARGS_MAP_ASYNC) {
CHECK_SECRET();
CHECK_ARGS("id");
auto ret = VideoStackManager::Instance().stopVideoStack(allArgs["id"]);
if (!ret) {
invoker(200, headerOut, "success");
} else {
invoker(200, headerOut, "failed");
}
val["code"] = ret;
val["msg"] = ret ? "failed" : "success";
invoker(200, headerOut, val.toStyledString());
});
#endif
}

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@ -115,72 +115,41 @@ std::string getValue(const mediakit::Parser &parser, Args &args, const First &fi
template<typename Args>
class HttpAllArgs {
mediakit::Parser* _parser = nullptr;
Args* _args = nullptr;
public:
HttpAllArgs(const mediakit::Parser &parser, Args &args) {
_get_args = [&args]() {
return (void *) &args;
};
_get_parser = [&parser]() -> const mediakit::Parser & {
return parser;
};
_get_value = [](HttpAllArgs &that, const std::string &key) {
return getValue(that.getParser(), that.getArgs(), key);
};
_clone = [&](HttpAllArgs &that) {
that._get_args = [args]() {
return (void *) &args;
};
that._get_parser = [parser]() -> const mediakit::Parser & {
return parser;
};
that._get_value = [](HttpAllArgs &that, const std::string &key) {
return getValue(that.getParser(), that.getArgs(), key);
};
that._cache_able = true;
};
}
const mediakit::Parser& parser;
Args& args;
HttpAllArgs(const HttpAllArgs &that) {
if (that._cache_able) {
_get_args = that._get_args;
_get_parser = that._get_parser;
_get_value = that._get_value;
_cache_able = true;
} else {
that._clone(*this);
HttpAllArgs(const mediakit::Parser &p, Args &a): parser(p), args(a) {}
HttpAllArgs(const HttpAllArgs &that): _parser(new mediakit::Parser(that.parser)),
_args(new Args(that.args)),
parser(*_parser), args(*_args) {}
~HttpAllArgs() {
if (_parser) {
delete _parser;
}
if (_args) {
delete _args;
}
}
template<typename Key>
toolkit::variant operator[](const Key &key) const {
return (toolkit::variant)_get_value(*(HttpAllArgs*)this, key);
return (toolkit::variant)getValue(parser, args, key);
}
const mediakit::Parser &getParser() const {
return _get_parser();
}
Args &getArgs() {
return *((Args *) _get_args());
}
const Args &getArgs() const {
return *((Args *) _get_args());
}
private:
bool _cache_able = false;
std::function<void *() > _get_args;
std::function<const mediakit::Parser &() > _get_parser;
std::function<std::string(HttpAllArgs &that, const std::string &key)> _get_value;
std::function<void(HttpAllArgs &that) > _clone;
};
#define API_ARGS_MAP toolkit::SockInfo &sender, mediakit::HttpSession::KeyValue &headerOut, const HttpAllArgs<ApiArgsType> &allArgs, Json::Value &val
using ArgsMap = HttpAllArgs<ApiArgsType>;
using ArgsJson = HttpAllArgs<Json::Value>;
using ArgsString = HttpAllArgs<std::string>;
#define API_ARGS_MAP toolkit::SockInfo &sender, mediakit::HttpSession::KeyValue &headerOut, const ArgsMap &allArgs, Json::Value &val
#define API_ARGS_MAP_ASYNC API_ARGS_MAP, const mediakit::HttpSession::HttpResponseInvoker &invoker
#define API_ARGS_JSON toolkit::SockInfo &sender, mediakit::HttpSession::KeyValue &headerOut, const HttpAllArgs<Json::Value> &allArgs, Json::Value &val
#define API_ARGS_JSON toolkit::SockInfo &sender, mediakit::HttpSession::KeyValue &headerOut, const ArgsJson &allArgs, Json::Value &val
#define API_ARGS_JSON_ASYNC API_ARGS_JSON, const mediakit::HttpSession::HttpResponseInvoker &invoker
#define API_ARGS_STRING toolkit::SockInfo &sender, mediakit::HttpSession::KeyValue &headerOut, const HttpAllArgs<std::string> &allArgs, Json::Value &val
#define API_ARGS_STRING toolkit::SockInfo &sender, mediakit::HttpSession::KeyValue &headerOut, const ArgsString &allArgs, Json::Value &val
#define API_ARGS_STRING_ASYNC API_ARGS_STRING, const mediakit::HttpSession::HttpResponseInvoker &invoker
#define API_ARGS_VALUE sender, headerOut, allArgs, val

