H.264 packetization mode 0 (try 2)

webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
index 27479458315de126fa3e551128a748b618a81714..a9c43bf66af35572470ec5014a89869effaadc1e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
@@ -11,6 +11,7 @@
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_h264.h"
 
 #include <string.h>
+
 #include <memory>
 #include <utility>
 #include <vector>
@@ -77,9 +78,16 @@ bool ParseStapAStartOffsets(const uint8_t* nalu_ptr,
 
 }  // namespace
 
-RtpPacketizerH264::RtpPacketizerH264(FrameType frame_type,
+RtpPacketizerH264::RtpPacketizerH264(const RTPVideoHeaderH264& parameters,
                                      size_t max_payload_len)
-    : max_payload_len_(max_payload_len) {}
+    : packetization_mode_(parameters.packetization_mode),
+      max_payload_len_(max_payload_len) {
+  if (packetization_mode_ == kH264PacketizationModeNotSet) {
+    LOG(LS_WARNING)
+        << "Creating H.264 RTP packetizer with default packetization mode";
+    packetization_mode_ = kH264PacketizationMode1;

[hbos, 2016/10/31 10:32:26]

From https://tools.ietf.org/html/rfc6184#section-6.2:
"This mode is in use when the value of the OPTIONAL packetization-mode media
type parameter is equal to 0 or the packetization-mode is not present."

If not present in the SDP, mode 0 is used. Does it make more sense for the
default to be Mode0 here as well or do we make sure to always set it if missing
in the SDP?

[hta-webrtc, 2016/10/31 15:03:04]

I started out with a RTC_CHECK here instead of the LOG(LS_WARNING). So I know
all callers set it - EXCEPT for a few android java tests that have been added in
the last few months - and I couldn't figure out which code path they were using
to get the uninitialized value.

So in the interest of not changing existing behavior, I made it LOG(WARNING) + a
default of mode 1.

[hbos, 2016/10/31 16:27:15]

Acknowledged.

+  }
+}
 
 RtpPacketizerH264::~RtpPacketizerH264() {
 }
@@ -163,6 +171,7 @@ void RtpPacketizerH264::SetPayloadData(
 void RtpPacketizerH264::GeneratePackets() {
   for (size_t i = 0; i < input_fragments_.size();) {
     if (input_fragments_[i].length > max_payload_len_) {
+      RTC_CHECK(packetization_mode_ == kH264PacketizationMode1);

[hbos, 2016/10/31 10:32:26]

nit: Here and other places in this file: RTC_CHECK_EQ

[hta-webrtc, 2016/10/31 15:03:04]

Acknowledged.

       PacketizeFuA(i);
       ++i;
     } else {
@@ -249,9 +258,11 @@ bool RtpPacketizerH264::NextPacket(uint8_t* buffer,
     input_fragments_.pop_front();
     RTC_CHECK_LE(*bytes_to_send, max_payload_len_);
   } else if (packet.aggregated) {
+    RTC_CHECK(packetization_mode_ == kH264PacketizationMode1);

[magjed_webrtc, 2016/10/31 13:00:50]

nit: use RTC_CHECK_EQ

     NextAggregatePacket(buffer, bytes_to_send);
     RTC_CHECK_LE(*bytes_to_send, max_payload_len_);
   } else {
+    RTC_CHECK(packetization_mode_ == kH264PacketizationMode1);
     NextFragmentPacket(buffer, bytes_to_send);
     RTC_CHECK_LE(*bytes_to_send, max_payload_len_);
   }