for webapi startsendrtp can send raw rtp

server/WebApi.cpp

@@ -1104,7 +1104,10 @@ void installWebApi() {
         if (!src) {
         if (!src) {
             throw ApiRetException("该媒体流不存在", API::OtherFailed);
             throw ApiRetException("该媒体流不存在", API::OtherFailed);
         }
         }
-
+        uint8_t pt = allArgs["pt"].empty() ? 96 : allArgs["pt"].as<uint8_t>();
+        bool use_ps = allArgs["use_ps"].empty() ? true : allArgs["use_ps"].as<bool>();
+        bool only_audio = allArgs["only_audio"].empty() ? true : allArgs["only_audio"].as<bool>();
+      
         //src_port为空时，则随机本地端口
         //src_port为空时，则随机本地端口
         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{
         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{
             if (ex) {
             if (ex) {
@@ -1113,7 +1116,7 @@ void installWebApi() {
             }
             }
             val["local_port"] = local_port;
             val["local_port"] = local_port;
             invoker(200, headerOut, val.toStyledString());
             invoker(200, headerOut, val.toStyledString());
-        });
+        },pt,use_ps,only_audio);
     });
     });
 
 
     api_regist("/index/api/stopSendRtp",[](API_ARGS_MAP){
     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);
     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){
+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){
     auto listener = _listener.lock();
     auto listener = _listener.lock();
     if (!listener) {
     if (!listener) {
         cb(0, SockException(Err_other, "尚未设置事件监听器"));
         cb(0, SockException(Err_other, "尚未设置事件监听器"));
         return;
         return;
     }
     }
-    return listener->startSendRtp(*this, dst_url, dst_port, ssrc, is_udp, src_port, cb);
+    return listener->startSendRtp(*this, dst_url, dst_port, ssrc, is_udp, src_port, cb, use_ps, only_audio);
 }
 }
 
 
 bool MediaSource::stopSendRtp(const string &ssrc) {
 bool MediaSource::stopSendRtp(const string &ssrc) {
@@ -720,12 +720,12 @@ vector<Track::Ptr> MediaSourceEventInterceptor::getMediaTracks(MediaSource &send
     return listener->getMediaTracks(sender, trackReady);
     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){
+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 ){
     auto listener = _listener.lock();
     auto listener = _listener.lock();
     if (listener) {
     if (listener) {
-        listener->startSendRtp(sender, dst_url, dst_port, ssrc, is_udp, src_port, cb);
+        listener->startSendRtp(sender, dst_url, dst_port, ssrc, is_udp, src_port, cb, pt, use_ps, only_audio);
     } else {
     } else {
-        MediaSourceEvent::startSendRtp(sender, dst_url, dst_port, ssrc, is_udp, src_port, cb);
+        MediaSourceEvent::startSendRtp(sender, dst_url, dst_port, ssrc, is_udp, src_port, cb, pt, use_ps, only_audio);
     }
     }
 }
 }
 
 

src/Common/MediaSource.h

@@ -86,7 +86,7 @@ public:
     // 获取所有track相关信息
     // 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>(); };
     virtual std::vector<Track::Ptr> getMediaTracks(MediaSource &sender, bool trackReady = true) const { return std::vector<Track::Ptr>(); };
     // 开始发送ps-rtp
     // 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) { cb(0, toolkit::SockException(toolkit::Err_other, "not implemented"));};
+    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"));};
     // 停止发送ps-rtp
     // 停止发送ps-rtp
     virtual bool stopSendRtp(MediaSource &sender, const std::string &ssrc) {return false; }
     virtual bool stopSendRtp(MediaSource &sender, const std::string &ssrc) {return false; }
 
 
@@ -117,7 +117,7 @@ public:
     bool setupRecord(MediaSource &sender, Recorder::type type, bool start, const std::string &custom_path, size_t max_second) override;
     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;
     bool isRecording(MediaSource &sender, Recorder::type type) override;
     std::vector<Track::Ptr> getMediaTracks(MediaSource &sender, bool trackReady = true) const 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) 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;
     bool stopSendRtp(MediaSource &sender, const std::string &ssrc) override;
     bool stopSendRtp(MediaSource &sender, const std::string &ssrc) override;
 
 
 private:
 private:
@@ -269,7 +269,7 @@ public:
     // 获取录制状态
     // 获取录制状态
     bool isRecording(Recorder::type type);
     bool isRecording(Recorder::type type);
     // 开始发送ps-rtp
     // 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);
+    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);
     // 停止发送ps-rtp
     // 停止发送ps-rtp
     bool stopSendRtp(const std::string &ssrc);
     bool stopSendRtp(const std::string &ssrc);
 
