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
diff --git a/webrtc/modules/interface/module_common_types.h b/webrtc/modules/interface/module_common_types.h
index 398808f..60bb7ab 100644
--- a/webrtc/modules/interface/module_common_types.h
+++ b/webrtc/modules/interface/module_common_types.h
@@ -61,8 +61,8 @@
 };
 
 struct RTPVideoHeaderH264 {
-  uint8_t nalu_header;
-  bool    single_nalu;
+  bool stap_a;
+  bool single_nalu;
 };
 
 union RTPVideoTypeHeader {
diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp
index 2a3ba74..10a1860 100644
--- a/webrtc/modules/modules.gyp
+++ b/webrtc/modules/modules.gyp
@@ -206,6 +206,7 @@
             'rtp_rtcp/source/rtcp_receiver_unittest.cc',
             'rtp_rtcp/source/rtcp_sender_unittest.cc',
             'rtp_rtcp/source/rtp_fec_unittest.cc',
+            'rtp_rtcp/source/rtp_format_h264_unittest.cc',
             'rtp_rtcp/source/rtp_format_vp8_unittest.cc',
             'rtp_rtcp/source/rtp_format_vp8_test_helper.cc',
             'rtp_rtcp/source/rtp_format_vp8_test_helper.h',
diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn
index e9a2e8d..a7499bc 100644
--- a/webrtc/modules/rtp_rtcp/BUILD.gn
+++ b/webrtc/modules/rtp_rtcp/BUILD.gn
@@ -79,6 +79,10 @@
     "source/rtp_sender_video.cc",
     "source/rtp_sender_video.h",
     "source/video_codec_information.h",
+    'source/rtp_format.cc',
+    'source/rtp_format.h',
+    'source/rtp_format_h264.cc',
+    'source/rtp_format_h264.h',
     "source/rtp_format_vp8.cc",
     "source/rtp_format_vp8.h",
     "source/rtp_format_video_generic.h",
diff --git a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
index c954aa2..2caa631 100644
--- a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -19,6 +19,17 @@
 
 namespace webrtc {
 
+class MockRtpData : public RtpData {
+ public:
+  MOCK_METHOD3(OnReceivedPayloadData,
+               int32_t(const uint8_t* payloadData,
+                       const uint16_t payloadSize,
+                       const WebRtcRTPHeader* rtpHeader));
+
+  MOCK_METHOD2(OnRecoveredPacket,
+               bool(const uint8_t* packet, int packet_length));
+};
+
 class MockRtpRtcp : public RtpRtcp {
  public:
   MOCK_METHOD1(ChangeUniqueId,
diff --git a/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc b/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc
index 0b12449..2a3b155 100644
--- a/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc
@@ -15,6 +15,7 @@
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/modules/rtp_rtcp/interface/fec_receiver.h"
+#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/source/fec_test_helper.h"
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
 
