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ZLMediaKit
Commit a916760a by xiongziliang
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整理webrtc c接口

parent 426d76f0
正在显示 包含 146 行增加 和 346 行删除
api/CMakeLists.txt
......@@ -28,7 +28,7 @@ file(GLOB API_SRC_LIST
source/*.cpp
source/*.h)
set(LINK_LIBRARIES ${MK_LINK_LIBRARIES} jsoncpp)
set(LINK_LIBRARIES ${MK_LINK_LIBRARIES})
if(IOS)
add_library(mk_api STATIC ${API_SRC_LIST})
......@@ -62,7 +62,7 @@ if(CMAKE_SYSTEM_NAME MATCHES "Linux")
elseif(CMAKE_SYSTEM_NAME MATCHES "Android")
target_link_libraries(mk_api log -Wl,--start-group ${LINK_LIBRARIES} -Wl,--end-group)
else()
target_link_libraries(mk_api jsoncpp ${LINK_LIBRARIES})
target_link_libraries(mk_api ${LINK_LIBRARIES})
endif()
generate_export_header(mk_api
......
api/include/mk_common.h
......@@ -167,6 +167,20 @@ API_EXPORT uint16_t API_CALL mk_rtp_server_start(uint16_t port);
*/
API_EXPORT uint16_t API_CALL mk_rtc_server_start(uint16_t port);
//获取webrtc answer sdp回调函数
typedef void(API_CALL *on_mk_webrtc_get_answer_sdp)(void *user_data, const char *answer, const char *err);
/**
* webrtc交换sdp，根据offer sdp生成answer sdp
* @param user_data 回调用户指针
* @param cb 回调函数
* @param type webrtc插件类型，支持echo,play,push
* @param offer webrtc offer sdp
* @param url rtc url, 例如 rtc://__defaultVhost/app/stream?key1=val1&key2=val2
*/
API_EXPORT void API_CALL mk_webrtc_get_answer_sdp(void *user_data, on_mk_webrtc_get_answer_sdp cb, const char *type,
const char *offer, const char *url);
/**
* 创建srt服务器
* @param port srt监听端口
......
api/include/mk_events_objects.h
......@@ -82,8 +82,6 @@ API_EXPORT uint16_t API_CALL mk_media_info_get_port(const mk_media_info ctx);
typedef void* mk_media_source;
//查找MediaSource的回调函数
typedef void(API_CALL *on_mk_media_source_find_cb)(void *user_data, const mk_media_source ctx);
//生成AnswerSdp回调函数
typedef void(API_CALL *on_mk_media_source_answersdp_cb)(void *user_data, const mk_media_source ctx, const char *answer, const char *err);
//MediaSource::getSchema()
API_EXPORT const char* API_CALL mk_media_source_get_schema(const mk_media_source ctx);
......@@ -136,9 +134,6 @@ API_EXPORT void API_CALL mk_media_source_find(const char *schema,
//MediaSource::for_each_media()
API_EXPORT void API_CALL mk_media_source_for_each(void *user_data, on_mk_media_source_find_cb cb, const char *schema,
const char *vhost, const char *app, const char *stream);
//MediaSource::find() + WebRtcPlayer::create + getOffer
API_EXPORT void API_CALL mk_media_source_answersdp(void *user_data, on_mk_media_source_answersdp_cb cb, const char *offer,
const char *schema, const char *vhost, const char *app, const char *stream, int from_mp4);
///////////////////////////////////////////HttpBody/////////////////////////////////////////////
//HttpBody对象的C映射
......@@ -182,14 +177,6 @@ API_EXPORT void API_CALL mk_http_response_invoker_do(const mk_http_response_invo
int response_code,
const char **response_header,
const mk_http_body response_body);
/**
* HttpSession::HttpResponseInvoker(const string &codeOut, const StrCaseMap &headerOut, const HttpBody::Ptr &body);
Parser();
SockInfo();
*/
API_EXPORT void API_CALL mk_webrtc_http_response_invoker_do( const mk_http_response_invoker invoker,
const mk_parser parser,
const mk_sock_info sender);
/**
* HttpSession::HttpResponseInvoker(const string &codeOut, const StrCaseMap &headerOut, const string &body);
......
