整理webrtc相关代码命名空间

0b355759 · ziyue · 15affeff · 0b355759 · 0b355759 · 0b355759
Commit 0b355759 authored Sep 18, 2022 by ziyue
--- a/api/source/mk_webrtc_private.h
+++ b/api/source/mk_webrtc_private.h
@@ -24,6 +24,8 @@
 #include "../webrtc/WebRtcPusher.h"
 #include "../webrtc/WebRtcTransport.h"
+using namespace mediakit;
 namespace API {
 typedef enum {
    NotFound = -500, //未找到

--- a/server/main.cpp
+++ b/server/main.cpp
@@ -286,7 +286,7 @@ int start_main(int argc,char *argv[]) {
            }
            return Socket::createSocket(new_poller, false);
        });
-        uint16_t rtcPort = mINI::Instance()[RTC::kPort];
+        uint16_t rtcPort = mINI::Instance()[Rtc::kPort];
 #endif//defined(ENABLE_WEBRTC)

--- a/tests/test_server.cpp
+++ b/tests/test_server.cpp
@@ -83,7 +83,7 @@ onceToken token1([](){
 }  // namespace mediakit
-#define REALM "realm_zlmedaikit"
+#define REALM "realm_zlmediakit"
 static map<string,FlvRecorder::Ptr> s_mapFlvRecorder;
 static mutex s_mtxFlvRecorder;

--- a/webrtc/Nack.cpp
+++ b/webrtc/Nack.cpp
@@ -12,7 +12,8 @@
 using namespace std;
 using namespace toolkit;
-using namespace mediakit;
+namespace mediakit {
 static constexpr uint32_t kMaxNackMS = 5 * 1000;
 static constexpr uint32_t kRtpCacheCheckInterval = 100;
@@ -92,7 +93,6 @@ int64_t NackList::getRtpStamp(uint16_t seq) {
    return it->second->getStampMS(false);
 }
 ////////////////////////////////////////////////////////////////////////////////////////////////
 void NackContext::received(uint16_t seq, bool is_rtx) {
@@ -101,7 +101,7 @@ void NackContext::received(uint16_t seq, bool is_rtx) {
    }
    if (is_rtx || (seq < _last_max_seq && !(seq < 1024 && _last_max_seq > UINT16_MAX - 1024))) {
        //重传包或
-        //seq回退，且非回环，那么这个应该是重传包
+        // seq回退，且非回环，那么这个应该是重传包
        onRtx(seq);
        return;
    }
@@ -127,7 +127,7 @@ void NackContext::received(uint16_t seq, bool is_rtx) {
        _seq.clear();
        _last_max_seq = max_seq;
    } else {
-        //seq不连续，有丢包
+        // seq不连续，有丢包
        if (min_seq == _last_max_seq + 1) {
            //前面部分seq是连续的，未丢包，移除之
            eraseFrontSeq();
@@ -135,7 +135,7 @@ void NackContext::received(uint16_t seq, bool is_rtx) {
        //有丢包，丢包从_last_max_seq开始
        auto nack_rtp_count = FCI_NACK::kBitSize;
-        if (max_seq  > nack_rtp_count + _last_max_seq) {
+        if (max_seq > nack_rtp_count + _last_max_seq) {
            vector<bool> vec;
            vec.resize(FCI_NACK::kBitSize, false);
            for (size_t i = 0; i < nack_rtp_count; ++i) {
@@ -170,7 +170,7 @@ void NackContext::eraseFrontSeq() {
    //前面部分seq是连续的，未丢包，移除之
    for (auto it = _seq.begin(); it != _seq.end();) {
        if (*it != _last_max_seq + 1) {
-            //seq不连续，丢包了
+            // seq不连续，丢包了
            break;
        }
        _last_max_seq = *it;
@@ -187,9 +187,9 @@ void NackContext::onRtx(uint16_t seq) {
    _nack_send_status.erase(it);
    if (rtt >= 0) {
-        //rtt不肯小于0
+        // rtt不肯小于0
        _rtt = rtt;
-        //InfoL << "rtt:" << rtt;
+        // InfoL << "rtt:" << rtt;
    }
 }
@@ -230,7 +230,7 @@ uint64_t NackContext::reSendNack() {
        //更新nack发送时间戳
        it->second.update_stamp = now;
        if (++(it->second.nack_count) == kNackMaxCount) {
-            //nack次数太多，移除之
+            // nack次数太多，移除之
            it = _nack_send_status.erase(it);
            continue;
        }
@@ -269,3 +269,5 @@ uint64_t NackContext::reSendNack() {
    //重传间隔不得低于5ms
    return max(_rtt, 5);
 }
+} // namespace mediakit
--- a/webrtc/Nack.h
+++ b/webrtc/Nack.h
@@ -14,38 +14,39 @@
 #include "Rtsp/Rtsp.h"
 #include "Rtcp/RtcpFCI.h"
+namespace mediakit {
 class NackList {
 public:
    NackList() = default;
    ~NackList() = default;
-    void pushBack(mediakit::RtpPacket::Ptr rtp);
+    void pushBack(RtpPacket::Ptr rtp);
-    void forEach(const mediakit::FCI_NACK &nack, const std::function<void(const mediakit::RtpPacket::Ptr &rtp)> &cb);