@@ -25,17 +26,6 @@
 
 namespace webrtc {
 
-class MockRtpData : public RtpData {
- public:
-  MOCK_METHOD3(OnReceivedPayloadData,
-      int32_t(const uint8_t* payloadData,
-              const uint16_t payloadSize,
-              const WebRtcRTPHeader* rtpHeader));
-
-  MOCK_METHOD2(OnRecoveredPacket,
-      bool(const uint8_t* packet, int packet_length));
-};
-
 class ReceiverFecTest : public ::testing::Test {
  protected:
   virtual void SetUp() {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format.cc b/webrtc/modules/rtp_rtcp/source/rtp_format.cc
new file mode 100644
index 0000000..da1f2df
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format.cc
@@ -0,0 +1,41 @@
+/*
+ *  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 "webrtc/modules/rtp_rtcp/source/rtp_format.h"
+
+#include "webrtc/modules/rtp_rtcp/source/rtp_format_h264.h"
+
+namespace webrtc {
+RtpPacketizer* RtpPacketizer::Create(RtpVideoCodecTypes type,
+                                     size_t max_payload_len) {
+  switch (type) {
+    case kRtpVideoH264:
+      return new RtpPacketizerH264(max_payload_len);
+    case kRtpVideoNone:
+    case kRtpVideoGeneric:
+    case kRtpVideoVp8:
+      assert(false);
+  }
+  return NULL;
+}
+
+RtpDepacketizer* RtpDepacketizer::Create(RtpVideoCodecTypes type,
+                                         RtpData* const callback) {
+  switch (type) {
+    case kRtpVideoH264:
+      return new RtpDepacketizerH264(callback);
+    case kRtpVideoNone:
+    case kRtpVideoGeneric:
+    case kRtpVideoVp8:
+      assert(false);
+  }
+  return NULL;
+}
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format.h b/webrtc/modules/rtp_rtcp/source/rtp_format.h
new file mode 100644
index 0000000..d7dba48
--- /dev/null
+++ b/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_
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
new file mode 100644
index 0000000..d986eae
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
@@ -0,0 +1,293 @@
+/*
+ *  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.h>
+
+#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_format_h264.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+
+namespace webrtc {
+namespace {
+
+enum Nalu {
+  kSlice = 1,
+  kIdr = 5,
+  kSei = 6,
+  kSps = 7,
+  kPps = 8,
+  kStapA = 24,
+  kFuA = 28
+};
+
+static const size_t kNalHeaderSize = 1;
+static const size_t kFuAHeaderSize = 2;
+
+// Bit masks for FU (A and B) indicators.
+enum NalDefs { kFBit = 0x80, kNriMask = 0x60, kTypeMask = 0x1F };
+
+// Bit masks for FU (A and B) headers.
+enum FuDefs { kSBit = 0x80, kEBit = 0x40, kRBit = 0x20 };
+
+void ParseSingleNalu(WebRtcRTPHeader* rtp_header,
+                     const uint8_t* payload_data,
+                     size_t payload_data_length) {
+  rtp_header->type.Video.codec = kRtpVideoH264;
+  rtp_header->type.Video.isFirstPacket = true;
+  RTPVideoHeaderH264* h264_header = &rtp_header->type.Video.codecHeader.H264;
+  h264_header->single_nalu = true;
+  h264_header->stap_a = false;
+
+  uint8_t nal_type = payload_data[0] & NalDefs::kTypeMask;
+  if (nal_type == Nalu::kStapA) {
+    nal_type = payload_data[3] & NalDefs::kTypeMask;
+    h264_header->stap_a = true;
+  }
+
+  switch (nal_type) {
+    case Nalu::kSps:
+    case Nalu::kPps:
+    case Nalu::kIdr:
+      rtp_header->frameType = kVideoFrameKey;
+      break;
+    default:
+      rtp_header->frameType = kVideoFrameDelta;
+      break;
+  }
+}
+
+void ParseFuaNalu(WebRtcRTPHeader* rtp_header,
+                  const uint8_t* payload_data,
+                  size_t payload_data_length,
+                  size_t* offset) {
+  uint8_t fnri = payload_data[0] & (NalDefs::kFBit | NalDefs::kNriMask);
+  uint8_t original_nal_type = payload_data[1] & NalDefs::kTypeMask;
+  bool first_fragment = (payload_data[1] & FuDefs::kSBit) > 0;
+
+  uint8_t original_nal_header = fnri | original_nal_type;
+  if (first_fragment) {
+    *offset = kNalHeaderSize;
+    uint8_t* payload = const_cast<uint8_t*>(payload_data + *offset);
+    payload[0] = original_nal_header;
+  } else {
+    *offset = kFuAHeaderSize;
+  }
+
+  if (original_nal_type == Nalu::kIdr) {
+    rtp_header->frameType = kVideoFrameKey;
+  } else {
+    rtp_header->frameType = kVideoFrameDelta;
+  }
+  rtp_header->type.Video.codec = kRtpVideoH264;
+  rtp_header->type.Video.isFirstPacket = first_fragment;
+  RTPVideoHeaderH264* h264_header = &rtp_header->type.Video.codecHeader.H264;
+  h264_header->single_nalu = false;
+  h264_header->stap_a = false;
+}
+}  // namespace
+
+RtpPacketizerH264::RtpPacketizerH264(size_t max_payload_len)
+    : payload_data_(NULL), payload_size_(0), max_payload_len_(max_payload_len) {
+}
+
+RtpPacketizerH264::~RtpPacketizerH264() {
+}
+
+void RtpPacketizerH264::SetPayloadData(
+    const uint8_t* payload_data,
+    size_t payload_size,
+    const RTPFragmentationHeader* fragmentation) {
+  assert(packets_.empty());
+  assert(fragmentation);
+  payload_data_ = payload_data;
+  payload_size_ = payload_size;
+  fragmentation_.CopyFrom(*fragmentation);
+  GeneratePackets();
+}
+
+void RtpPacketizerH264::GeneratePackets() {
+  for (size_t i = 0; i < fragmentation_.fragmentationVectorSize;) {
+    size_t fragment_offset = fragmentation_.fragmentationOffset[i];
+    size_t fragment_length = fragmentation_.fragmentationLength[i];
+    if (fragment_length > max_payload_len_) {
+      PacketizeFuA(fragment_offset, fragment_length);
+      ++i;
+    } else {
+      i = PacketizeStapA(i, fragment_offset, fragment_length);