api/source/mk_common.cpp
......@@ -261,11 +261,48 @@ API_EXPORT uint16_t API_CALL mk_rtc_server_start(uint16_t port) {
return 0;
}
#else
WarnL << "未启用该功能!";
WarnL << "未启用webrtc功能, 编译时请开启ENABLE_WEBRTC";
return 0;
#endif
}
class WebRtcArgsUrl : public mediakit::WebRtcArgs {
public:
WebRtcArgsUrl(std::string url) { _url = std::move(url); }
~WebRtcArgsUrl() = default;
toolkit::variant operator[](const std::string &key) const override {
if (key == "url") {
return _url;
}
return "";
}
private:
std::string _url;
};
API_EXPORT void API_CALL mk_webrtc_get_answer_sdp(void *user_data, on_mk_webrtc_get_answer_sdp cb, const char *type,
const char *offer, const char *url) {
#ifdef ENABLE_WEBRTC
assert(type && offer && url && cb);
auto session = std::make_shared<HttpSession>(Socket::createSocket());
std::string offer_str = offer;
WebRtcPluginManager::Instance().getAnswerSdp(*session, type, WebRtcArgsUrl(url),
[offer_str, session, user_data, cb](const WebRtcInterface &exchanger) mutable {
std::string sdp_answer;
try {
sdp_answer = const_cast<WebRtcInterface &>(exchanger).getAnswerSdp(offer_str);
cb(user_data, sdp_answer.data(), nullptr);
} catch (std::exception &ex) {
cb(user_data, nullptr, ex.what());
}
});
#else
WarnL << "未启用webrtc功能, 编译时请开启ENABLE_WEBRTC";
#endif
}
API_EXPORT uint16_t API_CALL mk_srt_server_start(uint16_t port) {
#ifdef ENABLE_SRT
try {
......
api/source/mk_events_objects.cpp
......@@ -18,11 +18,6 @@
#include "Http/HttpClient.h"
#include "Rtsp/RtspSession.h"
#ifdef ENABLE_WEBRTC
#include "jsoncpp/json.h"
#include "mk_webrtc_private.h"
#endif
using namespace toolkit;
using namespace mediakit;
......@@ -257,25 +252,6 @@ API_EXPORT void API_CALL mk_media_source_for_each(void *user_data, on_mk_media_s
}, schema ? schema : "", vhost ? vhost : "", app ? app : "", stream ? stream : "");
}
API_EXPORT void API_CALL mk_media_source_answersdp(void *user_data, on_mk_media_source_answersdp_cb cb, const char *offer,
const char *schema, const char *vhost, const char *app, const char *stream, int from_mp4) {
#ifdef ENABLE_WEBRTC
assert(offer && schema && vhost && app && stream && cb);
auto srcfound = MediaSource::find(schema, vhost, app, stream, from_mp4);
mediakit::MediaInfo info;
info._schema = RTC_SCHEMA;
info._host = vhost;
info._app = app;
info._streamid = stream;
try {
auto rtc = WebRtcPlayer::create(EventPollerPool::Instance().getPoller(), std::dynamic_pointer_cast<RtspMediaSource>(srcfound), info);
cb(user_data, srcfound.get(), rtc->getAnswerSdp(offer).c_str(), "");
} catch (std::exception &ex) {
cb(user_data, nullptr, nullptr, ex.what());
}
#endif
}
///////////////////////////////////////////HttpBody/////////////////////////////////////////////
API_EXPORT mk_http_body API_CALL mk_http_body_from_string(const char *str, size_t len){
assert(str);
......@@ -349,53 +325,6 @@ API_EXPORT void API_CALL mk_http_response_invoker_do(const mk_http_response_invo
(*invoker)(response_code,header,*body);
}
API_EXPORT void API_CALL mk_webrtc_http_response_invoker_do(const mk_http_response_invoker ctx_invoker,
const mk_parser ctx_parser,
const mk_sock_info ctx_sock ) {
assert(ctx_parser && ctx_invoker && ctx_sock);
#ifdef ENABLE_WEBRTC
static auto webrtc_cb = [](API_ARGS_STRING_ASYNC){