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@ -38,7 +38,7 @@
#endif
#if defined(ENABLE_VERSION)
#include "version.h"
#include "ZLMVersion.h"
#endif
#if !defined(_WIN32)

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@ -136,6 +136,15 @@ private:
toolkit::Timer::Ptr _async_close_timer;
};
template <typename MAP, typename KEY, typename TYPE>
static void getArgsValue(const MAP &allArgs, const KEY &key, TYPE &value) {
auto val = ((MAP &)allArgs)[key];
if (!val.empty()) {
value = (TYPE)val;
}
}
class ProtocolOption {
public:
ProtocolOption();
@ -243,15 +252,6 @@ public:
GET_OPT_VALUE(stream_replace);
GET_OPT_VALUE(max_track);
}
private:
template <typename MAP, typename KEY, typename TYPE>
static void getArgsValue(const MAP &allArgs, const KEY &key, TYPE &value) {
auto val = ((MAP &)allArgs)[key];
if (!val.empty()) {
value = (TYPE)val;
}
}
};
//该对象用于拦截感兴趣的MediaSourceEvent事件

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@ -294,8 +294,8 @@ void RtspUrl::setup(bool is_ssl, const string &url, const string &user, const st
splitUrl(ip, ip, port);
_url = std::move(url);
_user = strCoding::UrlDecode(std::move(user));
_passwd = strCoding::UrlDecode(std::move(passwd));
_user = strCoding::UrlDecodeComponent(user);
_passwd = strCoding::UrlDecodeComponent(passwd);
_host = std::move(ip);
_port = port;
_is_ssl = is_ssl;

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@ -30,7 +30,7 @@ struct StrCaseCompare {
class StrCaseMap : public std::multimap<std::string, std::string, StrCaseCompare> {
public:
using Super = multimap<std::string, std::string, StrCaseCompare>;
using Super = std::multimap<std::string, std::string, StrCaseCompare>;
std::string &operator[](const std::string &k) {
auto it = find(k);

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@ -14,7 +14,7 @@
using namespace toolkit;
#if defined(ENABLE_VERSION)
#include "version.h"
#include "ZLMVersion.h"
#endif
extern "C" {

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@ -36,6 +36,16 @@
#define CHECK(exp, ...) ::mediakit::Assert_ThrowCpp(!(exp), #exp, __FUNCTION__, __FILE__, __LINE__, ##__VA_ARGS__)
#endif // CHECK
#ifndef CHECK_RET
#define CHECK_RET(...) \
try { \
CHECK(__VA_ARGS__); \
} catch (AssertFailedException & ex) { \
WarnL << ex.what(); \
return; \
}
#endif
#ifndef MAX
#define MAX(a, b) ((a) > (b) ? (a) : (b))
#endif // MAX

