RTC: Refine code

trunk/src/app/srs_app_rtc_conn.cpp

@@ -58,6 +58,8 @@ using namespace std;
 #include <srs_app_rtc_source.hpp>
 #include <srs_protocol_utility.hpp>
 
+#define SRS_TICKID_RTCP 0
+
 SrsSecurityTransport::SrsSecurityTransport(SrsRtcConnection* s)
 {
     session_ = s;
@@ -227,6 +229,7 @@ SrsRtcPlayStream::SrsRtcPlayStream(SrsRtcConnection* s, SrsContextId parent_cid)
     nack_enabled_ = false;
 
     _srs_config->subscribe(this);
+    timer_ = new SrsHourGlass(this, 1000 * SRS_UTIME_MILLISECONDS);
 }
 
 SrsRtcPlayStream::~SrsRtcPlayStream()
@@ -234,6 +237,7 @@ SrsRtcPlayStream::~SrsRtcPlayStream()
     _srs_config->unsubscribe(this);
 
     srs_freep(trd);
+    srs_freep(timer_);
 
     if (true) {
         std::map<uint32_t, SrsRtcAudioSendTrack*>::iterator it;
@@ -321,6 +325,10 @@ srs_error_t SrsRtcPlayStream::start()
         return srs_error_wrap(err, "rtc_sender");
     }
 
+    if ((err = timer_->start()) != srs_success) {
+        return srs_error_wrap(err, "start timer");
+    }
+
     if (_srs_rtc_hijacker) {
         if ((err = _srs_rtc_hijacker->on_start_play(session_, this, session_->req)) != srs_success) {
             return srs_error_wrap(err, "on start play");
@@ -517,6 +525,17 @@ void SrsRtcPlayStream::nack_fetch(vector<SrsRtpPacket2*>& pkts, uint32_t ssrc, u
     }
 }
 
+srs_error_t SrsRtcPlayStream::notify(int type, srs_utime_t interval, srs_utime_t tick)
+{
+    srs_error_t err = srs_success;
+
+    if (!is_started) {
+        return err;
+    }
+
+    return err;
+}
+
 srs_error_t SrsRtcPlayStream::on_rtcp(char* data, int nb_data)
 {
     srs_error_t err = srs_success;
@@ -770,7 +789,7 @@ uint32_t SrsRtcPlayStream::get_video_publish_ssrc(uint32_t play_ssrc)
 
 SrsRtcPublishStream::SrsRtcPublishStream(SrsRtcConnection* session)
 {
-    report_timer = new SrsHourGlass(this, 200 * SRS_UTIME_MILLISECONDS);
+    timer_ = new SrsHourGlass(this, 200 * SRS_UTIME_MILLISECONDS);
 
     session_ = session;
     request_keyframe_ = false;
@@ -795,7 +814,7 @@ SrsRtcPublishStream::~SrsRtcPublishStream()
     }
 
     srs_freep(req);
-    srs_freep(report_timer);
+    srs_freep(timer_);
 }
 
 srs_error_t SrsRtcPublishStream::initialize(SrsRequest* r, SrsRtcStreamDescription* stream_desc)
@@ -841,17 +860,16 @@ srs_error_t SrsRtcPublishStream::start()
 {
     srs_error_t err = srs_success;
 
-    // If report_timer started, we think the publisher is started.
     if (is_started) {
         return err;
     }
 
-    if ((err = report_timer->tick(0 * SRS_UTIME_MILLISECONDS)) != srs_success) {
-        return srs_error_wrap(err, "hourglass tick");
+    if ((err = timer_->tick(SRS_TICKID_RTCP, 200 * SRS_UTIME_MILLISECONDS)) != srs_success) {
+        return srs_error_wrap(err, "rtcp tick");
     }
 
-    if ((err = report_timer->start()) != srs_success) {
-        return srs_error_wrap(err, "start report_timer");
+    if ((err = timer_->start()) != srs_success) {
+        return srs_error_wrap(err, "start timer");
     }
 
     if ((err = _srs_rtc_sources->fetch_or_create(req, &source)) != srs_success) {
@@ -1437,6 +1455,11 @@ srs_error_t SrsRtcPublishStream::notify(int type, srs_utime_t interval, srs_utim
 {
     srs_error_t err = srs_success;
 
+    if (!is_started) {
+        return err;
+    }
+
+    if (type == SRS_TICKID_RTCP) {
         // TODO: FIXME: Check error.
         send_rtcp_rr();
         send_rtcp_xr_rrtr();
@@ -1445,6 +1468,7 @@ srs_error_t SrsRtcPublishStream::notify(int type, srs_utime_t interval, srs_utim
         // We should not depends on the received packet,
         // instead we should send feedback every Nms.
         send_periodic_twcc();
+    }
 
     return err;
 }

trunk/src/app/srs_app_rtc_conn.hpp

@@ -166,13 +166,14 @@ public:
 };
 
 // A RTC play stream, client pull and play stream from SRS.
-class SrsRtcPlayStream : virtual public ISrsCoroutineHandler, virtual public ISrsReloadHandler
+class SrsRtcPlayStream : virtual public ISrsCoroutineHandler, virtual public ISrsReloadHandler, virtual public ISrsHourGlass
 {
-protected:
+private:
     SrsContextId _parent_cid;
     SrsCoroutine* trd;
     SrsRtcConnection* session_;
 private:
+    SrsHourGlass* timer_;
     // key: publish_ssrc, value: send track to process rtp/rtcp
     std::map<uint32_t, SrsRtcAudioSendTrack*> audio_tracks_;
     std::map<uint32_t, SrsRtcVideoSendTrack*> video_tracks_;
@@ -207,6 +208,9 @@ private:
     srs_error_t send_packets(SrsRtcStream* source, const std::vector<SrsRtpPacket2*>& pkts, SrsRtcPlayStreamStatistic& info);
 public:
     void nack_fetch(std::vector<SrsRtpPacket2*>& pkts, uint32_t ssrc, uint16_t seq);
+// interface ISrsHourGlass
+public:
+    virtual srs_error_t notify(int type, srs_utime_t interval, srs_utime_t tick);
 public:
     srs_error_t on_rtcp(char* data, int nb_data);
 private:
@@ -222,7 +226,7 @@ private:
 class SrsRtcPublishStream : virtual public ISrsHourGlass, virtual public ISrsRtpPacketDecodeHandler, virtual public ISrsRtcPublishStream
 {
 private:
-    SrsHourGlass* report_timer;
+    SrsHourGlass* timer_;
     uint64_t nn_audio_frames;
 private:
     SrsRtcConnection* session_;

trunk/src/app/srs_app_rtc_source.cpp

@@ -1506,7 +1506,6 @@ SrsRtcTrackDescription* SrsRtcStreamDescription::find_track_description_by_ssrc(
 
 SrsRtcTrackStatistic::SrsRtcTrackStatistic()
 {
-    last_written = 0;
     packets = 0;
     last_packets = 0;
     bytes = 0;

trunk/src/app/srs_app_rtc_source.hpp

@@ -407,8 +407,6 @@ public:
 class SrsRtcTrackStatistic
 {
 public:
-    srs_utime_t last_written;
-
 	// packets received or sent.
 	uint32_t packets;
 	// packets received or sent at last statistic time.