 

src/Common/MultiMediaSourceMuxer.cpp

@@ -213,9 +213,9 @@ 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){
+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){
 #if defined(ENABLE_RTPPROXY)
 #if defined(ENABLE_RTPPROXY)
-    RtpSender::Ptr rtp_sender = std::make_shared<RtpSender>(atoi(ssrc.data()));
+    RtpSender::Ptr rtp_sender = std::make_shared<RtpSender>(atoi(ssrc.data()),pt,use_ps,only_audio);
     weak_ptr<MultiMediaSourceMuxer> weak_self = shared_from_this();
     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(dst_url, dst_port, is_udp, src_port, [weak_self, rtp_sender, cb, ssrc](uint16_t local_port, const SockException &ex) {
         cb(local_port, ex);
         cb(local_port, ex);

src/Common/MultiMediaSourceMuxer.h

@@ -134,7 +134,7 @@ public:
      * @param is_udp 是否为udp
      * @param is_udp 是否为udp
      * @param cb 启动成功或失败回调
      * @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) 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;
 
 
     /**
     /**
      * 停止ps-rtp发送
      * 停止ps-rtp发送

src/Rtp/RawEncoder.cpp

+/*
+ * 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.
+ */
+
+#if defined(ENABLE_RTPPROXY)
+
+#include "RawEncoder.h"
+#include "Extension/H264Rtp.h"
+#include "Extension/AACRtp.h"
+#include "Extension/H265Rtp.h"
+#include "Extension/CommonRtp.h"
+#include "Extension/G711Rtp.h"
+#include "Rtsp/RtspMuxer.h"
+
+using namespace toolkit;
+
+namespace mediakit{
+
+RawEncoderImp::RawEncoderImp(uint32_t ssrc, uint8_t payload_type,bool sendAudio):_ssrc(ssrc),_payload_type(payload_type),_sendAudio(sendAudio) {
+
+}
+
+RawEncoderImp::~RawEncoderImp() {
+    InfoL << this << " " << printSSRC(_ssrc);
+}
+
+bool RawEncoderImp::addTrack(const Track::Ptr &track){
+    if(_sendAudio && track->getTrackType() == TrackType::TrackAudio && !_rtp_encoder){// audio
+        _rtp_encoder = createRtpEncoder(track);
+        _rtp_encoder->setRtpRing(std::make_shared<RtpRing::RingType>());
+        _rtp_encoder->getRtpRing()->setDelegate(std::make_shared<RingDelegateHelper>([this](RtpPacket::Ptr rtp, bool is_key){
+            onRTP(std::move(rtp));
+         }));
+        return true;
+    }
+
+    if(!_sendAudio && track->getTrackType()==TrackType::TrackVideo && !_rtp_encoder){
+         _rtp_encoder = createRtpEncoder(track);
+         _rtp_encoder->setRtpRing(std::make_shared<RtpRing::RingType>());
+        _rtp_encoder->getRtpRing()->setDelegate(std::make_shared<RingDelegateHelper>([this](RtpPacket::Ptr rtp, bool is_key){
+            onRTP(std::move(rtp));
+         }));
+        return true;
+    }
+    return true;
+}
+
+
+void RawEncoderImp::resetTracks(){
+    return;
+}
+
+  
+bool RawEncoderImp::inputFrame(const Frame::Ptr &frame){
+    if(frame->getTrackType() == TrackType::TrackAudio && _sendAudio && _rtp_encoder){
+        _rtp_encoder->inputFrame(frame);
+    }
+
+    if(frame->getTrackType() == TrackType::TrackVideo && !_sendAudio && _rtp_encoder){
+         _rtp_encoder->inputFrame(frame);
+    }
+    return true;
+}
+
+RtpCodec::Ptr RawEncoderImp::createRtpEncoder(const Track::Ptr &track){
+    GET_CONFIG(uint32_t,audio_mtu,Rtp::kAudioMtuSize);
+    GET_CONFIG(uint32_t,video_mtu,Rtp::kVideoMtuSize);
+    auto codec_id = track->getCodecId();
+    uint32_t sample_rate = 90000;
+    int channels = 1;
+    auto mtu = (track->getTrackType() == TrackVideo ? video_mtu : audio_mtu);
+    if(track->getTrackType() == TrackType::TrackAudio){
+        AudioTrack::Ptr audioTrack = std::dynamic_pointer_cast<AudioTrack>(track);
+        sample_rate = audioTrack->getAudioSampleRate();
+        channels = audioTrack->getAudioChannel();
+    }
+    switch (codec_id){
+        case CodecH264 : return std::make_shared<H264RtpEncoder>(_ssrc, mtu, sample_rate, _payload_type, 0);
+        case CodecH265 : return std::make_shared<H265RtpEncoder>(_ssrc, mtu, sample_rate, _payload_type, 0);
+        case CodecAAC : return std::make_shared<AACRtpEncoder>(_ssrc, mtu, sample_rate, _payload_type, 0);
+        case CodecL16 :
+        case CodecOpus : return std::make_shared<CommonRtpEncoder>(codec_id, _ssrc, mtu, sample_rate, _payload_type, 0);
+        case CodecG711A :
+        case CodecG711U : {
+            if (_payload_type == Rtsp::PT_PCMA || _payload_type == Rtsp::PT_PCMU) {
+                return std::make_shared<G711RtpEncoder>(codec_id, _ssrc, mtu, sample_rate, _payload_type, 0, channels);
+            }
+            return std::make_shared<CommonRtpEncoder>(codec_id, _ssrc, mtu, sample_rate, _payload_type, 0);
+        }
+        default : WarnL << "暂不支持该CodecId:" << codec_id; return nullptr;
+    }
+}
+
+}//namespace mediakit
+
+#endif//defined(ENABLE_RTPPROXY)