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@ -53,22 +53,6 @@ char HexStrToBin(const char *str) {
return (high << 4) | low;
}
string strCoding::UrlEncode(const string &str) {
string out;
size_t len = str.size();
for (size_t i = 0; i < len; ++i) {
char ch = str[i];
if (isalnum((uint8_t) ch)) {
out.push_back(ch);
} else {
char buf[4];
sprintf(buf, "%%%X%X", (uint8_t) ch >> 4, (uint8_t) ch & 0x0F);
out.append(buf);
}
}
return out;
}
string strCoding::UrlEncodePath(const string &str) {
const char *dont_escape = "!#&'*+:=?@/._-$,;~()";
string out;
@ -103,32 +87,6 @@ string strCoding::UrlEncodeComponent(const string &str) {
return out;
}
string strCoding::UrlDecode(const string &str) {
string output;
size_t i = 0, len = str.length();
while (i < len) {
if (str[i] == '%') {
if (i + 3 > len) {
// %后面必须还有两个字节才会反转义
output.append(str, i, len - i);
break;
}
char ch = HexStrToBin(&(str[i + 1]));
if (ch == -1) {
// %后面两个字节不是16进制字符串转义失败直接拼接3个原始字符
output.append(str, i, 3);
} else {
output += ch;
}
i += 3;
} else {
output += str[i];
++i;
}
}
return output;
}
string strCoding::UrlDecodePath(const string &str) {
const char *dont_unescape = "#$&+,/:;=?@";
string output;
@ -185,27 +143,6 @@ std::string strCoding::UrlDecodeComponent(const std::string &str) {
return output;
}
#if 0
#include "Util/onceToken.h"
static toolkit::onceToken token([]() {
auto str0 = strCoding::UrlDecode(
"rtsp%3A%2F%2Fadmin%3AJm13317934%25jm%40111.47.84.69%3A554%2FStreaming%2FChannels%2F101%3Ftransportmode%3Dunicast%26amp%3Bprofile%3DProfile_1");
auto str1 = strCoding::UrlDecode("%j1"); // 测试%后面两个字节不是16进制字符串
auto str2 = strCoding::UrlDecode("%a"); // 测试%后面字节数不够
auto str3 = strCoding::UrlDecode("%"); // 测试只有%
auto str4 = strCoding::UrlDecode("%%%"); // 测试多个%
auto str5 = strCoding::UrlDecode("%%%%40"); // 测试多个非法%后恢复正常解析
auto str6 = strCoding::UrlDecode("Jm13317934%jm"); // 测试多个非法%后恢复正常解析
cout << str0 << endl;
cout << str1 << endl;
cout << str2 << endl;
cout << str3 << endl;
cout << str4 << endl;
cout << str5 << endl;
cout << str6 << endl;
});
#endif
///////////////////////////////windows专用///////////////////////////////////
#if defined(_WIN32)
void UnicodeToGB2312(char* pOut, wchar_t uData)

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@ -18,10 +18,8 @@ namespace mediakit {
class strCoding {
public:
[[deprecated]] static std::string UrlEncode(const std::string &str); //url utf8编码, deprecated
static std::string UrlEncodePath(const std::string &str); //url路径 utf8编码
static std::string UrlEncodeComponent(const std::string &str); // url参数 utf8编码
[[deprecated]] static std::string UrlDecode(const std::string &str); //url utf8解码, deprecated
static std::string UrlDecodePath(const std::string &str); //url路径 utf8解码
static std::string UrlDecodeComponent(const std::string &str); // url参数 utf8解码
#if defined(_WIN32)

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@ -65,18 +65,18 @@ void HttpRequestSplitter::input(const char *data,size_t len) {
_content_len = onRecvHeader(header_ptr, header_size);
}
if(_remain_data_size <= 0){
//没有剩余数据,清空缓存
_remain_data.clear();
return;
}
/*
*
* HttpRequestSplitter::reset()
*/
tail_ref = tail_tmp;
if(_remain_data_size <= 0){
//没有剩余数据,清空缓存
_remain_data.clear();
return;
}
if(_content_len == 0){
//尚未找到http头缓存定位到剩余数据部分
_remain_data.assign(ptr,_remain_data_size);

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@ -683,18 +683,6 @@ void HttpSession::sendResponse(int code,
AsyncSender::onSocketFlushed(data);
}
string HttpSession::urlDecode(const string &str) {
auto ret = strCoding::UrlDecode(str);
#ifdef _WIN32
GET_CONFIG(string, charSet, Http::kCharSet);
bool isGb2312 = !strcasecmp(charSet.data(), "gb2312");
if (isGb2312) {
ret = strCoding::UTF8ToGB2312(ret);
}
#endif // _WIN32
return ret;
}
string HttpSession::urlDecodePath(const string &str) {
auto ret = strCoding::UrlDecodePath(str);
#ifdef _WIN32

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@ -44,7 +44,6 @@ public:
void onRecv(const toolkit::Buffer::Ptr &) override;
void onError(const toolkit::SockException &err) override;
void onManager() override;
[[deprecated]] static std::string urlDecode(const std::string &str);
static std::string urlDecodePath(const std::string &str);
static std::string urlDecodeComponent(const std::string &str);
void setTimeoutSec(size_t second);

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@ -29,7 +29,13 @@ public:
getRtmpRing()->setDelegate(_media_src);
}
~RtmpMediaSourceMuxer() override { RtmpMuxer::flush(); }
~RtmpMediaSourceMuxer() override {
try {
RtmpMuxer::flush();
} catch (std::exception &ex) {
WarnL << ex.what();
}
}
void setListener(const std::weak_ptr<MediaSourceEvent> &listener){
setDelegate(listener);