+    void forEach(const FCI_NACK &nack, const std::function<void(const RtpPacket::Ptr &rtp)> &cb);
 private:
    void popFront();
    uint32_t getCacheMS();
    int64_t getRtpStamp(uint16_t seq);
-    mediakit::RtpPacket::Ptr *getRtp(uint16_t seq);
+    RtpPacket::Ptr *getRtp(uint16_t seq);
 private:
    uint32_t _cache_ms_check = 0;
    std::deque<uint16_t> _nack_cache_seq;
-    std::unordered_map<uint16_t, mediakit::RtpPacket::Ptr> _nack_cache_pkt;
+    std::unordered_map<uint16_t, RtpPacket::Ptr> _nack_cache_pkt;
 };
 class NackContext {
 public:
    using Ptr = std::shared_ptr<NackContext>;
-    using onNack = std::function<void(const mediakit::FCI_NACK &nack)>;
+    using onNack = std::function<void(const FCI_NACK &nack)>;
    //最大保留的rtp丢包状态个数
    static constexpr auto kNackMaxSize = 2048;
-    //rtp丢包状态最长保留时间
+    // rtp丢包状态最长保留时间
    static constexpr auto kNackMaxMS = 3 * 1000;
-    //nack最多请求重传10次
+    // nack最多请求重传10次
    static constexpr auto kNackMaxCount = 10;
-    //nack重传频率，rtt的倍数
+    // nack重传频率，rtt的倍数
    static constexpr auto kNackIntervalRatio = 1.0f;
    NackContext() = default;
@@ -57,8 +58,8 @@ public:
 private:
    void eraseFrontSeq();
-    void doNack(const mediakit::FCI_NACK &nack, bool record_nack);
+    void doNack(const FCI_NACK &nack, bool record_nack);
-    void recordNack(const mediakit::FCI_NACK &nack);
+    void recordNack(const FCI_NACK &nack);
    void onRtx(uint16_t seq);
 private:
@@ -67,12 +68,14 @@ private:
    std::set<uint16_t> _seq;
    uint16_t _last_max_seq = 0;
-    struct NackStatus{
+    struct NackStatus {
        uint64_t first_stamp;
        uint64_t update_stamp;
        int nack_count = 0;
    };
-    std::map<uint16_t/*seq*/, NackStatus > _nack_send_status;
+    std::map<uint16_t /*seq*/, NackStatus> _nack_send_status;
 };
+} // namespace mediakit
 #endif //ZLMEDIAKIT_NACK_H
--- a/webrtc/RtpExt.cpp
+++ b/webrtc/RtpExt.cpp
@@ -17,7 +17,8 @@
 using namespace std;
 using namespace toolkit;
-using namespace mediakit;
+namespace mediakit {
 //https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01
 //https://tools.ietf.org/html/rfc5285
@@ -644,3 +645,4 @@ void RtpExtContext::onGetRtp(uint8_t pt, uint32_t ssrc, const string &rid){
    }
 }
+}// namespace mediakit
\ No newline at end of file
--- a/webrtc/RtpExt.h
+++ b/webrtc/RtpExt.h
@@ -17,6 +17,7 @@
 #include "Common/macros.h"
 #include "Rtsp/Rtsp.h"
+namespace mediakit {
 #define RTP_EXT_MAP(XX) \
    XX(ssrc_audio_level,            "urn:ietf:params:rtp-hdrext:ssrc-audio-level") \
@@ -53,7 +54,7 @@ public:
    friend class RtpExtContext;
    ~RtpExt() = default;
-    static std::map<uint8_t/*id*/, RtpExt/*data*/> getExtValue(const mediakit::RtpHeader *header);
+    static std::map<uint8_t/*id*/, RtpExt/*data*/> getExtValue(const RtpHeader *header);
    static RtpExtType getExtType(const std::string &url);
    static const std::string& getExtUrl(RtpExtType type);
    static const char *getExtName(RtpExtType type);
@@ -122,7 +123,7 @@ public:
    void setOnGetRtp(OnGetRtp cb);
    std::string getRid(uint32_t ssrc) const;
    void setRid(uint32_t ssrc, const std::string &rid);
-    RtpExt changeRtpExtId(const mediakit::RtpHeader *header, bool is_recv, std::string *rid_ptr = nullptr, RtpExtType type = RtpExtType::padding);
+    RtpExt changeRtpExtId(const RtpHeader *header, bool is_recv, std::string *rid_ptr = nullptr, RtpExtType type = RtpExtType::padding);
 private:
    void onGetRtp(uint8_t pt, uint32_t ssrc, const std::string &rid);
@@ -137,4 +138,5 @@ private:
    std::unordered_map<uint32_t/*simulcast ssrc*/, std::string/*rid*/> _ssrc_to_rid;
 };
+} //namespace mediakit
 #endif //ZLMEDIAKIT_RTPEXT_H
--- a/webrtc/Sdp.cpp
+++ b/webrtc/Sdp.cpp
@@ -14,9 +14,10 @@
 using namespace std;
 using namespace toolkit;
-using namespace mediakit;
-namespace RTC {
+namespace mediakit {
+namespace Rtc {
 #define RTC_FIELD "rtc."
