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);
-      ++i;
-    } else {
-      i = PacketizeStapA(i);
+    switch (packetization_mode_) {
+      case H264PacketizationMode::SingleNalUnit:
+        PacketizeSingleNalu(i);
+        ++i;
+        break;
+      case H264PacketizationMode::NonInterleaved:
+        if (input_fragments_[i].length > max_payload_len_) {
+          PacketizeFuA(i);
+          ++i;
+        } else {
+          i = PacketizeStapA(i);
+        }
+        break;

[hbos, 2016/12/02 10:21:26]

case default:
  RTC_NOTREACHED();

[sprang_webrtc, 2016/12/02 10:52:29]

I disagree. If you have an enum class with all values covered, leave default out
so you get a compile time error instead of runtime one.

[hbos, 2016/12/02 11:04:45]

Oh right, neat, no default!

[hta-webrtc, 2016/12/02 11:09:01]

With an enum, it's better to leave the default off, as the compiler will then
tell you that you've failed to handle something.

https://google.github.io/styleguide/cppguide.html#Loops_and_Switch_Statements

     }
   }
 }
 
 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_ == H264PacketizationMode::SingleNalUnit
+              ? "SingleNalUnit"
+              : "NonInterleaved");
+  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(packetization_mode_ == H264PacketizationMode::NonInterleaved);

[hbos, 2016/12/02 10:21:26]

nit: RTC_CHECK_EQ

[hta-webrtc, 2016/12/02 11:09:01]

I tried. The RTC_CHECK_EQ macro borks on "enum class" arguments.
(It was fine when it was just an enum.)

[hbos, 2016/12/02 11:58:42]

Acknowledged.

     NextAggregatePacket(buffer, bytes_to_send);
     RTC_CHECK_LE(*bytes_to_send, max_payload_len_);
   } else {
+    RTC_CHECK(packetization_mode_ == H264PacketizationMode::NonInterleaved);
     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