CHECK_ARGS("type");
auto type = allArgs["type"];
auto offer = allArgs.getArgs();
CHECK(!offer.empty(), "http body(webrtc offer sdp) is empty");
WebRtcPluginManager::Instance().getAnswerSdp(
*(static_cast<Session *>(&sender)), type, offer, WebRtcArgsImp(allArgs, sender.getIdentifier()),
[invoker, val, offer, headerOut](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"] = "*";
try {
val["sdp"] = const_cast<WebRtcInterface &>(exchanger).getAnswerSdp(offer);
val["id"] = exchanger.getIdentifier();
val["type"] = "answer";
invoker(200, headerOut, val.toStyledString());
} catch (std::exception &ex) {
val["code"] = API::Exception;
val["msg"] = ex.what();
invoker(200, headerOut, val.toStyledString());
}
});
};
Parser *parser = (Parser *)ctx_parser;
HttpSession::HttpResponseInvoker *invoker = (HttpSession::HttpResponseInvoker *)ctx_invoker;
SockInfo* sender = (SockInfo*)ctx_sock;
GET_CONFIG(std::string, charSet, Http::kCharSet);
HttpSession::KeyValue headerOut;
headerOut["Content-Type"] = std::string("application/json; charset=") + charSet;
Json::Value val;
val["code"] = API::Success;
webrtc_cb(*sender, headerOut, HttpAllArgs<std::string>(*parser, (std::string &)parser->Content()), val, *invoker);
#endif
};
API_EXPORT mk_http_response_invoker API_CALL mk_http_response_invoker_clone(const mk_http_response_invoker ctx){
assert(ctx);
HttpSession::HttpResponseInvoker *invoker = (HttpSession::HttpResponseInvoker *)ctx;
......
api/source/mk_webrtc_private.h deleted 100644 → 0
/*
* Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved.
*
* This file is part of ZLMediaKit(https://github.com/xia-chu/ZLMediaKit).
*
* Use of this source code is governed by MIT license that can be found in the
* LICENSE file in the root of the source tree. All contributing project authors
* may be found in the AUTHORS file in the root of the source tree.
*/
#ifdef ENABLE_WEBRTC
#ifndef MK_WEBRTC_API_H
#define MK_WEBRTC_API_H
#include "Common/Parser.h"
#include "Http/HttpSession.h"
#include "Network/Socket.h"
#include "jsoncpp/json.h"
#include <functional>
#include <string>
#include "../webrtc/WebRtcEchoTest.h"
#include "../webrtc/WebRtcPlayer.h"
#include "../webrtc/WebRtcPusher.h"
#include "../webrtc/WebRtcTransport.h"
using namespace mediakit;
namespace API {
typedef enum {
NotFound = -500, //未找到
Exception = -400, //代码抛异常
InvalidArgs = -300, //参数不合法
SqlFailed = -200, // sql执行失败
AuthFailed = -100, //鉴权失败
OtherFailed = -1, //业务代码执行失败，
Success = 0 //执行成功
} ApiErr;
} // namespace API
class ApiRetException : public std::runtime_error {
public:
ApiRetException(const char *str = "success", int code = API::Success)
: runtime_error(str) {
_code = code;
}
~ApiRetException() = default;
int code() { return _code; }
private:
int _code;
};
class AuthException : public ApiRetException {
public:
AuthException(const char *str)
: ApiRetException(str, API::AuthFailed) {}
~AuthException() = default;
};
class InvalidArgsException : public ApiRetException {
public:
InvalidArgsException(const char *str)
: ApiRetException(str, API::InvalidArgs) {}
~InvalidArgsException() = default;
};
class SuccessException : public ApiRetException {
public:
SuccessException()
: ApiRetException("success", API::Success) {}