 const string kPreferredCodecA = RTC_FIELD"preferredCodecA";
 const string kPreferredCodecV = RTC_FIELD"preferredCodecV";
@@ -1404,7 +1405,7 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){
    switch (type) {
        case TrackAudio: {
            //此处调整偏好的编码格式优先级
-            GET_CONFIG_FUNC(vector<CodecId>, s_preferred_codec, RTC::kPreferredCodecA, toCodecArray);
+            GET_CONFIG_FUNC(vector<CodecId>, s_preferred_codec, Rtc::kPreferredCodecA, toCodecArray);
            CHECK(!s_preferred_codec.empty(), "rtc音频偏好codec不能为空");
            preferred_codec = s_preferred_codec;
@@ -1423,7 +1424,7 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){
        }
        case TrackVideo: {
            //此处调整偏好的编码格式优先级
-            GET_CONFIG_FUNC(vector<CodecId>, s_preferred_codec, RTC::kPreferredCodecV, toCodecArray);
+            GET_CONFIG_FUNC(vector<CodecId>, s_preferred_codec, Rtc::kPreferredCodecV, toCodecArray);
            CHECK(!s_preferred_codec.empty(), "rtc视频偏好codec不能为空");
            preferred_codec = s_preferred_codec;
@@ -1811,3 +1812,5 @@ void RtcConfigure::onSelectPlan(RtcCodecPlan &plan, CodecId codec) const {
        plan.fmtp[kMode] = mode.empty() ? "0" : mode;
    }
 }
+} // namespace mediakit
\ No newline at end of file
--- a/webrtc/Sdp.h
+++ b/webrtc/Sdp.h
@@ -18,6 +18,8 @@
 #include "Extension/Frame.h"
 #include "Common/Parser.h"
+namespace mediakit {
 //https://datatracker.ietf.org/doc/rfc4566/?include_text=1
 //https://blog.csdn.net/aggresss/article/details/109850434
 //https://aggresss.blog.csdn.net/article/details/106436703
@@ -189,7 +191,7 @@ class SdpMedia : public SdpItem {
 public:
    // 5.14.  Media Descriptions ("m=")
    // m=<media> <port> <proto> <fmt> ...
-    mediakit::TrackType type;
+    TrackType type;
    uint16_t port;
    //RTP/AVP：应用场景为视频/音频的 RTP 协议。参考 RFC 3551
    //RTP/SAVP：应用场景为视频/音频的 SRTP 协议。参考 RFC 3711
@@ -374,7 +376,7 @@ class SdpAttrFmtp : public SdpItem {
 public:
    //fmtp:96 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f
    uint8_t pt;
-    std::map<std::string/*key*/, std::string/*value*/, mediakit::StrCaseCompare> fmtp;
+    std::map<std::string/*key*/, std::string/*value*/, StrCaseCompare> fmtp;
    void parse(const std::string &str) override;
    std::string toString() const override;
    const char* getKey() const override { return "fmtp";}
@@ -600,7 +602,7 @@ public:
    uint32_t channel = 0;
    //rtcp反馈
    std::set<std::string> rtcp_fb;
-    std::map<std::string/*key*/, std::string/*value*/, mediakit::StrCaseCompare> fmtp;
+    std::map<std::string/*key*/, std::string/*value*/, StrCaseCompare> fmtp;
    std::string getFmtp(const char *key) const;
 };
@@ -608,7 +610,7 @@ public:
 //rtc 媒体描述
 class RtcMedia{
 public:
-    mediakit::TrackType type{mediakit::TrackType::TrackInvalid};
+    TrackType type{TrackType::TrackInvalid};
    std::string mid;
    uint16_t port{0};
    SdpConnection addr;
@@ -675,8 +677,8 @@ public:
    void checkValid() const;
    std::string toString() const;
    std::string toRtspSdp() const;
-    const  RtcMedia *getMedia(mediakit::TrackType type) const;
+    const  RtcMedia *getMedia(TrackType type) const;
-    bool supportRtcpFb(const std::string &name, mediakit::TrackType type = mediakit::TrackType::TrackVideo) const;
+    bool supportRtcpFb(const std::string &name, TrackType type = TrackType::TrackVideo) const;
    bool supportSimulcast() const;
    bool isOnlyDatachannel() const;
@@ -706,10 +708,10 @@ public:
        std::set<std::string> rtcp_fb;
        std::map<RtpExtType, RtpDirection> extmap;
-        std::vector<mediakit::CodecId> preferred_codec;
+        std::vector<CodecId> preferred_codec;
        std::vector<SdpAttrCandidate> candidate;
-        void setDefaultSetting(mediakit::TrackType type);
+        void setDefaultSetting(TrackType type);
        void enableTWCC(bool enable = true);
        void enableREMB(bool enable = true);
    };
@@ -719,19 +721,19 @@ public:
    RtcTrackConfigure application;