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@ -165,14 +165,7 @@ void RtmpProtocol::sendResponse(int type, const string &str) {
void RtmpProtocol::sendInvoke(const string &cmd, const AMFValue &val) {
AMFEncoder enc;
if (val.type() == AMFType::AMF_OBJECT || val.type() == AMFType::AMF_NULL)
{
enc << cmd << ++_send_req_id << val;
}
else
{
enc << cmd << ++_send_req_id << AMFValue() << val;
}
sendRequest(MSG_CMD, enc.data());
}
@ -626,22 +619,12 @@ const char* RtmpProtocol::handle_rtmp(const char *data, size_t len) {
case 12:
chunk_data.is_abs_stamp = true;
chunk_data.stream_index = load_le32(header->stream_index);
_last_stream_index = chunk_data.stream_index;
case 8:
chunk_data.body_size = load_be24(header->body_size);
chunk_data.type_id = header->type_id;
_last_body_size = chunk_data.body_size;
_last_type_id = chunk_data.type_id;
case 4:
chunk_data.ts_field = load_be24(header->time_stamp);
}
switch (header->fmt) {
case 2:
chunk_data.type_id = _last_type_id;
chunk_data.body_size = _last_body_size;
case 1:
chunk_data.stream_index = _last_stream_index;
}
auto time_stamp = chunk_data.ts_field;
if (chunk_data.ts_field == 0xFFFFFF) {

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@ -11,7 +11,6 @@
#ifndef SRC_RTMP_RTMPPROTOCOL_H_
#define SRC_RTMP_RTMPPROTOCOL_H_
#include <cstdint>
#include <memory>
#include <string>
#include <functional>
@ -88,9 +87,6 @@ protected:
private:
bool _data_started = false;
int _now_chunk_id = 0;
uint32_t _last_stream_index = 0;
size_t _last_body_size = 0;
uint8_t _last_type_id = 0;
////////////ChunkSize////////////
size_t _chunk_size_in = DEFAULT_CHUNK_LEN;
size_t _chunk_size_out = DEFAULT_CHUNK_LEN;

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@ -163,18 +163,6 @@ void RtmpPusher::send_connect() {
}
void RtmpPusher::send_createStream() {
// Workaround : 兼容较旧的 FMS3.0
{
{
AMFValue obj(_stream_id);
sendInvoke("releaseStream", obj);
}
{
AMFValue obj(_stream_id);
sendInvoke("FCPublish", obj);
}
}
{
AMFValue obj(AMF_NULL);
sendInvoke("createStream", obj);
addOnResultCB([this](AMFDecoder &dec) {
@ -185,8 +173,6 @@ void RtmpPusher::send_createStream() {
});
}
}
#define RTMP_STREAM_LIVE "live"
void RtmpPusher::send_publish() {
AMFEncoder enc;