~SuccessException() = default;
};
using ApiArgsType = std::map<std::string, std::string, mediakit::StrCaseCompare>;
template <typename Args, typename First>
std::string getValue(Args &args, const First &first) {
return args[first];
}
template <typename First>
std::string getValue(Json::Value &args, const First &first) {
return args[first].asString();
}
template <typename First>
std::string getValue(std::string &args, const First &first) {
return "";
}
template <typename First>
std::string getValue(const mediakit::Parser &parser, const First &first) {
auto ret = parser.getUrlArgs()[first];
if (!ret.empty()) {
return ret;
}
return parser.getHeader()[first];
}
template <typename First>
std::string getValue(mediakit::Parser &parser, const First &first) {
return getValue((const mediakit::Parser &)parser, first);
}
template <typename Args, typename First>
std::string getValue(const mediakit::Parser &parser, Args &args, const First &first) {
auto ret = getValue(args, first);
if (!ret.empty()) {
return ret;
}
return getValue(parser, first);
}
template <typename Args>
class HttpAllArgs {
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;
};
}
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() = default;
template <typename Key>
toolkit::variant operator[](const Key &key) const {
return (toolkit::variant)_get_value(*(HttpAllArgs *)this, 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
#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_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_ASYNC API_ARGS_STRING, const mediakit::HttpSession::HttpResponseInvoker &invoker
#define API_ARGS_VALUE sender, headerOut, allArgs, val
//注册http请求参数是http原始请求信息的异步回复的http api
void api_regist(const std::string &api_path, const std::function<void(API_ARGS_STRING_ASYNC)> &func);
template <typename Args, typename First>
bool checkArgs(Args &args, const First &first) {
return !args[first].empty();
}
template <typename Args, typename First, typename... KeyTypes>
bool checkArgs(Args &args, const First &first, const KeyTypes &...keys) {
return checkArgs(args, first) && checkArgs(args, keys...);
}
//检查http url中或body中或http header参数是否为空的宏
#define CHECK_ARGS(...) \
if (!checkArgs(allArgs, ##__VA_ARGS__)) { \
throw InvalidArgsException("缺少必要参数:" #__VA_ARGS__); \
}
#ifdef ENABLE_WEBRTC
class WebRtcArgsImp : public WebRtcArgs {
public:
WebRtcArgsImp(const HttpAllArgs<std::string> &args, std::string session_id)
: _args(args)
, _session_id(std::move(session_id)) {}
~WebRtcArgsImp() override = default;
variant operator[](const std::string &key) const override {
if (key == "url") {
return getUrl();
}
return _args[key];
}
private:
std::string getUrl() const {
auto &allArgs = _args;
CHECK_ARGS("app", "stream");
return StrPrinter << RTC_SCHEMA << "://" << _args["Host"] << "/" << _args["app"] << "/" << _args["stream"]
<< "?" << _args.getParser().Params() + "&session=" + _session_id;
}
private:
HttpAllArgs<std::string> _args;
std::string _session_id;
};
#endif
#endif // MK_WEBRTC_API_H
#endif // ENABLE_WEBRTC
\ No newline at end of file
api/tests/server.c
......@@ -122,6 +122,72 @@ void API_CALL on_mk_media_no_reader(const mk_media_source sender) {
mk_media_source_get_stream(sender));