    void setDefaultSetting(std::string ice_ufrag, std::string ice_pwd, RtpDirection direction, const SdpAttrFingerprint &fingerprint);
-    void addCandidate(const SdpAttrCandidate &candidate, mediakit::TrackType type = mediakit::TrackInvalid);
+    void addCandidate(const SdpAttrCandidate &candidate, TrackType type = TrackInvalid);
    std::shared_ptr<RtcSession> createAnswer(const RtcSession &offer) const;
    void setPlayRtspInfo(const std::string &sdp);
-    void enableTWCC(bool enable = true, mediakit::TrackType type = mediakit::TrackInvalid);
+    void enableTWCC(bool enable = true, TrackType type = TrackInvalid);
-    void enableREMB(bool enable = true, mediakit::TrackType type = mediakit::TrackInvalid);
+    void enableREMB(bool enable = true, TrackType type = TrackInvalid);
 private:
    void matchMedia(const std::shared_ptr<RtcSession> &ret, const RtcMedia &media) const;
-    bool onCheckCodecProfile(const RtcCodecPlan &plan, mediakit::CodecId codec) const;
+    bool onCheckCodecProfile(const RtcCodecPlan &plan, CodecId codec) const;
-    void onSelectPlan(RtcCodecPlan &plan, mediakit::CodecId codec) const;
+    void onSelectPlan(RtcCodecPlan &plan, CodecId codec) const;
 private:
    RtcCodecPlan::Ptr _rtsp_video_plan;
@@ -748,5 +750,6 @@ private:
    ~SdpConst() = delete;
 };
+}// namespace mediakit
 #endif //ZLMEDIAKIT_SDP_H
--- a/webrtc/TwccContext.cpp
+++ b/webrtc/TwccContext.cpp
@@ -11,7 +11,7 @@
 #include "TwccContext.h"
 #include "Rtcp/RtcpFCI.h"
-using namespace mediakit;
+namespace mediakit {
 enum class ExtSeqStatus : int {
    normal = 0,
@@ -121,3 +121,5 @@ void TwccContext::clearStatus() {
 void TwccContext::setOnSendTwccCB(TwccContext::onSendTwccCB cb) {
    _cb = std::move(cb);
 }
+}// namespace mediakit
\ No newline at end of file
--- a/webrtc/TwccContext.h
+++ b/webrtc/TwccContext.h
@@ -16,6 +16,8 @@
 #include <functional>
 #include "Util/TimeTicker.h"
+namespace mediakit {
 class TwccContext {
 public:
    using onSendTwccCB = std::function<void(uint32_t ssrc, std::string fci)>;
@@ -44,5 +46,5 @@ private:
    onSendTwccCB _cb;
 };
+}// namespace mediakit
 #endif //ZLMEDIAKIT_TWCCCONTEXT_H
--- a/webrtc/WebRtcEchoTest.cpp
+++ b/webrtc/WebRtcEchoTest.cpp
@@ -10,6 +10,8 @@
 #include "WebRtcEchoTest.h"
+namespace mediakit {
 WebRtcEchoTest::Ptr WebRtcEchoTest::create(const EventPoller::Ptr &poller) {
    WebRtcEchoTest::Ptr ret(new WebRtcEchoTest(poller), [](WebRtcEchoTest *ptr) {
        ptr->onDestory();
@@ -48,4 +50,6 @@ void WebRtcEchoTest::onCheckSdp(SdpType type, RtcSession &sdp) {
            }
        }
    }
 }
\ No newline at end of file
+}// namespace mediakit
\ No newline at end of file
--- a/webrtc/WebRtcEchoTest.h
+++ b/webrtc/WebRtcEchoTest.h
@@ -13,6 +13,8 @@
 #include "WebRtcTransport.h"
+namespace mediakit {
 class WebRtcEchoTest : public WebRtcTransportImp {
 public:
    using Ptr = std::shared_ptr<WebRtcEchoTest>;
@@ -26,7 +28,7 @@ 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, mediakit::RtpPacket::Ptr rtp) 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 {};
@@ -34,4 +36,5 @@ private:
    WebRtcEchoTest(const EventPoller::Ptr &poller);
 };
+}// namespace mediakit
 #endif //ZLMEDIAKIT_WEBRTCECHOTEST_H
--- a/webrtc/WebRtcPlayer.cpp
+++ b/webrtc/WebRtcPlayer.cpp
@@ -11,7 +11,8 @@
 #include "WebRtcPlayer.h"
 using namespace std;
-using namespace mediakit;
+namespace mediakit {
 WebRtcPlayer::Ptr WebRtcPlayer::create(const EventPoller::Ptr &poller,
                                       const RtspMediaSource::Ptr &src,
@@ -88,3 +89,5 @@ void WebRtcPlayer::onRtcConfigure(RtcConfigure &configure) const {
    configure.audio.direction = configure.video.direction = RtpDirection::sendonly;
    configure.setPlayRtspInfo(_play_src->getSdp());
 }
+}// namespace mediakit
\ No newline at end of file
--- a/webrtc/WebRtcPlayer.h
+++ b/webrtc/WebRtcPlayer.h
@@ -13,30 +13,32 @@
 #include "WebRtcTransport.h"
+namespace mediakit {
 class WebRtcPlayer : public WebRtcTransportImp {
 public:
    using Ptr = std::shared_ptr<WebRtcPlayer>;
    ~WebRtcPlayer() override = default;
