完善startSendRtp接口

api/source/mk_events_objects.cpp

@@ -211,7 +211,14 @@ API_EXPORT int API_CALL mk_media_source_seek_to(const mk_media_source ctx,uint32
 API_EXPORT void API_CALL mk_media_source_start_send_rtp(const mk_media_source ctx, const char *dst_url, uint16_t dst_port, const char *ssrc, int is_udp, on_mk_media_source_send_rtp_result cb, void *user_data){
     assert(ctx && dst_url && ssrc);
     MediaSource *src = (MediaSource *)ctx;
-    src->startSendRtp(dst_url, dst_port, ssrc, is_udp, 0, [cb, user_data](uint16_t local_port, const SockException &ex){
+
+    MediaSourceEvent::SendRtpArgs args;
+    args.dst_url = dst_url;
+    args.dst_port = dst_port;
+    args.ssrc = ssrc;
+    args.is_udp = is_udp;
+
+    src->startSendRtp(args, [cb, user_data](uint16_t local_port, const SockException &ex){
         if (cb) {
             cb(user_data, local_port, ex.getErrCode(), ex.what());
         }

api/source/mk_media.cpp

@@ -239,8 +239,15 @@ API_EXPORT int API_CALL mk_media_input_audio(mk_media ctx, const void* data, int
 API_EXPORT void API_CALL mk_media_start_send_rtp(mk_media ctx, const char *dst_url, uint16_t dst_port, const char *ssrc, int is_udp, on_mk_media_send_rtp_result cb, void *user_data){
     assert(ctx && dst_url && ssrc);
     MediaHelper::Ptr* obj = (MediaHelper::Ptr*) ctx;
+
+    MediaSourceEvent::SendRtpArgs args;
+    args.dst_url = dst_url;
+    args.dst_port = dst_port;
+    args.ssrc = ssrc;
+    args.is_udp = is_udp;
+
     //sender参数无用
-    (*obj)->getChannel()->startSendRtp(*MediaSource::NullMediaSource, dst_url, dst_port, ssrc, is_udp, 0, [cb, user_data](uint16_t local_port, const SockException &ex){
+    (*obj)->getChannel()->startSendRtp(*MediaSource::NullMediaSource, args, [cb, user_data](uint16_t local_port, const SockException &ex){
         if (cb) {
             cb(user_data, local_port, ex.getErrCode(), ex.what());
         }

postman/ZLMediaKit.postman_collection.json

@@ -1481,6 +1481,24 @@
 							"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
 						}
 					]
 				}

server/WebApi.cpp

@@ -1104,19 +1104,26 @@ void installWebApi() {
         if (!src) {
             throw ApiRetException("该媒体流不存在", API::OtherFailed);
         }
-        uint8_t pt = allArgs["pt"].empty() ? 96 : allArgs["pt"].as<int>();
+
-        bool use_ps = allArgs["use_ps"].empty() ? true : allArgs["use_ps"].as<bool>();
+        MediaSourceEvent::SendRtpArgs args;
-        bool only_audio = allArgs["only_audio"].empty() ? true : allArgs["only_audio"].as<bool>();
+        args.dst_url = allArgs["dst_url"];
-        TraceL << "pt "<<int(pt)<<" ps "<<use_ps<<" audio "<<only_audio;
+        args.dst_port = allArgs["dst_port"];
-        //src_port为空时，则随机本地端口
+        args.ssrc = allArgs["ssrc"];
-        src->startSendRtp(allArgs["dst_url"], allArgs["dst_port"], allArgs["ssrc"], allArgs["is_udp"], allArgs["src_port"], [val, headerOut, invoker](uint16_t local_port, const SockException &ex) mutable{
+        args.is_udp = allArgs["is_udp"];
+        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 << "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());
-        },pt,use_ps,only_audio);
+        });
     });
 
     api_regist("/index/api/stopSendRtp",[](API_ARGS_MAP){

src/Common/MediaSource.cpp

@@ -237,13 +237,13 @@ bool MediaSource::isRecording(Recorder::type type){
     return listener->isRecording(*this, type);
 }
 
