send config frames once before sending DirectProxy RTP packets

webrtc/WebRtcPlayer.cpp

@@ -9,7 +9,10 @@
  */
 
 #include "WebRtcPlayer.h"
+
 #include "Common/config.h"
+#include "Extension/Factory.h"
+#include "Util/base64.h"
 
 using namespace std;
 
@@ -32,6 +35,9 @@ WebRtcPlayer::WebRtcPlayer(const EventPoller::Ptr &poller,
     _media_info = info;
     _play_src = src;
     CHECK(src);
+
+    GET_CONFIG(bool, direct_proxy, Rtsp::kDirectProxy);
+    _send_config_frames_once = direct_proxy;
 }
 
 void WebRtcPlayer::onStartWebRTC() {
@@ -56,6 +62,13 @@ void WebRtcPlayer::onStartWebRTC() {
             if (!strong_self) {
                 return;
             }
+
+            if (strong_self->_send_config_frames_once && !pkt->empty()) {
+                const auto &first_rtp = pkt->front();
+                strong_self->sendConfigFrames(first_rtp->getSeq(), first_rtp->sample_rate, first_rtp->getStamp(), first_rtp->ntp_stamp);
+                strong_self->_send_config_frames_once = false;
+            }
+
             size_t i = 0;
             pkt->for_each([&](const RtpPacket::Ptr &rtp) {
                 //TraceL<<"send track type:"<<rtp->type<<" ts:"<<rtp->getStamp()<<" ntp:"<<rtp->ntp_stamp<<" size:"<<rtp->getPayloadSize()<<" i:"<<i;
@@ -111,4 +124,37 @@ void WebRtcPlayer::onRtcConfigure(RtcConfigure &configure) const {
     configure.setPlayRtspInfo(playSrc->getSdp());
 }
 
+void WebRtcPlayer::sendConfigFrames(uint32_t before_seq, uint32_t sample_rate, uint32_t timestamp, uint64_t ntp_timestamp) {
+    auto play_src = _play_src.lock();
+    if (!play_src) {
+        return;
+    }
+    auto video_track = std::dynamic_pointer_cast<mediakit::VideoTrack>(play_src->getTrack(mediakit::TrackVideo));
+    if (!video_track) {
+        return;
+    }
+    auto frames = video_track->getConfigFrames();
+    if (frames.empty()) {
+        return;
+    }
+    auto encoder = mediakit::Factory::getRtpEncoderByCodecId(video_track->getCodecId(), 0);
+    if (!encoder) {
+        return;
+    }
+
+    GET_CONFIG(uint32_t, video_mtu, Rtp::kVideoMtuSize);
+    encoder->setRtpInfo(0, video_mtu, sample_rate, 0, 0, 0);
+
+    auto seq = before_seq - frames.size();
+    for (const auto &frame : video_track->getConfigFrames()) {
+        auto rtp = encoder->getRtpInfo().makeRtp(
+            TrackVideo, frame->data() + frame->prefixSize(), frame->size() - frame->prefixSize(), false, 0);
+        auto header = rtp->getHeader();
+        header->seq = htons(seq++);
+        header->stamp = htonl(timestamp);
+        rtp->ntp_stamp = ntp_timestamp;
+        onSendRtp(rtp, false);
+    }
+}
+
 }// namespace mediakit

webrtc/WebRtcPlayer.h

@@ -31,11 +31,17 @@ protected:
 private:
     WebRtcPlayer(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src, const MediaInfo &info);
 
+    void sendConfigFrames(uint32_t before_seq, uint32_t sample_rate, uint32_t timestamp, uint64_t ntp_timestamp);
+
 private:
     //媒体相关元数据
     MediaInfo _media_info;
     //播放的rtsp源
     std::weak_ptr<RtspMediaSource> _play_src;
+
+    // rtp 直接转发情况下通常会缺少 sps/pps, 在转发 rtp 前, 先发送一次相关帧信息, 部分情况下是可以播放的
+    bool _send_config_frames_once { false };
+
     //播放rtsp源的reader对象
     RtspMediaSource::RingType::RingReader::Ptr _reader;
 };