Add H.264 packetization.

This also includes:
- Creating new packetizer and depacketizer interfaces.
- Moved VP8 packetization was H264 packetization and depacketization to these interfaces. This is a work in progress and should be continued to get this 100% generic. This also required changing the return type for RtpFormatVp8::NextPacket(), which now returns bool instead of the index of the first partition.
- Created a Create() factory method for packetizers and depacketizers.

R=niklas.enbom@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6804 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/rtp_rtcp/source/rtp_format.h

@@ -0,0 +1,54 @@
+/*
+ *  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.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+
+namespace webrtc {
+
+class RtpPacketizer {
+ public:
+  static RtpPacketizer* Create(RtpVideoCodecTypes type, size_t max_payload_len);
+
+  virtual ~RtpPacketizer() {}
+
+  virtual void SetPayloadData(const uint8_t* payload_data,
+                              size_t payload_size,
+                              const RTPFragmentationHeader* fragmentation) = 0;
+
+  // Get the next payload with 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. The parameter 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 or false if there was no payload to packetize.
+  virtual bool NextPacket(uint8_t* buffer,
+                          size_t* bytes_to_send,
+                          bool* last_packet) = 0;
+};
+
+class RtpDepacketizer {
+ public:
+  static RtpDepacketizer* Create(RtpVideoCodecTypes type,
+                                 RtpData* const callback);
+
+  virtual ~RtpDepacketizer() {}
+
+  virtual bool Parse(WebRtcRTPHeader* rtp_header,
+                     const uint8_t* payload_data,
+                     size_t payload_data_length) = 0;
+};
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_