完善startSendRtp接口

2026-07-01 07:22:23 +08:00 · 2022-04-03 18:25:36 +08:00
parent de0738b1d1
commit 2818e371b8
11 changed files with 111 additions and 73 deletions
--- a/src/Common/MediaSource.cpp
+++ b/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);
    }
 }

--- a/src/Common/MediaSource.h
+++ b/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);

--- a/src/Common/MultiMediaSourceMuxer.cpp
+++ b/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"));
--- a/src/Common/MultiMediaSourceMuxer.h
+++ b/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
--- a/src/Rtp/RtpCache.h
+++ b/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:
--- a/src/Rtp/RtpSender.cpp
+++ b/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) {
-        _interface = std::make_shared<RtpCachePS>(
-            [this](std::shared_ptr<List<Buffer::Ptr>> list) { onFlushRtpList(std::move(list)); }, ssrc, payload_type);
-    }else{
-        _interface = std::make_shared<RtpCacheRaw>(
-            [this](std::shared_ptr<List<Buffer::Ptr>> list) { onFlushRtpList(std::move(list)); }, ssrc, payload_type,only_audio);
+    auto lam = [this](std::shared_ptr<List<Buffer::Ptr>> list) { onFlushRtpList(std::move(list)); };
+    if (args.use_ps) {
+        _interface = std::make_shared<RtpCachePS>(lam, atoi(args.ssrc.data()), args.pt);
+    } else {
+        _interface = std::make_shared<RtpCacheRaw>(lam, atoi(args.ssrc.data()), args.pt, args.only_audio);
    }
-}
-
-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) {
-        _socket->bindUdpSock(src_port);
+    if (args.is_udp) {
+        _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)) {
-                poller->async([dst_url, cb, local_port]() {
+            if (!SockUtil::getDomainIP(args.dst_url.data(), args.dst_port, addr)) {
+                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();
-    InfoL << "开始发送 rtp:" << _socket->get_peer_ip() << ":" << _socket->get_peer_port() << ", 是否为udp方式:" << _is_udp;
+    _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:" << _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
--- a/src/Rtp/RtpSender.h
+++ b/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;
-
-    /**
-     * 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);
+    RtpSender() = default;
+    ~RtpSender() override = default;

    /**
     * 开始发送ps-rtp包
-     * @param dst_url 目标ip或域名
-     * @param dst_port 目标端口
-     * @param is_udp 是否采用udp方式发送rtp
+     * @param args 发送参数
     * @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;
-    uint16_t _dst_port;
-	uint16_t _src_port;
+    MediaSourceEvent::SendRtpArgs _args;
    toolkit::Socket::Ptr _socket;
    toolkit::EventPoller::Ptr _poller;
    toolkit::Timer::Ptr _connect_timer;