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@ -23,10 +23,10 @@ namespace Rtc {
const string kPreferredCodecA = RTC_FIELD "preferredCodecA";
const string kPreferredCodecV = RTC_FIELD "preferredCodecV";
static onceToken token([]() {
mINI::Instance()[kPreferredCodecA] = "PCMU,PCMA,opus,mpeg4-generic";
mINI::Instance()[kPreferredCodecA] = "PCMA,PCMU,opus,mpeg4-generic";
mINI::Instance()[kPreferredCodecV] = "H264,H265,AV1,VP9,VP8";
});
}
} // namespace Rtc
using onCreateSdpItem = function<SdpItem::Ptr(const string &key, const string &value)>;
static map<string, onCreateSdpItem, StrCaseCompare> sdpItemCreator;
@ -73,9 +73,7 @@ public:
class DirectionInterfaceImp : public SdpItem, public DirectionInterface {
public:
DirectionInterfaceImp(RtpDirection direct){
direction = direct;
}
DirectionInterfaceImp(RtpDirection direct) { direction = direct; }
const char *getKey() const override { return getRtpDirectionString(getDirection()); }
RtpDirection getDirection() const override { return direction; }
@ -717,8 +715,7 @@ void SdpAttrCandidate::parse(const string &str) {
string SdpAttrCandidate::toString() const {
if (value.empty()) {
value = foundation + " " + to_string(component) + " " + transport + " " + to_string(priority) +
" " + address + " " + to_string(port) + " typ " + type;
value = foundation + " " + to_string(component) + " " + transport + " " + to_string(priority) + " " + address + " " + to_string(port) + " typ " + type;
for (auto &pr : arr) {
value += ' ';
value += pr.first;
@ -952,8 +949,7 @@ void RtcSession::loadFrom(const string &str) {
if (fmtp_it != fmtp_map.end()) {
plan.fmtp = fmtp_it->second.fmtp;
}
for (auto rtpfb_it = rtcpfb_map.find(pt);
rtpfb_it != rtcpfb_map.end() && rtpfb_it->second.pt == pt; ++rtpfb_it) {
for (auto rtpfb_it = rtcpfb_map.find(pt); rtpfb_it != rtcpfb_map.end() && rtpfb_it->second.pt == pt; ++rtpfb_it) {
plan.rtcp_fb.emplace(rtpfb_it->second.rtcp_type);
}
}
@ -986,8 +982,7 @@ void RtcSdpBase::toRtsp() {
case 'a': {
auto attr = dynamic_pointer_cast<SdpAttr>(*it);
CHECK(attr);
if (!strcasecmp(attr->detail->getKey(), "rtpmap")
|| !strcasecmp(attr->detail->getKey(), "fmtp")) {
if (!strcasecmp(attr->detail->getKey(), "rtpmap") || !strcasecmp(attr->detail->getKey(), "fmtp")) {
++it;
break;
}
@ -1221,9 +1216,9 @@ RtcSessionSdp::Ptr RtcSession::toRtcSessionSdp() const{
}
}
}
if(ice_lite)
if (ice_lite) {
sdp.addAttr(std::make_shared<SdpCommon>("ice-lite"));
}
return ret;
}
@ -1427,16 +1422,14 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){
preferred_codec = s_preferred_codec;
rtcp_fb = { SdpConst::kTWCCRtcpFb, SdpConst::kRembRtcpFb };
extmap = {
{RtpExtType::ssrc_audio_level, RtpDirection::sendrecv},
extmap = { { RtpExtType::ssrc_audio_level, RtpDirection::sendrecv },
{ RtpExtType::csrc_audio_level, RtpDirection::sendrecv },
{ RtpExtType::abs_send_time, RtpDirection::sendrecv },
{ RtpExtType::transport_cc, RtpDirection::sendrecv },
// rtx重传rtp时忽略sdes_mid类型的rtp ext,实测发现Firefox在接收rtx时如果存在sdes_mid的ext,将导致无法播放
//{RtpExtType::sdes_mid,RtpDirection::sendrecv},
{ RtpExtType::sdes_rtp_stream_id, RtpDirection::sendrecv },
{RtpExtType::sdes_repaired_rtp_stream_id, RtpDirection::sendrecv}
};
{ RtpExtType::sdes_repaired_rtp_stream_id, RtpDirection::sendrecv } };
break;
}
case TrackVideo: {
@ -1446,8 +1439,7 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){
preferred_codec = s_preferred_codec;
rtcp_fb = { SdpConst::kTWCCRtcpFb, SdpConst::kRembRtcpFb, "nack", "ccm fir", "nack pli" };
extmap = {
{RtpExtType::abs_send_time, RtpDirection::sendrecv},
extmap = { { RtpExtType::abs_send_time, RtpDirection::sendrecv },
{ RtpExtType::transport_cc, RtpDirection::sendrecv },
// rtx重传rtp时忽略sdes_mid类型的rtp ext,实测发现Firefox在接收rtx时如果存在sdes_mid的ext,将导致无法播放
//{RtpExtType::sdes_mid,RtpDirection::sendrecv},
@ -1460,8 +1452,7 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){
// 手机端推webrtc 会带有旋转角度rtc协议能正常播放 其他协议拉流画面旋转
//{RtpExtType::video_orientation, RtpDirection::sendrecv},
{ RtpExtType::toffset, RtpDirection::sendrecv },
{RtpExtType::framemarking, RtpDirection::sendrecv}
};
{ RtpExtType::framemarking, RtpDirection::sendrecv } };
break;
}
case TrackApplication: {
@ -1471,8 +1462,7 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){
}
}
void RtcConfigure::setDefaultSetting(string ice_ufrag, string ice_pwd, RtpDirection direction,
const SdpAttrFingerprint &fingerprint) {
void RtcConfigure::setDefaultSetting(string ice_ufrag, string ice_pwd, RtpDirection direction, const SdpAttrFingerprint &fingerprint) {
video.setDefaultSetting(TrackVideo);
audio.setDefaultSetting(TrackAudio);
application.setDefaultSetting(TrackApplication);
@ -1683,8 +1673,7 @@ RETRY:
answer_media.role = mathDtlsRole(offer_media.role);
// 如果codec匹配失败那么禁用该track
answer_media.direction = check_codec ? matchDirection(offer_media.direction, configure.direction)
: RtpDirection::inactive;
answer_media.direction = check_codec ? matchDirection(offer_media.direction, configure.direction) : RtpDirection::inactive;
if (answer_media.direction == RtpDirection::invalid) {
continue;
}