-    static Ptr create(const EventPoller::Ptr &poller, const mediakit::RtspMediaSource::Ptr &src, const mediakit::MediaInfo &info);
+    static Ptr create(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src, const MediaInfo &info);
 protected:
    ///////WebRtcTransportImp override///////
    void onStartWebRTC() override;
    void onDestory() override;
    void onRtcConfigure(RtcConfigure &configure) const override;
-    void onRecvRtp(MediaTrack &track, const std::string &rid, mediakit::RtpPacket::Ptr rtp) override {};
+    void onRecvRtp(MediaTrack &track, const std::string &rid, RtpPacket::Ptr rtp) override {};
 private:
-    WebRtcPlayer(const EventPoller::Ptr &poller, const mediakit::RtspMediaSource::Ptr &src, const mediakit::MediaInfo &info);
+    WebRtcPlayer(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src, const MediaInfo &info);
 private:
    //媒体相关元数据
-    mediakit::MediaInfo _media_info;
+    MediaInfo _media_info;
    //播放的rtsp源
-    mediakit::RtspMediaSource::Ptr _play_src;
+    RtspMediaSource::Ptr _play_src;
    //播放rtsp源的reader对象
-    mediakit::RtspMediaSource::RingType::RingReader::Ptr _reader;
+    RtspMediaSource::RingType::RingReader::Ptr _reader;
 };
+}// namespace mediakit
-#endif //ZLMEDIAKIT_WEBRTCPLAYER_H
+#endif // ZLMEDIAKIT_WEBRTCPLAYER_H
--- a/webrtc/WebRtcPusher.cpp
+++ b/webrtc/WebRtcPusher.cpp
@@ -11,13 +11,14 @@
 #include "WebRtcPusher.h"
 using namespace std;
-using namespace mediakit;
+namespace mediakit {
 WebRtcPusher::Ptr WebRtcPusher::create(const EventPoller::Ptr &poller,
                                       const RtspMediaSourceImp::Ptr &src,
                                       const std::shared_ptr<void> &ownership,
                                       const MediaInfo &info,
-                                       const mediakit::ProtocolOption &option) {
+                                       const ProtocolOption &option) {
    WebRtcPusher::Ptr ret(new WebRtcPusher(poller, src, ownership, info, option), [](WebRtcPusher *ptr) {
        ptr->onDestory();
        delete ptr;
@@ -30,7 +31,7 @@ WebRtcPusher::WebRtcPusher(const EventPoller::Ptr &poller,
                           const RtspMediaSourceImp::Ptr &src,
                           const std::shared_ptr<void> &ownership,
                           const MediaInfo &info,
-                           const mediakit::ProtocolOption &option) : WebRtcTransportImp(poller) {
+                           const ProtocolOption &option) : WebRtcTransportImp(poller) {
    _media_info = info;
    _push_src = src;
    _push_src_ownership = ownership;
@@ -142,7 +143,7 @@ void WebRtcPusher::onRtcConfigure(RtcConfigure &configure) const {
    configure.audio.direction = configure.video.direction = RtpDirection::recvonly;
 }
-float WebRtcPusher::getLossRate(MediaSource &sender,mediakit::TrackType type){
+float WebRtcPusher::getLossRate(MediaSource &sender,TrackType type){
    return WebRtcTransportImp::getLossRate(type);
 }
@@ -155,4 +156,6 @@ void WebRtcPusher::OnDtlsTransportClosed(const RTC::DtlsTransport *dtlsTransport
 void WebRtcPusher::onRtcpBye(){
    _push_src = nullptr;
     WebRtcTransportImp::onRtcpBye();
 }
\ No newline at end of file
+}// namespace mediakit
\ No newline at end of file
--- a/webrtc/WebRtcPusher.h
+++ b/webrtc/WebRtcPusher.h
@@ -13,19 +13,21 @@
 #include "WebRtcTransport.h"
-class WebRtcPusher : public WebRtcTransportImp, public mediakit::MediaSourceEvent {
+namespace mediakit {
+class WebRtcPusher : public WebRtcTransportImp, public MediaSourceEvent {
 public:
    using Ptr = std::shared_ptr<WebRtcPusher>;
    ~WebRtcPusher() override = default;
-    static Ptr create(const EventPoller::Ptr &poller, const mediakit::RtspMediaSourceImp::Ptr &src,
+    static Ptr create(const EventPoller::Ptr &poller, const RtspMediaSourceImp::Ptr &src,
-                      const std::shared_ptr<void> &ownership, const mediakit::MediaInfo &info, const mediakit::ProtocolOption &option);