-void MediaSource::startSendRtp(const string &dst_url, uint16_t dst_port, const string &ssrc, bool is_udp, uint16_t src_port, const function<void(uint16_t local_port, const SockException &ex)> &cb,uint8_t pt, bool use_ps,bool only_audio){
+void MediaSource::startSendRtp(const MediaSourceEvent::SendRtpArgs &args, const std::function<void(uint16_t, const toolkit::SockException &)> cb) {
     auto listener = _listener.lock();
     if (!listener) {
         cb(0, SockException(Err_other, "尚未设置事件监听器"));
         return;
     }
-    return listener->startSendRtp(*this, dst_url, dst_port, ssrc, is_udp, src_port, cb, pt, use_ps, only_audio);
+    return listener->startSendRtp(*this, args, cb);
 }
 
 bool MediaSource::stopSendRtp(const string &ssrc) {
@@ -720,12 +720,12 @@ vector<Track::Ptr> MediaSourceEventInterceptor::getMediaTracks(MediaSource &send
     return listener->getMediaTracks(sender, trackReady);
 }
 
-void MediaSourceEventInterceptor::startSendRtp(MediaSource &sender, const string &dst_url, uint16_t dst_port, const string &ssrc, bool is_udp, uint16_t src_port, const function<void(uint16_t local_port, const SockException &ex)> &cb, uint8_t pt, bool use_ps,bool only_audio ){
+void MediaSourceEventInterceptor::startSendRtp(MediaSource &sender, const MediaSourceEvent::SendRtpArgs &args, const std::function<void(uint16_t, const toolkit::SockException &)> cb) {
     auto listener = _listener.lock();
     if (listener) {
-        listener->startSendRtp(sender, dst_url, dst_port, ssrc, is_udp, src_port, cb, pt, use_ps, only_audio);
+        listener->startSendRtp(sender, args, cb);
     } else {
-        MediaSourceEvent::startSendRtp(sender, dst_url, dst_port, ssrc, is_udp, src_port, cb, pt, use_ps, only_audio);
+        MediaSourceEvent::startSendRtp(sender, args, cb);
     }
 }
 

src/Common/MediaSource.h

@@ -50,7 +50,7 @@ enum class MediaOriginType : uint8_t {
 std::string getOriginTypeString(MediaOriginType type);
 
 class MediaSource;
-class MediaSourceEvent{
+class MediaSourceEvent {
 public:
     friend class MediaSource;
     MediaSourceEvent(){};
@@ -85,8 +85,29 @@ public:
     virtual bool isRecording(MediaSource &sender, Recorder::type type) { return false; };
     // 获取所有track相关信息
     virtual std::vector<Track::Ptr> getMediaTracks(MediaSource &sender, bool trackReady = true) const { return std::vector<Track::Ptr>(); };
+
+    class SendRtpArgs {
+    public:
+        // 是否采用udp方式发送rtp
+        bool is_udp = true;
+        // rtp采用ps还是es方式
+        bool use_ps = true;
+        //发送es流时指定是否只发送纯音频流
+        bool only_audio = true;
+        // rtp payload type
+        uint8_t pt = 96;
+        // 指定rtp ssrc
+        std::string ssrc;
+        // 指定本地发送端口
+        uint16_t src_port = 0;
+        // 发送目标端口
+        uint16_t dst_port;
+        // 发送目标主机地址，可以是ip或域名
+        std::string dst_url;
+    };
+
     // 开始发送ps-rtp
-    virtual void startSendRtp(MediaSource &sender, const std::string &dst_url, uint16_t dst_port, const std::string &ssrc, bool is_udp, uint16_t src_port, const std::function<void(uint16_t local_port, const toolkit::SockException &ex)> &cb, uint8_t pt=96, bool use_ps = true,bool only_audio = true) { cb(0, toolkit::SockException(toolkit::Err_other, "not implemented"));};
+    virtual void startSendRtp(MediaSource &sender, const SendRtpArgs &args, const std::function<void(uint16_t, const toolkit::SockException &)> cb) { cb(0, toolkit::SockException(toolkit::Err_other, "not implemented"));};
     // 停止发送ps-rtp
     virtual bool stopSendRtp(MediaSource &sender, const std::string &ssrc) {return false; }
 