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@ -77,11 +77,7 @@ enum class DtlsRole {
actpass,
};
enum class SdpType {
invalid = -1,
offer,
answer
};
enum class SdpType { invalid = -1, offer, answer };
DtlsRole getDtlsRole(const std::string &str);
const char *getDtlsRoleString(DtlsRole role);
@ -92,17 +88,11 @@ class SdpItem {
public:
using Ptr = std::shared_ptr<SdpItem>;
virtual ~SdpItem() = default;
virtual void parse(const std::string &str) {
value = str;
}
virtual std::string toString() const {
return value;
}
virtual void parse(const std::string &str) { value = str; }
virtual std::string toString() const { return value; }
virtual const char *getKey() const = 0;
void reset() {
value.clear();
}
void reset() { value.clear(); }
protected:
mutable std::string value;
@ -259,9 +249,7 @@ public:
void parse(const std::string &str) override;
std::string toString() const override;
const char *getKey() const override { return "msid-semantic"; }
bool empty() const {
return msid.empty();
}
bool empty() const { return msid.empty(); }
};
class SdpAttrRtcp : public SdpItem {
@ -271,12 +259,11 @@ public:
std::string nettype { "IN" };
std::string addrtype { "IP4" };
std::string address { "0.0.0.0" };
void parse(const std::string &str) override;;
void parse(const std::string &str) override;
;
std::string toString() const override;
const char *getKey() const override { return "rtcp"; }
bool empty() const {
return address.empty() || !port;
}
bool empty() const { return address.empty() || !port; }
};
class SdpAttrIceUfrag : public SdpItem {
@ -364,9 +351,9 @@ public:
// a=rtcp-fb:98 nack pli
// a=rtcp-fb:120 nack 支持 nack 重传nack (Negative-Acknowledgment) 。
// a=rtcp-fb:120 nack pli 支持 nack 关键帧重传PLI (Picture Loss Indication) 。
//a=rtcp-fb:120 ccm fir 支持编码层关键帧请求CCM (Codec Control Message)FIR (Full Intra Request ),通常与 nack pli 有同样的效果,但是 nack pli 是用于重传时的关键帧请求。
//a=rtcp-fb:120 goog-remb 支持 REMB (Receiver Estimated Maximum Bitrate) 。
//a=rtcp-fb:120 transport-cc 支持 TCC (Transport Congest Control) 。
// a=rtcp-fb:120 ccm fir 支持编码层关键帧请求CCM (Codec Control Message)FIR (Full Intra Request ),通常与 nack pli 有同样的效果,但是 nack pli
// 是用于重传时的关键帧请求。 a=rtcp-fb:120 goog-remb 支持 REMB (Receiver Estimated Maximum Bitrate) 。 a=rtcp-fb:120 transport-cc 支持 TCC (Transport
// Congest Control) 。
uint8_t pt;
std::string rtcp_type;
void parse(const std::string &str) override;

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@ -27,7 +27,6 @@ protected:
void onRtp(const char *buf, size_t len, uint64_t stamp_ms) override;
void onRtcp(const char *buf, size_t len) override;
void onRecvRtp(MediaTrack &track, const std::string &rid, RtpPacket::Ptr rtp) override {};
void onBeforeEncryptRtp(const char *buf, int &len, void *ctx) override {};
void onBeforeEncryptRtcp(const char *buf, int &len, void *ctx) override {};

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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);

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@ -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:
@ -27,10 +27,9 @@ protected:
void onStartWebRTC() override;
void onDestory() override;
void onRtcConfigure(RtcConfigure &configure) const override;
void onRecvRtp(MediaTrack &track, const std::string &rid, RtpPacket::Ptr rtp) 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

View File

@ -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;