+                      const std::shared_ptr<void> &ownership, const MediaInfo &info, const ProtocolOption &option);
 protected:
    ///////WebRtcTransportImp override///////
    void onStartWebRTC() override;
    void onDestory() override;
    void onRtcConfigure(RtcConfigure &configure) const override;
-    void onRecvRtp(MediaTrack &track, const std::string &rid, mediakit::RtpPacket::Ptr rtp) override;
+    void onRecvRtp(MediaTrack &track, const std::string &rid, RtpPacket::Ptr rtp) override;
    void onRtcpBye() override;
    ////  dtls相关的回调 ////
   void OnDtlsTransportClosed(const RTC::DtlsTransport *dtlsTransport) override;
@@ -33,35 +35,36 @@ protected:
 protected:
    ///////MediaSourceEvent override///////
    // 关闭
-    bool close(mediakit::MediaSource &sender) override;
+    bool close(MediaSource &sender) override;
    // 播放总人数
-    int totalReaderCount(mediakit::MediaSource &sender) override;
+    int totalReaderCount(MediaSource &sender) override;
    // 获取媒体源类型
-    mediakit::MediaOriginType getOriginType(mediakit::MediaSource &sender) const override;
+    MediaOriginType getOriginType(MediaSource &sender) const override;
    // 获取媒体源url或者文件路径
-    std::string getOriginUrl(mediakit::MediaSource &sender) const override;
+    std::string getOriginUrl(MediaSource &sender) const override;
    // 获取媒体源客户端相关信息
-    std::shared_ptr<SockInfo> getOriginSock(mediakit::MediaSource &sender) const override;
+    std::shared_ptr<SockInfo> getOriginSock(MediaSource &sender) const override;
    // 获取丢包率
-    float getLossRate(mediakit::MediaSource &sender,mediakit::TrackType type) override;
+    float getLossRate(MediaSource &sender,TrackType type) override;
 private:
-    WebRtcPusher(const EventPoller::Ptr &poller, const mediakit::RtspMediaSourceImp::Ptr &src,
+    WebRtcPusher(const EventPoller::Ptr &poller, const RtspMediaSourceImp::Ptr &src,
-                 const std::shared_ptr<void> &ownership, const mediakit::MediaInfo &info, const mediakit::ProtocolOption &option);
+                 const std::shared_ptr<void> &ownership, const MediaInfo &info, const ProtocolOption &option);
 private:
    bool _simulcast = false;
    //断连续推延时
    uint32_t _continue_push_ms = 0;
    //媒体相关元数据
-    mediakit::MediaInfo _media_info;
+    MediaInfo _media_info;
    //推流的rtsp源
-    mediakit::RtspMediaSourceImp::Ptr _push_src;
+    RtspMediaSourceImp::Ptr _push_src;
    //推流所有权
    std::shared_ptr<void> _push_src_ownership;
    //推流的rtsp源,支持simulcast
-    std::unordered_map<std::string/*rid*/, mediakit::RtspMediaSource::Ptr> _push_src_sim;
+    std::unordered_map<std::string/*rid*/, RtspMediaSource::Ptr> _push_src_sim;
    std::unordered_map<std::string/*rid*/, std::shared_ptr<void> > _push_src_sim_ownership;
 };
+}// namespace mediakit
 #endif //ZLMEDIAKIT_WEBRTCPUSHER_H
--- a/webrtc/WebRtcSession.cpp
+++ b/webrtc/WebRtcSession.cpp
@@ -12,7 +12,8 @@
 #include "Util/util.h"
 using namespace std;
-using namespace mediakit;
+namespace mediakit {
 static string getUserName(const Buffer::Ptr &buffer) {
    auto buf = buffer->data();
@@ -85,7 +86,7 @@ void WebRtcSession::onError(const SockException &err) {
 }
 void WebRtcSession::onManager() {
-    GET_CONFIG(float, timeoutSec, RTC::kTimeOutSec);
+    GET_CONFIG(float, timeoutSec, Rtc::kTimeOutSec);
    if (!_transport && _ticker.createdTime() > timeoutSec * 1000) {
        shutdown(SockException(Err_timeout, "illegal webrtc connection"));
        return;
@@ -96,5 +97,6 @@ void WebRtcSession::onManager() {
    }
 }
+}// namespace mediakit
--- a/webrtc/WebRtcSession.h
+++ b/webrtc/WebRtcSession.h
@@ -16,6 +16,8 @@
 #include "IceServer.hpp"
 #include "WebRtcTransport.h"
+namespace mediakit {
 class WebRtcSession : public UdpSession {
 public:
    WebRtcSession(const Socket::Ptr &sock);
@@ -35,5 +37,6 @@ private:
    std::shared_ptr<WebRtcTransportImp> _transport;
 };
+}// namespace mediakit
 #endif //ZLMEDIAKIT_WEBRTCSESSION_H
--- a/webrtc/WebRtcTransport.cpp
+++ b/webrtc/WebRtcTransport.cpp
@@ -8,15 +8,18 @@
 * may be found in the AUTHORS file in the root of the source tree.
