[EXPERIMENTAL] Generic stereo codec with index header sending merged frames

webrtc/modules/video_coding/frame_object.cc

 /*
  *  Copyright (c) 2016 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/video_coding/frame_object.h"
 #include "webrtc/common_video/h264/h264_common.h"
 #include "webrtc/modules/video_coding/packet_buffer.h"
 #include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/logging.h"
 
 namespace webrtc {
 namespace video_coding {
 
 FrameObject::FrameObject()
     : picture_id(0),
       spatial_layer(0),
       timestamp(0),
       num_references(0),
       inter_layer_predicted(false) {}
 
 RtpFrameObject::RtpFrameObject(PacketBuffer* packet_buffer,
                                uint16_t first_seq_num,
                                uint16_t last_seq_num,
                                size_t frame_size,
                                int times_nacked,
                                int64_t received_time)
     : packet_buffer_(packet_buffer),
       first_seq_num_(first_seq_num),
       last_seq_num_(last_seq_num),
       received_time_(received_time),
       times_nacked_(times_nacked) {
+  LOG(LS_ERROR) << __func__ << first_seq_num << " " << last_seq_num;
   VCMPacket* first_packet = packet_buffer_->GetPacket(first_seq_num);
   RTC_CHECK(first_packet);
 
   // RtpFrameObject members
   frame_type_ = first_packet->frameType;
   codec_type_ = first_packet->codec;
 
+  if (first_packet->video_header.stereoInfo.frame_index == 0 &&
+      codec_type_ == kVideoCodecStereo) {
+    uint16_t index = first_seq_num;
+    while (index != last_seq_num) {
+      VCMPacket* packet = packet_buffer_->GetPacket(first_seq_num);
+      if (packet->video_header.stereoInfo.frame_index > 0)
+        break;
+      index = (index + 1) % packet_buffer_->size_;
+    }
+    last_seq_num_ = index;
+
+    _codecSpecificInfo.stereoInfo.num_frames = 1;
+    RtpFrameObject* alpha =
+        new RtpFrameObject(packet_buffer, index, last_seq_num, frame_size,
+                           times_nacked, received_time);
+    _codecSpecificInfo.stereoInfo.encoded_images[0] = alpha;
+    _codecSpecificInfo.stereoInfo.codec_specific_infos[0] =
+        alpha->CodecSpecific();
+  }
+
   // TODO(philipel): Remove when encoded image is replaced by FrameObject.
   // VCMEncodedFrame members
   CopyCodecSpecific(&first_packet->video_header);
   _completeFrame = true;
   _payloadType = first_packet->payloadType;
   _timeStamp = first_packet->timestamp;
   ntp_time_ms_ = first_packet->ntp_time_ms_;
 
   // Setting frame's playout delays to the same values
   // as of the first packet's.
@@ skipping 134 matching lines @@
 rtc::Optional<RTPVideoTypeHeader> RtpFrameObject::GetCodecHeader() const {
   rtc::CritScope lock(&packet_buffer_->crit_);
   VCMPacket* packet = packet_buffer_->GetPacket(first_seq_num_);
   if (!packet)
     return rtc::Optional<RTPVideoTypeHeader>();
   return rtc::Optional<RTPVideoTypeHeader>(packet->video_header.codecHeader);
 }
 
 }  // namespace video_coding
 }  // namespace webrtc