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@ -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;

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@ -378,6 +378,12 @@ void WebRtcTransport::setRemoteDtlsFingerprint(const RtcSession &remote) {
}
void WebRtcTransport::onRtcConfigure(RtcConfigure &configure) const {
SdpAttrFingerprint fingerprint;
fingerprint.algorithm = _offer_sdp->media[0].fingerprint.algorithm;
fingerprint.hash = getFingerprint(fingerprint.algorithm, _dtls_transport);
configure.setDefaultSetting(
_ice_server->GetUsernameFragment(), _ice_server->GetPassword(), RtpDirection::sendrecv, fingerprint);
// 开启remb后关闭twcc因为开启twcc后remb无效
GET_CONFIG(size_t, remb_bit_rate, Rtc::kRembBitRate);
configure.enableTWCC(!remb_bit_rate);
@ -407,12 +413,7 @@ std::string WebRtcTransport::getAnswerSdp(const string &offer) {
setRemoteDtlsFingerprint(*_offer_sdp);
//// sdp 配置 ////
SdpAttrFingerprint fingerprint;
fingerprint.algorithm = _offer_sdp->media[0].fingerprint.algorithm;
fingerprint.hash = getFingerprint(fingerprint.algorithm, _dtls_transport);
RtcConfigure configure;
configure.setDefaultSetting(
_ice_server->GetUsernameFragment(), _ice_server->GetPassword(), RtpDirection::sendrecv, fingerprint);
onRtcConfigure(configure);
//// 生成answer sdp ////
@ -431,10 +432,6 @@ static bool isDtls(char *buf) {
return ((*buf > 19) && (*buf < 64));
}
static string getPeerAddress(RTC::TransportTuple *tuple) {
return tuple->get_peer_ip();
}
void WebRtcTransport::inputSockData(char *buf, int len, RTC::TransportTuple *tuple) {
if (RTC::StunPacket::IsStun((const uint8_t *)buf, len)) {
std::unique_ptr<RTC::StunPacket> packet(RTC::StunPacket::Parse((const uint8_t *)buf, len));
@ -451,7 +448,7 @@ void WebRtcTransport::inputSockData(char *buf, int len, RTC::TransportTuple *tup
}
if (isRtp(buf, len)) {
if (!_srtp_session_recv) {
WarnL << "received rtp packet when dtls not completed from:" << getPeerAddress(tuple);
WarnL << "received rtp packet when dtls not completed from:" << tuple->get_peer_ip();
return;
}
if (_srtp_session_recv->DecryptSrtp((uint8_t *)buf, &len)) {
@ -461,7 +458,7 @@ void WebRtcTransport::inputSockData(char *buf, int len, RTC::TransportTuple *tup
}
if (isRtcp(buf, len)) {
if (!_srtp_session_recv) {
WarnL << "received rtcp packet when dtls not completed from:" << getPeerAddress(tuple);
WarnL << "received rtcp packet when dtls not completed from:" << tuple->get_peer_ip();
return;
}
if (_srtp_session_recv->DecryptSrtcp((uint8_t *)buf, &len)) {
@ -533,8 +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();
}
@ -674,7 +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;
@ -769,7 +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 {
@ -783,12 +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);
}
///////////////////////////////////////////////////////////////////
@ -1278,21 +1278,14 @@ void WebRtcPluginManager::registerPlugin(const string &type, Plugin cb) {
_map_creator[type] = std::move(cb);
}
std::string exchangeSdp(const WebRtcInterface &exchanger, const std::string& offer) {
return const_cast<WebRtcInterface &>(exchanger).getAnswerSdp(offer);
}
void setLocalIp(const WebRtcInterface& exchanger, const std::string& localIp) {
return const_cast<WebRtcInterface &>(exchanger).setLocalIp(localIp);
}
void WebRtcPluginManager::setListener(Listener cb) {
lock_guard<mutex> lck(_mtx_creator);
_listener = std::move(cb);
}
void WebRtcPluginManager::getAnswerSdp(Session &sender, const string &type, const WebRtcArgs &args, const onCreateRtc &cb_in) {
onCreateRtc cb;
void WebRtcPluginManager::negotiateSdp(Session &sender, const string &type, const WebRtcArgs &args, const onCreateWebRtc &cb_in) {
onCreateWebRtc cb;
lock_guard<mutex> lck(_mtx_creator);
if (_listener) {
auto listener = _listener;
@ -1308,21 +1301,19 @@ void WebRtcPluginManager::getAnswerSdp(Session &sender, const string &type, cons
auto it = _map_creator.find(type);
if (it == _map_creator.end()) {
cb(WebRtcException(SockException(Err_other, "the type can not supported")));
cb_in(WebRtcException(SockException(Err_other, "the type can not supported")));
return;
}
it->second(sender, args, cb);
}
void echo_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
void echo_plugin(Session &sender, const WebRtcArgs &args, const onCreateWebRtc &cb) {
cb(*WebRtcEchoTest::create(EventPollerPool::Instance().getPoller()));
}
void push_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
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;
@ -1361,7 +1352,7 @@ void push_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginMana
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);
};
@ -1374,12 +1365,10 @@ void push_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginMana
}
}
void play_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
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;
@ -1395,7 +1384,7 @@ void play_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginMana
}
// 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);
});
};
@ -1408,14 +1397,35 @@ void play_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginMana
}
}
static void set_webrtc_cands(const WebRtcArgs &args, const WebRtcInterface &rtc) {
static void setWebRtcArgs(const WebRtcArgs &args, WebRtcInterface &rtc) {
{
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()), 120, "udp");
auto ice_cand = makeIceCandidate(ip_port[0], atoi(ip_port[1].data()), preferred_tcp ? 100 : 120, "udp");
cands.emplace_back(std::move(*ice_cand));
}
}
@ -1424,23 +1434,26 @@ static void set_webrtc_cands(const WebRtcArgs &args, const WebRtcInterface &rtc)
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");
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
const_cast<WebRtcInterface &>(rtc).setIceCandidate(std::move(cands));
rtc.setIceCandidate(std::move(cands));
}
}
}
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) {
set_webrtc_cands(args, rtc);
setWebRtcArgs(args, const_cast<WebRtcInterface&>(rtc));
});
});