@@ -117,7 +138,7 @@ public:
     bool setupRecord(MediaSource &sender, Recorder::type type, bool start, const std::string &custom_path, size_t max_second) override;
     bool isRecording(MediaSource &sender, Recorder::type type) override;
     std::vector<Track::Ptr> getMediaTracks(MediaSource &sender, bool trackReady = true) const override;
-    void startSendRtp(MediaSource &sender, const std::string &dst_url, uint16_t dst_port, const std::string &ssrc, bool is_udp, uint16_t src_port, const std::function<void(uint16_t local_port, const toolkit::SockException &ex)> &cb, uint8_t pt=96, bool use_ps = true,bool only_audio = true) override;
+    void startSendRtp(MediaSource &sender, const SendRtpArgs &args, const std::function<void(uint16_t, const toolkit::SockException &)> cb) override;
     bool stopSendRtp(MediaSource &sender, const std::string &ssrc) override;
 
 private:
@@ -269,7 +290,7 @@ public:
     // 获取录制状态
     bool isRecording(Recorder::type type);
     // 开始发送ps-rtp
-    void startSendRtp(const std::string &dst_url, uint16_t dst_port, const std::string &ssrc, bool is_udp, uint16_t src_port, const std::function<void(uint16_t local_port, const toolkit::SockException &ex)> &cb , uint8_t pt = 96, bool use_ps = true,bool only_audio = true);
+    void startSendRtp(const MediaSourceEvent::SendRtpArgs &args, const std::function<void(uint16_t, const toolkit::SockException &)> cb);
     // 停止发送ps-rtp
     bool stopSendRtp(const std::string &ssrc);
 

src/Common/MultiMediaSourceMuxer.cpp

@@ -213,11 +213,11 @@ bool MultiMediaSourceMuxer::isRecording(MediaSource &sender, Recorder::type type
     }
 }
 
-void MultiMediaSourceMuxer::startSendRtp(MediaSource &, const string &dst_url, uint16_t dst_port, const string &ssrc, bool is_udp, uint16_t src_port, const function<void(uint16_t local_port, const SockException &ex)> &cb ,uint8_t pt, bool use_ps,bool only_audio){
+void MultiMediaSourceMuxer::startSendRtp(MediaSource &, const MediaSourceEvent::SendRtpArgs &args, const std::function<void(uint16_t, const toolkit::SockException &)> cb) {
 #if defined(ENABLE_RTPPROXY)
-    RtpSender::Ptr rtp_sender = std::make_shared<RtpSender>(atoi(ssrc.data()),pt,use_ps,only_audio);
+    auto rtp_sender = std::make_shared<RtpSender>();
     weak_ptr<MultiMediaSourceMuxer> weak_self = shared_from_this();
-    rtp_sender->startSend(dst_url, dst_port, is_udp, src_port, [weak_self, rtp_sender, cb, ssrc](uint16_t local_port, const SockException &ex) {
+    rtp_sender->startSend(args, [args, weak_self, rtp_sender, cb](uint16_t local_port, const SockException &ex) {
         cb(local_port, ex);
         auto strong_self = weak_self.lock();
         if (!strong_self || ex) {
@@ -228,7 +228,7 @@ void MultiMediaSourceMuxer::startSendRtp(MediaSource &, const string &dst_url, u
         }
         rtp_sender->addTrackCompleted();
         lock_guard<mutex> lck(strong_self->_rtp_sender_mtx);
-        strong_self->_rtp_sender[ssrc] = rtp_sender;
+        strong_self->_rtp_sender[args.ssrc] = rtp_sender;
     });
 #else
     cb(0, SockException(Err_other, "该功能未启用，编译时请打开ENABLE_RTPPROXY宏"));

src/Common/MultiMediaSourceMuxer.h

@@ -134,7 +134,7 @@ public:
      * @param is_udp 是否为udp
      * @param cb 启动成功或失败回调
      */
-    void startSendRtp(MediaSource &sender, const std::string &dst_url, uint16_t dst_port, const std::string &ssrc, bool is_udp, uint16_t src_port, const std::function<void(uint16_t local_port, const toolkit::SockException &ex)> &cb ,uint8_t pt = 96, bool use_ps = true,bool only_audio = true) override;
+    void startSendRtp(MediaSource &sender, const MediaSourceEvent::SendRtpArgs &args, const std::function<void(uint16_t, const toolkit::SockException &)> cb) override;
 
