Make ostream<< for enum class H264PacketizationMode

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 227 matching lines @@
   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");
+      << 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(RtpPacketToSend* rtp_packet,
                                    bool* last_packet) {
   RTC_DCHECK(rtp_packet);
   RTC_DCHECK(last_packet);
   if (packets_.empty()) {
     *last_packet = true;
     return false;
   }
 
   PacketUnit packet = packets_.front();
   if (packet.first_fragment && packet.last_fragment) {
     // Single NAL unit packet.
     size_t bytes_to_send = packet.source_fragment.length;
     uint8_t* buffer = rtp_packet->AllocatePayload(bytes_to_send);
     memcpy(buffer, packet.source_fragment.buffer, bytes_to_send);
     packets_.pop();
     input_fragments_.pop_front();
   } else if (packet.aggregated) {
-    RTC_CHECK(packetization_mode_ == H264PacketizationMode::NonInterleaved);
+    RTC_CHECK_EQ(packetization_mode_, H264PacketizationMode::NonInterleaved);

[magjed_webrtc, 2016/12/06 15:31:33]

nit: You should swap the arguments, i.e.
RTC_CHECK_EQ(H264PacketizationMode::NonInterleaved, packetization_mode_);

     NextAggregatePacket(rtp_packet);
   } else {
-    RTC_CHECK(packetization_mode_ == H264PacketizationMode::NonInterleaved);
+    RTC_CHECK_EQ(packetization_mode_, H264PacketizationMode::NonInterleaved);
     NextFragmentPacket(rtp_packet);
   }
   RTC_DCHECK_LE(rtp_packet->payload_size(), max_payload_len_);
   *last_packet = packets_.empty();
   rtp_packet->SetMarker(*last_packet);
   return true;
 }
 
 void RtpPacketizerH264::NextAggregatePacket(RtpPacketToSend* rtp_packet) {
   uint8_t* buffer = rtp_packet->AllocatePayload(max_payload_len_);
@@ skipping 329 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