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

modules/rtp_rtcp/source/rtp_format_video_stereo.cc

Index: modules/rtp_rtcp/source/rtp_format_video_stereo.cc
diff --git a/modules/rtp_rtcp/source/rtp_format_video_stereo.cc b/modules/rtp_rtcp/source/rtp_format_video_stereo.cc
new file mode 100644
index 0000000000000000000000000000000000000000..451a92929dc62814911fd04732fba8949117c992
--- /dev/null
+++ b/modules/rtp_rtcp/source/rtp_format_video_stereo.cc
@@ -0,0 +1,116 @@
+/*
+ *  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 "modules/include/module_common_types.h"
+#include "modules/rtp_rtcp/source/rtp_format_video_stereo.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+static const size_t kStereoHeaderMarkerLength = 1;
+static const size_t kStereoHeaderLength = sizeof(RTPVideoStereoInfo);
+
+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 - kStereoHeaderMarkerLength -
+                       kStereoHeaderLength),
+      last_packet_reduction_len_(last_packet_reduction_len),
+      packetizer_(RtpPacketizer::Create(stereoInfo->stereoCodecType,
+                                        max_payload_len_,
+                                        last_packet_reduction_len_,
+                                        rtp_type_header,
+                                        stereoInfo,
+                                        kVideoFrameDelta)),
+      stereoInfo_(stereoInfo) {}
+
+RtpPacketizerStereo::~RtpPacketizerStereo() {}
+
+size_t RtpPacketizerStereo::SetPayloadData(
+    const uint8_t* payload_data,
+    size_t payload_size,
+    const RTPFragmentationHeader* fragmentation) {
+  header_marker_ = RtpFormatVideoStereo::kFirstPacketBit;
+  return packetizer_->SetPayloadData(payload_data, payload_size, fragmentation);
+}
+
+bool RtpPacketizerStereo::NextPacket(RtpPacketToSend* packet) {
+  RTC_DCHECK(packet);
+  const bool rv = packetizer_->NextPacket(packet);
+  RTC_CHECK(rv);
+
+  const bool first_packet =
+      header_marker_ == RtpFormatVideoStereo::kFirstPacketBit;
+  size_t header_length = first_packet
+                             ? kStereoHeaderMarkerLength + kStereoHeaderLength
+                             : kStereoHeaderMarkerLength;
+
+  std::unique_ptr<RtpPacketToSend> copied_packet(new RtpPacketToSend(*packet));
+  uint8_t* wrapped_payload =
+      packet->AllocatePayload(header_length + packet->payload_size());
+  RTC_DCHECK(wrapped_payload);
+  wrapped_payload[0] = header_marker_;
+  header_marker_ &= ~RtpFormatVideoStereo::kFirstPacketBit;
+  if (first_packet) {
+    memcpy(&wrapped_payload[kStereoHeaderMarkerLength], stereoInfo_,
+           kStereoHeaderLength);
+  }
+  auto payload = copied_packet->payload();
+  memcpy(&wrapped_payload[header_length], payload.data(), payload.size());
+  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 marker_header = *payload_data++;
+  --payload_data_length;
+  const bool first_packet =
+      (marker_header & RtpFormatVideoStereo::kFirstPacketBit) != 0;
+
+  if (first_packet) {
+    memcpy(&parsed_payload->type.Video.stereoInfo, payload_data,
+           kStereoHeaderLength);
+    payload_data += kStereoHeaderLength;
+    payload_data_length -= kStereoHeaderLength;
+  }
+  const bool rv =
+      depacketizer_.Parse(parsed_payload, payload_data, payload_data_length);
+  RTC_DCHECK(rv);
+  RTC_DCHECK_EQ(parsed_payload->type.Video.is_first_packet_in_frame,
+                first_packet);
+  parsed_payload->type.Video.codec = kRtpVideoStereo;
+  return rv;
+}
+}  // namespace webrtc