     /**
      * 停止ps-rtp发送

src/Rtp/RtpCache.h

@@ -51,7 +51,7 @@ protected:
 
 class RtpCacheRaw : public RtpCache, public RawEncoderImp{
 public:
-    RtpCacheRaw(onFlushed cb, uint32_t ssrc, uint8_t payload_type = 96,bool sendAudio = true) : RtpCache(std::move(cb)), RawEncoderImp(ssrc, payload_type,sendAudio) {};
+    RtpCacheRaw(onFlushed cb, uint32_t ssrc, uint8_t payload_type = 96, bool sendAudio = true) : RtpCache(std::move(cb)), RawEncoderImp(ssrc, payload_type,sendAudio) {};
     ~RtpCacheRaw() override = default;
 
 protected:

src/Rtp/RtpSender.cpp

@@ -19,37 +19,28 @@ using namespace toolkit;
 
 namespace mediakit{
 
-RtpSender::RtpSender(uint32_t ssrc, uint8_t payload_type,bool use_ps, bool only_audio) {
+void RtpSender::startSend(const MediaSourceEvent::SendRtpArgs &args, const function<void(uint16_t local_port, const SockException &ex)> &cb){
+    _args = args;
     _poller = EventPollerPool::Instance().getPoller();
-    if (use_ps) {
+    auto lam = [this](std::shared_ptr<List<Buffer::Ptr>> list) { onFlushRtpList(std::move(list)); };
-        _interface = std::make_shared<RtpCachePS>(
+    if (args.use_ps) {
-            [this](std::shared_ptr<List<Buffer::Ptr>> list) { onFlushRtpList(std::move(list)); }, ssrc, payload_type);
+        _interface = std::make_shared<RtpCachePS>(lam, atoi(args.ssrc.data()), args.pt);
-    }else{
+    } else {
-        _interface = std::make_shared<RtpCacheRaw>(
+        _interface = std::make_shared<RtpCacheRaw>(lam, atoi(args.ssrc.data()), args.pt, args.only_audio);
-            [this](std::shared_ptr<List<Buffer::Ptr>> list) { onFlushRtpList(std::move(list)); }, ssrc, payload_type,only_audio);
     }
-}
-
-RtpSender::~RtpSender() {}
-
-void RtpSender::startSend(const string &dst_url, uint16_t dst_port, bool is_udp, uint16_t src_port, const function<void(uint16_t local_port, const SockException &ex)> &cb){
-    _is_udp = is_udp;
     _socket = Socket::createSocket(_poller, false);
-    _dst_url = dst_url;
-    _dst_port = dst_port;
-	_src_port = src_port;
     weak_ptr<RtpSender> weak_self = shared_from_this();
-    if (is_udp) {
+    if (args.is_udp) {
-        _socket->bindUdpSock(src_port);
+        _socket->bindUdpSock(args.src_port);
         auto poller = _poller;
         auto local_port = _socket->get_local_port();
-        WorkThreadPool::Instance().getPoller()->async([cb, dst_url, dst_port, weak_self, poller, local_port]() {
+        WorkThreadPool::Instance().getPoller()->async([cb, args, weak_self, poller, local_port]() {
             struct sockaddr addr;
             //切换线程目的是为了dns解析放在后台线程执行
-            if (!SockUtil::getDomainIP(dst_url.data(), dst_port, addr)) {
+            if (!SockUtil::getDomainIP(args.dst_url.data(), args.dst_port, addr)) {
-                poller->async([dst_url, cb, local_port]() {
+                poller->async([args, cb, local_port]() {
                     //切回自己的线程
-                    cb(local_port, SockException(Err_dns, StrPrinter << "dns解析域名失败:" << dst_url));
+                    cb(local_port, SockException(Err_dns, StrPrinter << "dns解析域名失败:" << args.dst_url));
                 });
                 return;
             }
@@ -66,7 +57,7 @@ void RtpSender::startSend(const string &dst_url, uint16_t dst_port, bool is_udp,
             });
         });
     } else {
-        _socket->connect(dst_url, dst_port, [cb, weak_self](const SockException &err) {
+        _socket->connect(args.dst_url, args.dst_port, [cb, weak_self](const SockException &err) {
             auto strong_self = weak_self.lock();
             if (strong_self) {
                 if (!err) {
@@ -77,7 +68,7 @@ void RtpSender::startSend(const string &dst_url, uint16_t dst_port, bool is_udp,
             } else {
                 cb(0, err);
             }
-        }, 5.0F, "0.0.0.0", src_port);
+        }, 5.0F, "0.0.0.0", args.src_port);
     }
 }
 