}
//按照json转义规则转义webrtc answer sdp
static char *escape_string(const char *ptr){
char *escaped = malloc(2 * strlen(ptr));
char *ptr_escaped = escaped;
while (1) {
switch (*ptr) {
case '\r': {
*(ptr_escaped++) = '\\';
*(ptr_escaped++) = 'r';
break;
}
case '\n': {
*(ptr_escaped++) = '\\';
*(ptr_escaped++) = 'n';
break;
}
case '\t': {
*(ptr_escaped++) = '\\';
*(ptr_escaped++) = 't';
break;
}
default: {
*(ptr_escaped++) = *ptr;
if (!*ptr) {
return escaped;
}
break;
}
}
++ptr;
}
}
static void on_mk_webrtc_get_answer_sdp_func(void *user_data, const char *answer, const char *err) {
const char *response_header[] = { "Content-Type", "application/json", "Access-Control-Allow-Origin", "*" , NULL};
if (answer) {
answer = escape_string(answer);
}
size_t len = answer ? 2 * strlen(answer) : 1024;
char *response_content = (char *)malloc(len);
if (answer) {
snprintf(response_content, len,
"{"
"\"sdp\":\"%s\","
"\"type\":\"answer\","
"\"code\":0"
"}",
answer);
} else {
snprintf(response_content, len,
"{"
"\"msg\":\"%s\","
"\"code\":-1"
"}",
err);
}
mk_http_response_invoker_do_string(user_data, 200, response_header, response_content);
mk_http_response_invoker_clone_release(user_data);
free(response_content);
if (answer) {
free(answer);
}
}
/**
* 收到http api请求广播(包括GET/POST)
* @param parser http请求内容对象
......@@ -156,7 +222,7 @@ void API_CALL on_mk_http_request(const mk_parser parser,
*consumed = 1;
//拦截api: /api/test
if(strcmp(url,"/api/test") == 0) {
if (strcmp(url, "/api/test") == 0) {
const char *response_header[] = { "Content-Type", "text/html", NULL };
const char *content = "<html>"
"<head>"
......@@ -171,12 +237,16 @@ void API_CALL on_mk_http_request(const mk_parser parser,
mk_http_body body = mk_http_body_from_string(content, 0);
mk_http_response_invoker_do(invoker, 200, response_header, body);
mk_http_body_release(body);
}
//拦截api: /index/api/webrtc
else if(strcmp(url,"/index/api/webrtc") == 0){
mk_webrtc_http_response_invoker_do(invoker,parser,sender);
}
else{
} else if (strcmp(url, "/index/api/webrtc") == 0) {
//拦截api: /index/api/webrtc
char rtc_url[1024];
snprintf(rtc_url, sizeof(rtc_url), "rtc://%s/%s/%s?%s", mk_parser_get_header(parser, "Host"),
mk_parser_get_url_param(parser, "app"), mk_parser_get_url_param(parser, "stream"),
mk_parser_get_url_params(parser));
mk_webrtc_get_answer_sdp(mk_http_response_invoker_clone(invoker), on_mk_webrtc_get_answer_sdp_func,
mk_parser_get_url_param(parser, "type"), mk_parser_get_content(parser, NULL), rtc_url);
} else {
*consumed = 0;
return;
}
......
server/WebApi.cpp
......@@ -1578,9 +1578,9 @@ void installWebApi() {
auto offer = allArgs.getArgs();
CHECK(!offer.empty(), "http body(webrtc offer sdp) is empty");
WebRtcPluginManager::Instance().getAnswerSdp(
*(static_cast<Session *>(&sender)), type, offer, WebRtcArgsImp(allArgs, sender.getIdentifier()),
[invoker, val, offer, headerOut](const WebRtcInterface &exchanger) mutable {
WebRtcPluginManager::Instance().getAnswerSdp(*(static_cast<Session *>(&sender)), type,
WebRtcArgsImp(allArgs, sender.getIdentifier()),
[invoker, val, offer, headerOut](const WebRtcInterface &exchanger) mutable {
//设置返回类型
headerOut["Content-Type"] = HttpFileManager::getContentType(".json");
//设置跨域
......
