Add support for VP9 packetization/depacketization.

webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h

+/*
+ *  Copyright (c) 2015 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.
+ */
+
+//
+// This file contains the declaration of the VP9 packetizer class.
+// A packetizer object is created for each encoded video frame. The
+// constructor is called with the payload data and size.
+//
+// After creating the packetizer, the method NextPacket is called
+// repeatedly to get all packets for the frame. The method returns
+// false as long as there are more packets left to fetch.
+//
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP9_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP9_H_
+
+#include <queue>
+#include <string>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class RtpPacketizerVp9 : public RtpPacketizer {
+ public:
+  RtpPacketizerVp9(const RTPVideoHeaderVP9& hdr, size_t max_payload_length);
+
+  virtual ~RtpPacketizerVp9();
+
+  ProtectionType GetProtectionType() override;
+
+  StorageType GetStorageType(uint32_t retransmission_settings) override;
+
+  std::string ToString() override;
+
+  // The payload data must be one encoded VP9 frame.
+  void SetPayloadData(const uint8_t* payload,
+                      size_t payload_size,
+                      const RTPFragmentationHeader* fragmentation) override;
+
+  // Gets the next payload with VP9 payload header.
+  // |buffer| is a pointer to where the output will be written.
+  // |bytes_to_send| is an output variable that will contain number of bytes
+  // written to buffer.
+  // |last_packet| is true for the last packet of the frame, false otherwise
+  // (i.e. call the function again to get the next packet).
+  // Returns true on success, false otherwise.
+  bool NextPacket(uint8_t* buffer,
+                  size_t* bytes_to_send,
+                  bool* last_packet) override;
+
+  typedef struct {
+    size_t payload_start_pos;
+    size_t size;
+    bool layer_begin;
+    bool layer_end;
+  } PacketInfo;
+  typedef std::queue<PacketInfo> PacketInfoQueue;
+
+ private:
+  // Calculates all packet sizes and loads info to packet queue.
+  void GeneratePackets();
+
+  // Writes the payload descriptor header and copies payload to the |buffer|.
+  // |packet_info| determines which part of the payload to write.
+  // |bytes_to_send| contains the number of written bytes to the buffer.
+  // Returns true on success, false otherwise.
+  bool WriteHeaderAndPayload(const PacketInfo& packet_info,
+                             uint8_t* buffer,
+                             size_t* bytes_to_send) const;
+
+  // Writes payload descriptor header to |buffer|.
+  // Returns true on success, false otherwise.
+  bool WriteHeader(const PacketInfo& packet_info,
+                   uint8_t* buffer,
+                   size_t* header_length) const;
+
+  const RTPVideoHeaderVP9 hdr_;
+  const size_t max_payload_length_;  // The max length in bytes of one packet.
+  const uint8_t* payload_;           // The payload data to be packetized.
+  size_t payload_size_;              // The size in bytes of the payload data.
+  PacketInfoQueue packets_;
+
+  DISALLOW_COPY_AND_ASSIGN(RtpPacketizerVp9);
+};
+
+
+class RtpDepacketizerVp9 : public RtpDepacketizer {
+ public:
+  virtual ~RtpDepacketizerVp9() {}
+
+  bool Parse(ParsedPayload* parsed_payload,
+             const uint8_t* payload,
+             size_t payload_length) override;
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP9_H_