H.264 packetization mode 0 (try 3)

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.
  */
 
@@ skipping 59 matching lines @@
     length_remaining -= nalu_size;
 
     offsets->push_back(offset + kStapAHeaderSize);
     offset += kLengthFieldSize + nalu_size;
   }
   return true;
 }
 
 }  // namespace
 
-RtpPacketizerH264::RtpPacketizerH264(FrameType frame_type,
-                                     size_t max_payload_len)
-    : max_payload_len_(max_payload_len) {}
+RtpPacketizerH264::RtpPacketizerH264(size_t max_payload_len,
+                                     H264PacketizationMode packetization_mode)
+    : max_payload_len_(max_payload_len),
+      packetization_mode_(packetization_mode) {}
 
 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 62 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_) {
-      PacketizeFuA(i);
+    if (packetization_mode_ == kH264PacketizationMode0) {
+      PacketizeSingleNalu(i);
       ++i;
     } else {
-      i = PacketizeStapA(i);
+      RTC_CHECK_EQ(packetization_mode_, kH264PacketizationMode1);

[sprang_webrtc, 2016/12/01 16:58:57]

If you use enum class you can add switch statement here and get a compilation
error if the model is not covered by a case.

[hta-webrtc, 2016/12/01 18:26:27]

Done.

+      if (input_fragments_[i].length > max_payload_len_) {
+        PacketizeFuA(i);
+        ++i;
+      } else {
+        i = PacketizeStapA(i);
+      }
     }
   }
 }
 
 void RtpPacketizerH264::PacketizeFuA(size_t fragment_index) {
   // Fragment payload into packets (FU-A).
   // Strip out the original header and leave room for the FU-A header.
   const Fragment& fragment = input_fragments_[fragment_index];
 
   size_t fragment_length = fragment.length - kNalHeaderSize;
@@ skipping 42 matching lines @@
     // we need to add the STAP-A NALU header and a length field for the first
     // NALU of this packet.
     if (aggregated_fragments == 0)
       fragment_headers_length += kNalHeaderSize + kLengthFieldSize;
     ++aggregated_fragments;
   }
   packets_.back().last_fragment = true;
   return fragment_index;
 }
 
+void RtpPacketizerH264::PacketizeSingleNalu(size_t fragment_index) {
+  // Add a single NALU to the queue, no aggregation.
+  size_t payload_size_left = max_payload_len_;
+  const Fragment* fragment = &input_fragments_[fragment_index];
+  RTC_CHECK_GE(payload_size_left, fragment->length)
+      << "Payload size left " << payload_size_left << ", fragment length "
+      << fragment->length << ", packetization mode " << packetization_mode_;
+  RTC_CHECK_GT(fragment->length, 0u);
+  packets_.push(PacketUnit(*fragment, true /* first */, true /* last */,
+                           false /* aggregated */, fragment->buffer[0]));
+}
+
 bool RtpPacketizerH264::NextPacket(uint8_t* buffer,
                                    size_t* bytes_to_send,
                                    bool* last_packet) {
   *bytes_to_send = 0;
   if (packets_.empty()) {
     *bytes_to_send = 0;
     *last_packet = true;
     return false;
   }
 
   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 330 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