webrtc/WebRtcTransport.cpp
......@@ -1114,27 +1114,23 @@ void WebRtcPluginManager::registerPlugin(const string &type, Plugin cb) {
_map_creator[type] = std::move(cb);
}
void WebRtcPluginManager::getAnswerSdp(
Session &sender, const string &type, const string &offer, const WebRtcArgs &args, const onCreateRtc &cb) {
void WebRtcPluginManager::getAnswerSdp(Session &sender, const string &type, const WebRtcArgs &args, const onCreateRtc &cb) {
lock_guard<mutex> lck(_mtx_creator);
auto it = _map_creator.find(type);
if (it == _map_creator.end()) {
cb(WebRtcException(SockException(Err_other, "the type can not supported")));
return;
}
it->second(sender, offer, args, cb);
it->second(sender, args, cb);
}
void echo_plugin(
Session &sender, const string &offer, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
void echo_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
cb(*WebRtcEchoTest::create(EventPollerPool::Instance().getPoller()));
}
void push_plugin(
Session &sender, const string &offer_sdp, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
void push_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
MediaInfo info(args["url"]);
Broadcast::PublishAuthInvoker invoker = [cb, offer_sdp,
info](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;
......@@ -1173,26 +1169,23 @@ void push_plugin(
push_src_ownership = push_src->getOwnership();
push_src->setProtocolOption(option);
}
auto rtc
= WebRtcPusher::create(EventPollerPool::Instance().getPoller(), push_src, push_src_ownership, info, option);
auto rtc = WebRtcPusher::create(EventPollerPool::Instance().getPoller(), push_src, push_src_ownership, info, option);
push_src->setListener(rtc);
cb(*rtc);
};
// rtsp推流需要鉴权
auto flag = NoticeCenter::Instance().emitEvent(
Broadcast::kBroadcastMediaPublish, MediaOriginType::rtc_push, info, invoker, static_cast<SockInfo &>(sender));
auto flag = NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastMediaPublish, MediaOriginType::rtc_push, info, invoker, static_cast<SockInfo &>(sender));
if (!flag) {
// 该事件无人监听,默认不鉴权
invoker("", ProtocolOption());
}
}
void play_plugin(
Session &sender, const string &offer_sdp, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
void play_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
MediaInfo info(args["url"]);
auto session_ptr = sender.shared_from_this();
Broadcast::AuthInvoker invoker = [cb, offer_sdp, info, session_ptr](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;
......@@ -1214,8 +1207,7 @@ void play_plugin(
};
// 广播通用播放url鉴权事件
auto flag = NoticeCenter::Instance().emitEvent(
Broadcast::kBroadcastMediaPlayed, info, invoker, static_cast<SockInfo &>(sender));
auto flag = NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastMediaPlayed, info, invoker, static_cast<SockInfo &>(sender));
if (!flag) {
// 该事件无人监听,默认不鉴权
invoker("");
......
webrtc/WebRtcTransport.h
......@@ -332,12 +332,12 @@ public:
class WebRtcPluginManager {
public:
using onCreateRtc = std::function<void(const WebRtcInterface &rtc)>;
using Plugin = std::function<void(Session &sender, const std::string &offer, const WebRtcArgs &args, const onCreateRtc &cb)>;
using Plugin = std::function<void(Session &sender, const WebRtcArgs &args, const onCreateRtc &cb)>;
static WebRtcPluginManager &Instance();
void registerPlugin(const std::string &type, Plugin cb);
void getAnswerSdp(Session &sender, const std::string &type, const std::string &offer, const WebRtcArgs &args, const onCreateRtc &cb);
void getAnswerSdp(Session &sender, const std::string &type, const WebRtcArgs &args, const onCreateRtc &cb);
private:
WebRtcPluginManager() = default;
......
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