 */
-#include "WebRtcTransport.h"
+#include <iostream>
+#include <srtp2/srtp.h>
+#include "RtpExt.h"
 #include "Rtcp/Rtcp.h"
 #include "Rtcp/RtcpFCI.h"
-#include "RtpExt.h"
 #include "Rtsp/RtpReceiver.h"
+#include "WebRtcTransport.h"
-#include <srtp2/srtp.h>
+#include "WebRtcEchoTest.h"
+#include "WebRtcPlayer.h"
-#include <iostream>
+#include "WebRtcPusher.h"
 #define RTP_SSRC_OFFSET 1
 #define RTX_SSRC_OFFSET 2
@@ -26,10 +29,11 @@
 #define RTP_MSID RTP_MSLABEL " " RTP_LABEL
 using namespace std;
-using namespace mediakit;
+namespace mediakit {
 // RTC配置项目
-namespace RTC {
+namespace Rtc {
 #define RTC_FIELD "rtc."
 // rtp和rtcp接受超时时间
 const string kTimeOutSec = RTC_FIELD "timeoutSec";
@@ -246,7 +250,7 @@ void WebRtcTransport::setRemoteDtlsFingerprint(const RtcSession &remote) {
 void WebRtcTransport::onRtcConfigure(RtcConfigure &configure) const {
    // 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);
+    GET_CONFIG(size_t, remb_bit_rate, Rtc::kRembBitRate);
    configure.enableTWCC(!remb_bit_rate);
 }
@@ -368,7 +372,7 @@ void WebRtcTransportImp::onCreate() {
    registerSelf();
    weak_ptr<WebRtcTransportImp> weak_self = static_pointer_cast<WebRtcTransportImp>(shared_from_this());
-    GET_CONFIG(float, timeoutSec, RTC::kTimeOutSec);
+    GET_CONFIG(float, timeoutSec, Rtc::kTimeOutSec);
    _timer = std::make_shared<Timer>(
        timeoutSec / 2,
        [weak_self]() {
@@ -511,7 +515,7 @@ void WebRtcTransportImp::onStartWebRTC() {
 void WebRtcTransportImp::onCheckAnswer(RtcSession &sdp) {
    // 修改answer sdp的ip、端口信息
-    GET_CONFIG_FUNC(std::vector<std::string>, extern_ips, RTC::kExternIP, [](string str) {
+    GET_CONFIG_FUNC(std::vector<std::string>, extern_ips, Rtc::kExternIP, [](string str) {
        std::vector<std::string> ret;
        if (str.length()) {
            ret = split(str, ",");
@@ -525,7 +529,7 @@ void WebRtcTransportImp::onCheckAnswer(RtcSession &sdp) {
        m.rtcp_addr.reset();
        m.rtcp_addr.address = m.addr.address;
-        GET_CONFIG(uint16_t, local_port, RTC::kPort);
+        GET_CONFIG(uint16_t, local_port, Rtc::kPort);
        m.rtcp_addr.port = local_port;
        m.port = m.rtcp_addr.port;
        sdp.origin.address = m.addr.address;
@@ -592,9 +596,9 @@ makeIceCandidate(std::string ip, uint16_t port, uint32_t priority = 100, std::st
 void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
    WebRtcTransport::onRtcConfigure(configure);
-    GET_CONFIG(uint16_t, local_port, RTC::kPort);
+    GET_CONFIG(uint16_t, local_port, Rtc::kPort);
    // 添加接收端口candidate信息
-    GET_CONFIG_FUNC(std::vector<std::string>, extern_ips, RTC::kExternIP, [](string str) {
+    GET_CONFIG_FUNC(std::vector<std::string>, extern_ips, Rtc::kExternIP, [](string str) {
        std::vector<std::string> ret;
        if (str.length()) {
            ret = split(str, ",");
@@ -698,7 +702,7 @@ std::shared_ptr<RtpChannel> MediaTrack::getRtpChannel(uint32_t ssrc) const {
    return it_chn->second;
 }
-float WebRtcTransportImp::getLossRate(mediakit::TrackType type) {
+float WebRtcTransportImp::getLossRate(TrackType type) {
    for (auto &pr : _ssrc_to_track) {
        auto ssrc = pr.first;
        auto &track = pr.second;
@@ -942,7 +946,7 @@ void WebRtcTransportImp::onSortedRtp(MediaTrack &track, const string &rid, RtpPa
        sendRtcpPli(rtp->getSSRC());
        // 开启remb，则发送remb包调节比特率
-        GET_CONFIG(size_t, remb_bit_rate, RTC::kRembBitRate);
+        GET_CONFIG(size_t, remb_bit_rate, Rtc::kRembBitRate);
        if (remb_bit_rate && _answer_sdp->supportRtcpFb(SdpConst::kRembRtcpFb)) {
            sendRtcpRemb(rtp->getSSRC(), remb_bit_rate);
        }
@@ -1121,10 +1125,6 @@ void WebRtcPluginManager::getAnswerSdp(
    it->second(sender, offer, args, cb);
 }
-#include "WebRtcEchoTest.h"
-#include "WebRtcPlayer.h"
-#include "WebRtcPusher.h"
 void echo_plugin(
    Session &sender, const string &offer, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
