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

webrtc/modules/rtp_rtcp/source/rtp_format_video_stereo.h

Index: webrtc/modules/rtp_rtcp/source/rtp_format_video_stereo.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_video_stereo.h b/webrtc/modules/rtp_rtcp/source/rtp_format_video_stereo.h
new file mode 100644
index 0000000000000000000000000000000000000000..3eca8be2614b67363bdf844a8d615c91f558b9bb
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_video_stereo.h
@@ -0,0 +1,73 @@
+/*
+ *  Copyright (c) 2013 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.
+ */
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_STEREO_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_STEREO_H_
+
+#include <string>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+namespace RtpFormatVideoStereo {
+static const uint8_t kFirstPacketBit = 0x02;
+}  // namespace RtpFormatVideoStereo
+
+class RtpPacketizerStereo : public RtpPacketizer {
+ public:
+  RtpPacketizerStereo(size_t max_payload_len,
+                      size_t last_packet_reduction_len,
+                      const RTPVideoTypeHeader* rtp_type_header,
+                      const RTPVideoStereoInfo* stereoInfo);
+
+  virtual ~RtpPacketizerStereo();
+
+  // Returns total number of packets to be generated.
+  size_t SetPayloadData(const uint8_t* payload_data,
+                        size_t payload_size,
+                        const RTPFragmentationHeader* fragmentation) override;
+
+  // Get the next payload with generic payload header.
+  // Write payload and set marker bit of the |packet|.
+  // Returns true on success, false otherwise.
+  bool NextPacket(RtpPacketToSend* packet) override;
+
+  ProtectionType GetProtectionType() override;
+
+  StorageType GetStorageType(uint32_t retransmission_settings) override;
+
+  std::string ToString() override;
+
+ private:
+  const size_t max_payload_len_;
+  const size_t last_packet_reduction_len_;
+  uint8_t frame_index_;
+  const RTPVideoStereoInfo* stereoInfo_;
+  std::vector<std::unique_ptr<RtpPacketizer>> packetizers_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerStereo);
+};
+
+class RtpDepacketizerStereo : public RtpDepacketizer {
+ public:
+  virtual ~RtpDepacketizerStereo() {}
+
+  bool Parse(ParsedPayload* parsed_payload,
+             const uint8_t* payload_data,
+             size_t payload_data_length) override;
+
+ private:
+  RtpDepacketizerVp9 depacketizer_;
+};
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_STEREO_H_