Refactor RTC publisher code

trunk/src/app/srs_app_rtc_conn.cpp

@@ -1935,9 +1935,12 @@ srs_error_t SrsRtcPublisher::on_audio(SrsUdpMuxSocket* skt, SrsRtpSharedPacket*
         return srs_error_wrap(err, "rtp opus demux failed");
     }
 
+    // TODO: FIXME: Rename it.
+    // TODO: FIXME: Error check.
     rtp_audio_queue->insert(rtp_pkt);
 
-    if (rtp_audio_queue->get_and_clean_if_needed_rqeuest_key_frame()) {
+    if (rtp_audio_queue->get_and_clean_if_needed_request_key_frame()) {
+        // TODO: FIXME: Check error.
         send_rtcp_fb_pli(skt, audio_ssrc);
     }
 
@@ -1946,27 +1949,6 @@ srs_error_t SrsRtcPublisher::on_audio(SrsUdpMuxSocket* skt, SrsRtpSharedPacket*
     return collect_audio_frame();
 }
 
-srs_error_t SrsRtcPublisher::on_video(SrsUdpMuxSocket* skt, SrsRtpSharedPacket* rtp_pkt)
-{
-    srs_error_t err = srs_success;
-
-    rtp_pkt->rtp_payload_header = new SrsRtpH264Header();
-
-    if ((err = rtp_h264_demuxer->parse(rtp_pkt)) != srs_success) {
-        return srs_error_wrap(err, "rtp h264 demux failed");
-    }
-
-    rtp_video_queue->insert(rtp_pkt);
-
-    if (rtp_video_queue->get_and_clean_if_needed_rqeuest_key_frame()) {
-        send_rtcp_fb_pli(skt, video_ssrc);
-    }
-
-    check_send_nacks(rtp_video_queue, video_ssrc, skt);
-
-    return collect_video_frame();
-}
-
 srs_error_t SrsRtcPublisher::collect_audio_frame()
 {
     srs_error_t err = srs_success;
@@ -1980,6 +1962,8 @@ srs_error_t SrsRtcPublisher::collect_audio_frame()
                 frames.size(), frames[i].front()->rtp_header.get_sequence(), frames[i].back()->rtp_header.get_sequence());
         }
 
+        // TODO: FIXME: Write audio frame to source.
+
         for (size_t n = 0; n < frames[i].size(); ++n) {
             srs_freep(frames[i][n]);
         }
@@ -1988,6 +1972,27 @@ srs_error_t SrsRtcPublisher::collect_audio_frame()
     return err;
 }
 
+srs_error_t SrsRtcPublisher::on_video(SrsUdpMuxSocket* skt, SrsRtpSharedPacket* rtp_pkt)
+{
+    srs_error_t err = srs_success;
+
+    rtp_pkt->rtp_payload_header = new SrsRtpH264Header();
+
+    if ((err = rtp_h264_demuxer->parse(rtp_pkt)) != srs_success) {
+        return srs_error_wrap(err, "rtp h264 demux failed");
+    }
+
+    rtp_video_queue->insert(rtp_pkt);
+
+    if (rtp_video_queue->get_and_clean_if_needed_request_key_frame()) {
+        send_rtcp_fb_pli(skt, video_ssrc);
+    }
+
+    check_send_nacks(rtp_video_queue, video_ssrc, skt);
+
+    return collect_video_frame();
+}
+
 srs_error_t SrsRtcPublisher::collect_video_frame()
 {
     srs_error_t err = srs_success;

trunk/src/app/srs_app_rtc_conn.hpp

@@ -292,10 +292,11 @@ private:
     srs_error_t send_rtcp_fb_pli(SrsUdpMuxSocket* skt, uint32_t ssrc);
 private:
     srs_error_t on_audio(SrsUdpMuxSocket* skt, SrsRtpSharedPacket* rtp_pkt);
+    srs_error_t collect_audio_frame();
+private:
     srs_error_t on_video(SrsUdpMuxSocket* skt, SrsRtpSharedPacket* rtp_pkt);
 private:
     srs_error_t collect_video_frame();
-    srs_error_t collect_audio_frame();
 public:
     void update_sendonly_socket(SrsUdpMuxSocket* skt);
 // interface ISrsHourGlass

trunk/src/app/srs_app_rtp_queue.cpp

@@ -171,6 +171,7 @@ srs_error_t SrsRtpQueue::insert(SrsRtpSharedPacket* rtp_pkt)
 
     uint16_t seq = rtp_pkt->rtp_header.get_sequence();
 
+    // TODO: FIXME: Update time for each packet, may hurt performance.
     srs_utime_t now = srs_update_system_time();
 
     // First packet recv, init head_sequence and highest_sequence.
@@ -181,6 +182,7 @@ srs_error_t SrsRtpQueue::insert(SrsRtpSharedPacket* rtp_pkt)
 
         ++num_of_packet_received_;
 
+        // TODO: FIXME: Covert time to srs_utime_t.
         last_trans_time_ = now/1000 - rtp_pkt->rtp_header.get_timestamp()/90;
     } else {
         SrsRtpNackInfo* nack_info = NULL;
@@ -267,6 +269,7 @@ srs_error_t SrsRtpQueue::insert(SrsRtpSharedPacket* rtp_pkt)
         delete old_pkt;
     }
 
+    // TODO: FIXME: Change to ptr of ptr.
     old_pkt = rtp_pkt->copy();
 
     // Marker bit means the last packet of frame received.
@@ -295,7 +298,7 @@ void SrsRtpQueue::get_and_clean_collected_frames(std::vector<std::vector<SrsRtpS
     frames.swap(frames_);
 }
 
-bool SrsRtpQueue::get_and_clean_if_needed_rqeuest_key_frame()
+bool SrsRtpQueue::get_and_clean_if_needed_request_key_frame()
 {
     if (request_key_frame_) {
         request_key_frame_ = false;

trunk/src/app/srs_app_rtp_queue.hpp

@@ -142,7 +142,7 @@ public:
     srs_error_t remove(uint16_t seq);
 public:
     void get_and_clean_collected_frames(std::vector<std::vector<SrsRtpSharedPacket*> >& frames);
-    bool get_and_clean_if_needed_rqeuest_key_frame();
+    bool get_and_clean_if_needed_request_key_frame();
     void notify_drop_seq(uint16_t seq);
     void notify_nack_list_full();
 public: