Implement FrameMarking header extension support

webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
index 94503dcdf1c8c9464a0ffb1b1b67edd689233713..0edbc3104525a6987ebdfc4355314eb580328747 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -301,6 +301,50 @@ bool RTPSenderVideo::SendVideo(RtpVideoCodecTypes video_type,
   rtp_header->SetPayloadType(payload_type);
   rtp_header->SetTimestamp(rtp_timestamp);
   rtp_header->set_capture_time_ms(capture_time_ms);
+
+  // Set Frame Marks.
+  FrameMarks frame_marks;
+  bool frame_marking_enabled = true;
+
+  // Common info
+  frame_marks.start_of_frame = true;
+  frame_marks.end_of_frame = false;
+  frame_marks.independent = (frame_type == kVideoFrameKey);
+
+  // Codec specific.
+  switch (video_type) {

[nisse-webrtc, 2017/08/22 07:44:46]

Should we really add more codec-specific logic in this layer?

Since the information represented by frame marks are supposed to be
codec-independent, they ought to be made available by the encoder classes in
some way so that the information can be used without caring about which encoder
was used.

[sergio.garcia.murillo, 2017/08/25 10:49:38]

The information is codec specific and varies within layers of the same frame, so
it can't be added to the image information (it is only available after the rtp
packetization). This logic could be moved to a framemarking helper function
instead.

+    case kRtpVideoH264:
+      // Nothing to add
+      frame_marks.discardable = false;
+      frame_marks.temporal_layer_id = kNoTemporalIdx;
+      frame_marks.layer_id = kNoSpatialIdx;
+      frame_marks.tl0_pic_idx = static_cast<uint8_t>(kNoTl0PicIdx);
+      break;
+    case kRtpVideoVp8:
+      frame_marks.discardable = video_header->codecHeader.VP8.nonReference;
+      frame_marks.base_layer_sync = video_header->codecHeader.VP8.layerSync;
+      frame_marks.temporal_layer_id = video_header->codecHeader.VP8.temporalIdx;
+      frame_marks.layer_id = kNoSpatialIdx;
+      frame_marks.tl0_pic_idx = video_header->codecHeader.VP8.tl0PicIdx;
+      break;
+    case kRtpVideoVp9:
+      frame_marks.discardable = false;
+      // Layer id format is codec dependant.
+      frame_marks.temporal_layer_id =
+          video_header->codecHeader.VP9.temporal_idx;
+      frame_marks.layer_id =
+          FrameMarking::CreateLayerId(video_header->codecHeader.VP9);
+      frame_marks.tl0_pic_idx = video_header->codecHeader.VP9.tl0_pic_idx;
+      break;
+    default:
+      // Do not use frame marking.
+      frame_marking_enabled = false;
+  }
+  // Only add frame marking for known codecs.
+  if (frame_marking_enabled)
+    // Add extension header for frame marking.
+    rtp_header->SetExtension<FrameMarking>(frame_marks);

[danilchap, 2017/07/17 15:14:39]

[ RUN      ] VideoSendStreamTest.Vp9FlexModeRefCount
==10472==WARNING: MemorySanitizer: use-of-uninitialized-value
    #0 0x1ffddb0 in webrtc::FrameMarking::ValueSize(webrtc::FrameMarks const&)
webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc:451:10
    #1 0x20411d8 in SetExtension<webrtc::FrameMarking, webrtc::FrameMarks>
webrtc/modules/rtp_rtcp/source/rtp_packet.h:186:29
    #2 0x20411d8 in
webrtc::RTPSenderVideo::SendVideo(webrtc::RtpVideoCodecTypes, webrtc::FrameType,
signed char, unsigned int, long, unsigned char const*, unsigned long,
webrtc::RTPFragmentationHeader const*, webrtc::RTPVideoHeader const*)
webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc:346:0

[sergio.garcia.murillo, 2017/08/25 10:49:38]

Done. Created default constructor to initialize members.

+
   auto last_packet = rtc::MakeUnique<RtpPacketToSend>(*rtp_header);
 
   size_t fec_packet_overhead;
@@ -381,6 +425,7 @@ bool RTPSenderVideo::SendVideo(RtpVideoCodecTypes video_type,
 
   bool first_frame = first_frame_sent_();
   for (size_t i = 0; i < num_packets; ++i) {
+    bool first = (i == 0);
     bool last = (i + 1) == num_packets;
     auto packet = last ? std::move(last_packet)
                        : rtc::MakeUnique<RtpPacketToSend>(*rtp_header);
@@ -389,6 +434,16 @@ bool RTPSenderVideo::SendVideo(RtpVideoCodecTypes video_type,
     RTC_DCHECK_LE(packet->payload_size(),
                   last ? max_data_payload_length - last_packet_reduction_len
                        : max_data_payload_length);
+
+    // Only add frame marking for known codecs.
+    if (frame_marking_enabled) {
+      // Update start and end marks.
+      frame_marks.start_of_frame = first;
+      frame_marks.end_of_frame = last;
+      // Update extension header for frame marking.
+      packet->SetExtension<FrameMarking>(frame_marks);
+    }
+
     if (!rtp_sender_->AssignSequenceNumber(packet.get()))
       return false;