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

webrtc/modules/rtp_rtcp/source/rtp_format_video_stereo.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_format_video_stereo.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_video_stereo.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_video_stereo.cc
new file mode 100644
index 0000000000000000000000000000000000000000..6903e1e5639d86d9a7eb0a93d78b553f965cdcd8
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_video_stereo.cc
@@ -0,0 +1,118 @@
+/*
+ *  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 <string>
+
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_format_video_stereo.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h"
+
+namespace webrtc {
+
+static const size_t kStereoHeaderLength = 1;
+
+RtpPacketizerStereo::RtpPacketizerStereo(
+    size_t max_payload_len,
+    size_t last_packet_reduction_len,
+    const RTPVideoTypeHeader* rtp_type_header,
+    const RTPVideoStereoInfo* stereoInfo)
+    : max_payload_len_(max_payload_len - kStereoHeaderLength),
+      last_packet_reduction_len_(last_packet_reduction_len),
+      frame_index_(0),
+      stereoInfo_(stereoInfo) {
+  packetizers_.emplace_back(RtpPacketizer::Create(
+      stereoInfo->stereoCodecType, max_payload_len_, last_packet_reduction_len_,
+      rtp_type_header, stereoInfo, kVideoFrameDelta));
+  for (int i = 0; i < stereoInfo->num_frames; ++i) {
+    packetizers_.emplace_back(
+        RtpPacketizer::Create(stereoInfo->stereoCodecType, max_payload_len_,
+                              last_packet_reduction_len_,
+                              &(stereoInfo->rtp_video_headers[i]->codecHeader),
+                              stereoInfo, kVideoFrameDelta));
+  }
+}
+
+RtpPacketizerStereo::~RtpPacketizerStereo() {}
+
+size_t RtpPacketizerStereo::SetPayloadData(
+    const uint8_t* payload_data,
+    size_t payload_size,
+    const RTPFragmentationHeader* fragmentation) {
+  size_t num_packets = packetizers_[0]->SetPayloadData(
+      payload_data, payload_size, fragmentation);
+  for (size_t i = 0; i < packetizers_.size() - 1; ++i) {
+    num_packets += packetizers_[i + 1]->SetPayloadData(
+        stereoInfo_->encoded_images[i]->_buffer,
+        stereoInfo_->encoded_images[i]->_length,
+        stereoInfo_->fragmentations[i]);
+  }
+  return num_packets;
+}
+
+bool RtpPacketizerStereo::NextPacket(RtpPacketToSend* packet) {
+  RTC_DCHECK(packet);
+  RTC_DCHECK_LE(frame_index_, static_cast<uint8_t>(packetizers_.size()));
+  const bool rv = packetizers_[frame_index_]->NextPacket(packet);
+  RTC_CHECK(rv);
+
+  const bool last_packet = packet->Marker();
+  if (last_packet &&
+      frame_index_ < static_cast<uint8_t>(packetizers_.size()) - 1) {
+    packet->SetMarker(false);
+  }
+
+  std::unique_ptr<RtpPacketToSend> packet_copy(new RtpPacketToSend(*packet));
+  uint8_t* wrapped_payload =
+      packet->AllocatePayload(kStereoHeaderLength + packet->payload_size());
+  RTC_DCHECK(wrapped_payload);
+  wrapped_payload[0] = frame_index_;
+  auto payload = packet_copy->payload();
+  memcpy(&wrapped_payload[kStereoHeaderLength], payload.data(), payload.size());
+
+  frame_index_ = last_packet ? frame_index_ + 1 : frame_index_;
+  return rv;
+}
+
+ProtectionType RtpPacketizerStereo::GetProtectionType() {
+  return kProtectedPacket;
+}
+
+StorageType RtpPacketizerStereo::GetStorageType(
+    uint32_t retransmission_settings) {
+  return kDontRetransmit;
+}
+
+std::string RtpPacketizerStereo::ToString() {
+  return "RtpPacketizerStereo";
+}
+
+bool RtpDepacketizerStereo::Parse(ParsedPayload* parsed_payload,
+                                  const uint8_t* payload_data,
+                                  size_t payload_data_length) {
+  assert(parsed_payload != NULL);
+  if (payload_data_length == 0) {
+    LOG(LS_ERROR) << "Empty payload.";
+    return false;
+  }
+
+  uint8_t frame_index = *payload_data++;
+  --payload_data_length;
+
+  const bool rv =
+      depacketizer_.Parse(parsed_payload, payload_data, payload_data_length);
+  parsed_payload->type.Video.stereoInfo.frame_index = frame_index;
+  RTC_DCHECK(rv);
+  if (frame_index > 0)
+    parsed_payload->type.Video.is_first_packet_in_frame = false;
+  parsed_payload->type.Video.codec = kRtpVideoStereo;
+  return rv;
+}
+}  // namespace webrtc