    cb(*WebRtcEchoTest::create(EventPollerPool::Instance().getPoller()));
@@ -1227,3 +1227,5 @@ static onceToken s_rtc_auto_register([]() {
    WebRtcPluginManager::Instance().registerPlugin("push", push_plugin);
    WebRtcPluginManager::Instance().registerPlugin("play", play_plugin);
 });
+}// namespace mediakit
\ No newline at end of file
--- a/webrtc/WebRtcTransport.h
+++ b/webrtc/WebRtcTransport.h
@@ -27,8 +27,10 @@
 #include "TwccContext.h"
 #include "SctpAssociation.hpp"
+namespace mediakit {
 //RTC配置项目
-namespace RTC {
+namespace Rtc {
 extern const std::string kPort;
 extern const std::string kTimeOutSec;
 }//namespace RTC
@@ -200,7 +202,7 @@ public:
    //for send rtp
    NackList nack_list;
-    mediakit::RtcpContext::Ptr rtcp_context_send;
+    RtcpContext::Ptr rtcp_context_send;
    //for recv rtp
    std::unordered_map<std::string/*rid*/, std::shared_ptr<RtpChannel> > rtp_channel;
@@ -211,13 +213,13 @@ struct WrappedMediaTrack {
    MediaTrack::Ptr track;
    explicit WrappedMediaTrack(MediaTrack::Ptr ptr): track(ptr) {}
    virtual ~WrappedMediaTrack() {}
-    virtual void inputRtp(const char *buf, size_t len, uint64_t stamp_ms, mediakit::RtpHeader *rtp) = 0;
+    virtual void inputRtp(const char *buf, size_t len, uint64_t stamp_ms, RtpHeader *rtp) = 0;
 };
 struct WrappedRtxTrack: public WrappedMediaTrack {
    explicit WrappedRtxTrack(MediaTrack::Ptr ptr)
        : WrappedMediaTrack(std::move(ptr)) {}
-    void inputRtp(const char *buf, size_t len, uint64_t stamp_ms, mediakit::RtpHeader *rtp) override;
+    void inputRtp(const char *buf, size_t len, uint64_t stamp_ms, RtpHeader *rtp) override;
 };
 class WebRtcTransportImp;
@@ -229,7 +231,7 @@ struct WrappedRtpTrack : public WrappedMediaTrack {
        , _transport(t) {}
    TwccContext& _twcc_ctx;
    WebRtcTransportImp& _transport;
-    void inputRtp(const char *buf, size_t len, uint64_t stamp_ms, mediakit::RtpHeader *rtp) override;
+    void inputRtp(const char *buf, size_t len, uint64_t stamp_ms, RtpHeader *rtp) override;
 };
 class WebRtcTransportImp : public WebRtcTransport {
@@ -243,7 +245,7 @@ public:
    uint64_t getDuration() const;
    bool canSendRtp() const;
    bool canRecvRtp() const;
-    void onSendRtp(const mediakit::RtpPacket::Ptr &rtp, bool flush, bool rtx = false);
+    void onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool rtx = false);
    void createRtpChannel(const std::string &rid, uint32_t ssrc, MediaTrack &track);
@@ -262,14 +264,14 @@ protected:
    void onCreate() override;
    void onDestory() override;
    void onShutdown(const SockException &ex) override;
-    virtual void onRecvRtp(MediaTrack &track, const std::string &rid, mediakit::RtpPacket::Ptr rtp) = 0;
+    virtual void onRecvRtp(MediaTrack &track, const std::string &rid, RtpPacket::Ptr rtp) = 0;
    void updateTicker();
-    float getLossRate(mediakit::TrackType type);
+    float getLossRate(TrackType type);
    void onRtcpBye() override;
 private:
-    void onSortedRtp(MediaTrack &track, const std::string &rid, mediakit::RtpPacket::Ptr rtp);
+    void onSortedRtp(MediaTrack &track, const std::string &rid, RtpPacket::Ptr rtp);
-    void onSendNack(MediaTrack &track, const mediakit::FCI_NACK &nack, uint32_t ssrc);
+    void onSendNack(MediaTrack &track, const FCI_NACK &nack, uint32_t ssrc);
    void onSendTwcc(uint32_t ssrc, const std::string &twcc_fci);
    void registerSelf();
@@ -343,4 +345,6 @@ private:
 private:
    mutable std::mutex _mtx_creator;
    std::unordered_map<std::string, Plugin> _map_creator;
 };
\ No newline at end of file
+}// namespace mediakit
\ No newline at end of file
--- a/webrtc/readme.md
+++ b/webrtc/readme.md
@@ -15,7 +15,10 @@
 - srtp相关功能：
   - SrtpSession.cpp
   - SrtpSession.hpp
+- datachannel相关功能：
+  - SctpAssociation.cpp
+  - SctpAssociation.hpp
 以上源码有一定的修改和裁剪，感谢MediaSoup开源项目及作者，
 用户在使用本项目的同时，应该同时遵循MediaSoup的开源协议。