src/Rtp/RawEncoder.h

+/*
+ * 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.
+ */
+
+#ifndef ZLMEDIAKIT_RAWENCODER_H
+#define ZLMEDIAKIT_RAWENCODER_H
+
+#if defined(ENABLE_RTPPROXY)
+
+
+#include "Common/MediaSink.h"
+#include "Common/Stamp.h"
+#include "Extension/CommonRtp.h"
+
+namespace mediakit{
+
+class RawEncoderImp : public MediaSinkInterface{
+public:
+    RawEncoderImp(uint32_t ssrc, uint8_t payload_type = 96, bool sendAudio = true);
+    ~RawEncoderImp() override;
+
+    /**
+     * 添加音视频轨道
+     */
+    bool addTrack(const Track::Ptr &track) override;
+
+
+    /**
+     * 重置音视频轨道
+     */
+    void resetTracks() override;
+
+    /**
+     * 输入帧数据
+     */
+    bool inputFrame(const Frame::Ptr &frame) override;
+
+protected:
+    //rtp打包后回调
+    virtual void onRTP(toolkit::Buffer::Ptr rtp) = 0;
+private:
+    RtpCodec::Ptr createRtpEncoder(const Track::Ptr &track);
+
+    uint32_t _ssrc;
+    uint8_t _payload_type;
+    bool _sendAudio;
+    RtpCodec::Ptr _rtp_encoder;
+};
+
+}//namespace mediakit
+
+#endif //ENABLE_RTPPROXY
+#endif //ZLMEDIAKIT_RAWENCODER_H

src/Rtp/RtpCache.cpp

@@ -34,6 +34,12 @@ void RtpCachePS::onRTP(Buffer::Ptr buffer) {
     input(stamp, std::move(buffer));
     input(stamp, std::move(buffer));
 }
 }
 
 
+void RtpCacheRaw::onRTP(Buffer::Ptr buffer) {
+    auto rtp = std::static_pointer_cast<RtpPacket>(buffer);
+    auto stamp = rtp->getStampMS();
+    input(stamp, std::move(buffer));
+}
+
 }//namespace mediakit
 }//namespace mediakit
 
 
 #endif//#if defined(ENABLE_RTPPROXY)
 #endif//#if defined(ENABLE_RTPPROXY)
\ No newline at end of file

src/Rtp/RtpCache.h

@@ -14,6 +14,7 @@
 #if defined(ENABLE_RTPPROXY)
 #if defined(ENABLE_RTPPROXY)
 
 
 #include "PSEncoder.h"
 #include "PSEncoder.h"
+#include "RawEncoder.h"
 #include "Extension/CommonRtp.h"
 #include "Extension/CommonRtp.h"
 
 
 namespace mediakit{
 namespace mediakit{
@@ -47,6 +48,16 @@ protected:
     void onRTP(toolkit::Buffer::Ptr rtp)  override;
     void onRTP(toolkit::Buffer::Ptr rtp)  override;
 };
 };
 
 
+
+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() override = default;
+
+protected:
+    void onRTP(toolkit::Buffer::Ptr rtp)  override;
+};
+
 }//namespace mediakit
 }//namespace mediakit
 #endif//ENABLE_RTPPROXY
 #endif//ENABLE_RTPPROXY
 #endif //ZLMEDIAKIT_RTPCACHE_H
 #endif //ZLMEDIAKIT_RTPCACHE_H

src/Rtp/RtpSender.cpp

@@ -19,11 +19,15 @@ using namespace toolkit;
 
 
 namespace mediakit{
 namespace mediakit{
 
 
-RtpSender::RtpSender(uint32_t ssrc, uint8_t payload_type) {
+RtpSender::RtpSender(uint32_t ssrc, uint8_t payload_type,bool use_ps, bool only_audio) {
     _poller = EventPollerPool::Instance().getPoller();
     _poller = EventPollerPool::Instance().getPoller();
-    _interface = std::make_shared<RtpCachePS>([this](std::shared_ptr<List<Buffer::Ptr> > list) {
-        onFlushRtpList(std::move(list));
-    }, ssrc, payload_type);
+    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);
+    }
 }
 }
 
 
 RtpSender::~RtpSender() {}
 RtpSender::~RtpSender() {}

src/Rtp/RtpSender.h

@@ -27,8 +27,10 @@ public:
      * 构造函数，创建GB28181 RTP发送客户端
      * 构造函数，创建GB28181 RTP发送客户端
      * @param ssrc rtp的ssrc
      * @param ssrc rtp的ssrc
      * @param payload_type 国标中ps-rtp的pt一般为96
      * @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);
+    RtpSender(uint32_t ssrc, uint8_t payload_type = 96,bool use_ps = true,bool only_audio = true);
 
 
     /**
     /**
      * 开始发送ps-rtp包
      * 开始发送ps-rtp包