View File

@ -42,13 +42,10 @@ 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) {}
};
std::string exchangeSdp(const WebRtcInterface &exchanger, const std::string& offer);
void setLocalIp(const WebRtcInterface &exchanger, const std::string &localIp);
class WebRtcException : public WebRtcInterface {
public:
WebRtcException(const SockException &ex) : _ex(ex) {};
@ -88,7 +85,7 @@ public:
* @param offer offer sdp
* @return answer sdp
*/
std::string getAnswerSdp(const std::string &offer) override;
std::string getAnswerSdp(const std::string &offer) override final;
/**
* id
@ -252,14 +249,16 @@ 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;
@ -273,7 +272,7 @@ protected:
void onCreate() override;
void onDestory() override;
void onShutdown(const SockException &ex) override;
virtual void onRecvRtp(MediaTrack &track, const std::string &rid, RtpPacket::Ptr rtp) = 0;
virtual void onRecvRtp(MediaTrack &track, const std::string &rid, RtpPacket::Ptr rtp) {}
void updateTicker();
float getLossRate(TrackType type);
void onRtcpBye() override;
@ -289,7 +288,7 @@ 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 +309,8 @@ 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 {
@ -333,21 +332,20 @@ private:
class WebRtcArgs : public std::enable_shared_from_this<WebRtcArgs> {
public:
virtual ~WebRtcArgs() = default;
virtual variant operator[](const std::string &key) const = 0;
};
using onCreateWebRtc = std::function<void(const WebRtcInterface &rtc)>;
class WebRtcPluginManager {
public:
using onCreateRtc = std::function<void(const WebRtcInterface &rtc)>;
using Plugin = std::function<void(Session &sender, const WebRtcArgs &args, const onCreateRtc &cb)>;
using Plugin = std::function<void(Session &sender, const WebRtcArgs &args, const onCreateWebRtc &cb)>;
using Listener = std::function<void(Session &sender, const std::string &type, const WebRtcArgs &args, const WebRtcInterface &rtc)>;
static WebRtcPluginManager &Instance();
void registerPlugin(const std::string &type, Plugin cb);
void getAnswerSdp(Session &sender, const std::string &type, const WebRtcArgs &args, const onCreateRtc &cb);
void setListener(Listener cb);
void negotiateSdp(Session &sender, const std::string &type, const WebRtcArgs &args, const onCreateWebRtc &cb);
private:
WebRtcPluginManager() = default;