H.264 packetization mode 0 (try 2)

webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc

 /*
  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_h264.h"
 
 #include <string.h>
+
 #include <memory>
 #include <utility>
 #include <vector>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/common_video/h264/sps_vui_rewriter.h"
 #include "webrtc/common_video/h264/h264_common.h"
@@ skipping 46 matching lines @@
     length_remaining -= nalu_size;
 
     offsets->push_back(offset + kStapAHeaderSize);
     offset += kLengthFieldSize + nalu_size;
   }
   return true;
 }
 
 }  // 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;
+  }
+}
 
 RtpPacketizerH264::~RtpPacketizerH264() {
 }
 
 RtpPacketizerH264::Fragment::Fragment(const uint8_t* buffer, size_t length)
     : buffer(buffer), length(length) {}
 RtpPacketizerH264::Fragment::Fragment(const Fragment& fragment)
     : buffer(fragment.buffer), length(fragment.length) {}
 
 void RtpPacketizerH264::SetPayloadData(
@@ skipping 63 matching lines @@
 
     if (!updated_sps)
       input_fragments_.push_back(Fragment(buffer, length));
   }
   GeneratePackets();
 }
 
 void RtpPacketizerH264::GeneratePackets() {
   for (size_t i = 0; i < input_fragments_.size();) {
     if (input_fragments_[i].length > max_payload_len_) {
+      RTC_CHECK_EQ(packetization_mode_, kH264PacketizationMode1);
       PacketizeFuA(i);
       ++i;
     } else {
       i = PacketizeStapA(i);
     }
   }
 }
 
 void RtpPacketizerH264::PacketizeFuA(size_t fragment_index) {
   // Fragment payload into packets (FU-A).
@@ skipping 66 matching lines @@
   PacketUnit packet = packets_.front();
 
   if (packet.first_fragment && packet.last_fragment) {
     // Single NAL unit packet.
     *bytes_to_send = packet.source_fragment.length;
     memcpy(buffer, packet.source_fragment.buffer, *bytes_to_send);
     packets_.pop();
     input_fragments_.pop_front();
     RTC_CHECK_LE(*bytes_to_send, max_payload_len_);
   } else if (packet.aggregated) {
+    RTC_CHECK_EQ(packetization_mode_, kH264PacketizationMode1);
     NextAggregatePacket(buffer, bytes_to_send);
     RTC_CHECK_LE(*bytes_to_send, max_payload_len_);
   } else {
+    RTC_CHECK_EQ(packetization_mode_, kH264PacketizationMode1);
     NextFragmentPacket(buffer, bytes_to_send);
     RTC_CHECK_LE(*bytes_to_send, max_payload_len_);
   }
   *last_packet = packets_.empty();
   return true;
 }
 
 void RtpPacketizerH264::NextAggregatePacket(uint8_t* buffer,
                                             size_t* bytes_to_send) {
   PacketUnit* packet = &packets_.front();
@@ skipping 328 matching lines @@
   h264->packetization_type = kH264FuA;
   h264->nalu_type = original_nal_type;
   if (first_fragment) {
     h264->nalus[h264->nalus_length] = nalu;
     h264->nalus_length = 1;
   }
   return true;
 }
 
 }  // namespace webrtc