Fix packetization logic to leave space for extensions in the last packet

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 b89aefef53e44977f2bda96438bd16cb3b4572af..de146bc92dc233ad0838ec9138a447363ba284f7 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -300,6 +300,8 @@ 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);
+  std::unique_ptr<RtpPacketToSend> rtp_header_with_extensions(

[danilchap, 2017/05/12 08:34:40]

this name makes false impression rtp_header doesn't contain any extensions
may be call it last_packet?

[ilnik, 2017/05/12 09:17:57]

Done.

+      new RtpPacketToSend(*rtp_header));

[danilchap, 2017/05/12 08:34:40]

in base/ptr_util.h there is helper for this scenario of using new:
auto last_packet = rtc::MakeUnique<RtpPacketToSend>(*rtp_header);

[ilnik, 2017/05/12 09:17:58]

Done.

 
   size_t fec_packet_overhead;
   bool red_enabled;
@@ -322,12 +324,13 @@ bool RTPSenderVideo::SendVideo(RtpVideoCodecTypes video_type,
       VideoRotation current_rotation = video_header->rotation;
       if (frame_type == kVideoFrameKey || current_rotation != last_rotation_ ||
           current_rotation != kVideoRotation_0)
-        rtp_header->SetExtension<VideoOrientation>(current_rotation);
+        rtp_header_with_extensions->SetExtension<VideoOrientation>(
+            current_rotation);
       last_rotation_ = current_rotation;
       // Report content type only for key frames.
       if (frame_type == kVideoFrameKey &&
           video_header->content_type != VideoContentType::UNSPECIFIED) {
-        rtp_header->SetExtension<VideoContentTypeExtension>(
+        rtp_header_with_extensions->SetExtension<VideoContentTypeExtension>(
             video_header->content_type);
       }
     }
@@ -350,10 +353,13 @@ bool RTPSenderVideo::SendVideo(RtpVideoCodecTypes video_type,
                            (rtp_sender_->RtxStatus() ? kRtxHeaderSize : 0);
   RTC_DCHECK_LE(packet_capacity, rtp_header->capacity());
   RTC_DCHECK_GT(packet_capacity, rtp_header->headers_size());
+  RTC_DCHECK_GT(packet_capacity, rtp_header_with_extensions->headers_size());
   size_t max_data_payload_length = packet_capacity - rtp_header->headers_size();
+  size_t last_packet_extensions_len =
+      rtp_header_with_extensions->headers_size() - rtp_header->headers_size();
 
   std::unique_ptr<RtpPacketizer> packetizer(RtpPacketizer::Create(
-      video_type, max_data_payload_length,
+      video_type, max_data_payload_length, last_packet_extensions_len,
       video_header ? &(video_header->codecHeader) : nullptr, frame_type));
   // Media packet storage.
   StorageType storage = packetizer->GetStorageType(retransmission_settings);
@@ -366,14 +372,24 @@ bool RTPSenderVideo::SendVideo(RtpVideoCodecTypes video_type,
   packetizer->SetPayloadData(payload_data, payload_size, frag);
 
   bool first_frame = first_frame_sent_();
-  bool first = true;
   bool last = false;

[danilchap, 2017/05/12 08:34:40]

is it used outside for loop?

[ilnik, 2017/05/12 09:17:57]

No. Moved inside.

-  while (!last) {
-    std::unique_ptr<RtpPacketToSend> packet(new RtpPacketToSend(*rtp_header));
-
-    if (!packetizer->NextPacket(packet.get(), &last))
+  size_t total_packets = packetizer->TotalPackets();

[danilchap, 2017/05/12 08:34:40]

may be num_packets

[ilnik, 2017/05/12 09:17:57]

Done.

+  for (size_t current_packet = 0; current_packet < total_packets;

[danilchap, 2017/05/12 08:34:40]

may be name iterator to show it is not a packet (specially that there lot's of
variables nearby that are), but an index.
Simple 'i' might work.

[ilnik, 2017/05/12 09:17:57]

Done.

+       ++current_packet) {
+    RtpPacketToSend* rtp_header_to_use = rtp_header.get();
+    last = (current_packet + 1) == total_packets;
+    if (last)
+      rtp_header_to_use = rtp_header_with_extensions.get();
+    std::unique_ptr<RtpPacketToSend> packet(
+        new RtpPacketToSend(*rtp_header_to_use));
+    if (!packetizer->NextPacket(packet.get()))
       return false;
-    RTC_DCHECK_LE(packet->payload_size(), max_data_payload_length);
+    if (last) {

[danilchap, 2017/05/12 08:34:40]

may be instead of branch
RTC_DCHECK_LE(packet->size(), packet_capacity);

[ilnik, 2017/05/12 09:17:57]

Great idea. Done.

+      RTC_DCHECK_LE(packet->payload_size(),
+                    max_data_payload_length - last_packet_extensions_len);
+    } else {
+      RTC_DCHECK_LE(packet->payload_size(), max_data_payload_length);
+    }
 
     if (!rtp_sender_->AssignSequenceNumber(packet.get()))
       return false;
@@ -392,7 +408,7 @@ bool RTPSenderVideo::SendVideo(RtpVideoCodecTypes video_type,
     }
 
     if (first_frame) {
-      if (first) {
+      if (current_packet == 0) {
         LOG(LS_INFO)
             << "Sent first RTP packet of the first video frame (pre-pacer)";
       }
@@ -401,7 +417,6 @@ bool RTPSenderVideo::SendVideo(RtpVideoCodecTypes video_type,
             << "Sent last RTP packet of the first video frame (pre-pacer)";
       }
     }
-    first = false;
   }
 
   TRACE_EVENT_ASYNC_END1("webrtc", "Video", capture_time_ms, "timestamp",