@@ -85,7 +76,7 @@ void RtpSender::onConnect(){
     _is_connect = true;
     //加大发送缓存,防止udp丢包之类的问题
     SockUtil::setSendBuf(_socket->rawFD(), 4 * 1024 * 1024);
-    if (!_is_udp) {
+    if (!_args.is_udp) {
         //关闭tcp no_delay并开启MSG_MORE, 提高发送性能
         SockUtil::setNoDelay(_socket->rawFD(), false);
         _socket->setSendFlags(SOCKET_DEFAULE_FLAGS | FLAG_MORE);
@@ -99,8 +90,8 @@ void RtpSender::onConnect(){
         }
     });
     //获取本地端口，断开重连后确保端口不变
-    _src_port = _socket->get_local_port();
+    _args.src_port = _socket->get_local_port();
-    InfoL << "开始发送 rtp:" << _socket->get_peer_ip() << ":" << _socket->get_peer_port() << ", 是否为udp方式:" << _is_udp;
+    InfoL << "开始发送 rtp:" << _socket->get_peer_ip() << ":" << _socket->get_peer_port() << ", 是否为udp方式:" << _args.is_udp;
 }
 
 bool RtpSender::addTrack(const Track::Ptr &track){
@@ -128,7 +119,7 @@ void RtpSender::onFlushRtpList(shared_ptr<List<Buffer::Ptr> > rtp_list) {
         return;
     }
 
-    auto is_udp = _is_udp;
+    auto is_udp = _args.is_udp;
     auto socket = _socket;
     _poller->async([rtp_list, is_udp, socket]() {
         size_t i = 0;
@@ -150,9 +141,9 @@ void RtpSender::onErr(const SockException &ex, bool is_connect) {
 
     //监听socket断开事件，方便重连
     if (is_connect) {
-        WarnL << "重连" << _dst_url << ":" << _dst_port << "失败, 原因为:" << ex.what();
+        WarnL << "重连" << _args.dst_url << ":" << _args.dst_port << "失败, 原因为:" << ex.what();
     } else {
-        WarnL << "停止发送 rtp:" <<  _dst_url << ":" << _dst_port << ", 原因为:" << ex.what();
+        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();
@@ -161,7 +152,7 @@ void RtpSender::onErr(const SockException &ex, bool is_connect) {
         if (!strong_self) {
             return false;
         }
-        strong_self->startSend(strong_self->_dst_url, strong_self->_dst_port, strong_self->_is_udp, strong_self->_src_port, [weak_self](uint16_t local_port, const SockException &ex){
+        strong_self->startSend(strong_self->_args, [weak_self](uint16_t local_port, const SockException &ex){
             auto strong_self = weak_self.lock();
             if (strong_self && ex) {
                 //连接失败且本对象未销毁，那么重试连接

src/Rtp/RtpSender.h

@@ -21,25 +21,15 @@ class RtpSender : public MediaSinkInterface, public std::enable_shared_from_this
 public:
     typedef std::shared_ptr<RtpSender> Ptr;
 
-    ~RtpSender() override;
+    RtpSender() = default;
-
+    ~RtpSender() override = default;
-    /**
-     * 构造函数，创建GB28181 RTP发送客户端
-     * @param ssrc rtp的ssrc
-     * @param payload_type 国标中ps-rtp的pt一般为96
-     * @param use_ps 是否打包为PS然后发送
-     * @param only_audio use_ps 为false 时有效，指定发送音频还是视频
-     */
-    RtpSender(uint32_t ssrc, uint8_t payload_type = 96,bool use_ps = true,bool only_audio = true);
 
     /**
      * 开始发送ps-rtp包
-     * @param dst_url 目标ip或域名
+     * @param args 发送参数
-     * @param dst_port 目标端口
-     * @param is_udp 是否采用udp方式发送rtp
      * @param cb 连接目标端口是否成功的回调
      */
-    void startSend(const std::string &dst_url, uint16_t dst_port, bool is_udp, uint16_t src_port, const std::function<void(uint16_t local_port, const toolkit::SockException &ex)> &cb);
+    void startSend(const MediaSourceEvent::SendRtpArgs &args, const std::function<void(uint16_t local_port, const toolkit::SockException &ex)> &cb);
 
     /**
      * 输入帧数据
@@ -72,11 +62,8 @@ private:
     void onErr(const toolkit::SockException &ex, bool is_connect = false);
 
 private:
-    bool _is_udp;
     bool _is_connect = false;
-    std::string _dst_url;
+    MediaSourceEvent::SendRtpArgs _args;
-    uint16_t _dst_port;
-	uint16_t _src_port;
     toolkit::Socket::Ptr _socket;
     toolkit::EventPoller::Ptr _poller;
     toolkit::Timer::Ptr _connect_timer;