+    }
+  }
+}
+
+void RtpPacketizerH264::PacketizeFuA(size_t fragment_offset,
+                                     size_t fragment_length) {
+  // Fragment payload into packets (FU-A).
+  // Strip out the original header and leave room for the FU-A header.
+  fragment_length -= kNalHeaderSize;
+  size_t offset = fragment_offset + kNalHeaderSize;
+  size_t bytes_available = max_payload_len_ - kFuAHeaderSize;
+  size_t fragments =
+      (fragment_length + (bytes_available - 1)) / bytes_available;
+  size_t avg_size = (fragment_length + fragments - 1) / fragments;
+  while (fragment_length > 0) {
+    size_t packet_length = avg_size;
+    if (fragment_length < avg_size)
+      packet_length = fragment_length;
+    uint8_t header = payload_data_[fragment_offset];
+    packets_.push(Packet(offset,
+                         packet_length,
+                         offset - kNalHeaderSize == fragment_offset,
+                         fragment_length == packet_length,
+                         false,
+                         header));
+    offset += packet_length;
+    fragment_length -= packet_length;
+  }
+}
+
+int RtpPacketizerH264::PacketizeStapA(size_t fragment_index,
+                                      size_t fragment_offset,
+                                      size_t fragment_length) {
+  // Aggregate fragments into one packet (STAP-A).
+  size_t payload_size_left = max_payload_len_;
+  int aggregated_fragments = 0;
+  assert(payload_size_left >= fragment_length);
+  while (payload_size_left >= fragment_length) {
+    if (fragment_length > 0) {
+      assert(fragment_length > 0);
+      uint8_t header = payload_data_[fragment_offset];
+      packets_.push(Packet(fragment_offset,
+                           fragment_length,
+                           aggregated_fragments == 0,
+                           false,
+                           true,
+                           header));
+      // If we are going to try to aggregate more fragments into this packet
+      // we need to add the STAP-A NALU header.
+      if (aggregated_fragments == 0)
+        payload_size_left -= kNalHeaderSize;
+      payload_size_left -= fragment_length;
+      ++aggregated_fragments;
+    }
+    // Next fragment.
+    ++fragment_index;
+    if (fragment_index == fragmentation_.fragmentationVectorSize)
+      break;
+    fragment_offset = fragmentation_.fragmentationOffset[fragment_index];
+    fragment_length = fragmentation_.fragmentationLength[fragment_index];
+  }
+  packets_.back().last_fragment = true;
+  return fragment_index;
+}
+
+bool RtpPacketizerH264::NextPacket(uint8_t* buffer,
+                                   size_t* bytes_to_send,
+                                   bool* last_packet) {
+  *bytes_to_send = 0;
+  if (packets_.empty()) {
+    *bytes_to_send = 0;
+    *last_packet = true;
+    return false;
+  }
+
+  Packet packet = packets_.front();
+
+  if (packet.first_fragment && packet.last_fragment) {
+    // Single NAL unit packet.
+    *bytes_to_send = packet.size;
+    memcpy(buffer, &payload_data_[packet.offset], packet.size);
+    packets_.pop();
+  } else if (packet.aggregated) {
+    NextAggregatePacket(buffer, bytes_to_send);
+  } else {
+    NextFragmentPacket(buffer, bytes_to_send);
+  }
+  *last_packet = packets_.empty();
+  assert(*bytes_to_send <= max_payload_len_);
+  return true;
+}
+
+void RtpPacketizerH264::NextAggregatePacket(uint8_t* buffer,
+                                            size_t* bytes_to_send) {
+  Packet packet = packets_.front();
+  assert(packet.first_fragment);
+  // STAP-A NALU header.
+  buffer[0] = (packet.header & (kFBit | kNriMask)) | kStapA;
+  int index = kNalHeaderSize;
+  *bytes_to_send += kNalHeaderSize;
+  while (packet.aggregated) {
+    // Add NAL unit length field.
+    RtpUtility::AssignUWord16ToBuffer(&buffer[index], packet.size);
+    index += 2;
+    *bytes_to_send += 2;
+    // Add NAL unit.
+    memcpy(&buffer[index], &payload_data_[packet.offset], packet.size);
+    index += packet.size;
+    *bytes_to_send += packet.size;
+    packets_.pop();
+    if (packet.last_fragment)
+      break;
+    packet = packets_.front();
+  }
+  assert(packet.last_fragment);
+}
+
+void RtpPacketizerH264::NextFragmentPacket(uint8_t* buffer,
+                                           size_t* bytes_to_send) {
+  Packet packet = packets_.front();
+  // NAL unit fragmented over multiple packets (FU-A).
+  // We do not send original NALU header, so it will be replaced by the
+  // FU indicator header of the first packet.
+  uint8_t fu_indicator = (packet.header & (kFBit | kNriMask)) | kFuA;
+  uint8_t fu_header = 0;
+
+  // S | E | R | 5 bit type.
+  fu_header |= (packet.first_fragment ? kSBit : 0);
+  fu_header |= (packet.last_fragment ? kEBit : 0);
+  uint8_t type = packet.header & kTypeMask;
+  fu_header |= type;
+  buffer[0] = fu_indicator;
+  buffer[1] = fu_header;
+
+  if (packet.last_fragment) {
+    *bytes_to_send = packet.size + kFuAHeaderSize;
+    memcpy(buffer + kFuAHeaderSize, &payload_data_[packet.offset], packet.size);
+  } else {
+    *bytes_to_send = packet.size + kFuAHeaderSize;
+    memcpy(buffer + kFuAHeaderSize, &payload_data_[packet.offset], packet.size);
+  }
+  packets_.pop();
+}
+
+RtpDepacketizerH264::RtpDepacketizerH264(RtpData* const callback)
+    : callback_(callback) {
+}
+
+bool RtpDepacketizerH264::Parse(WebRtcRTPHeader* rtp_header,
+                                const uint8_t* payload_data,
+                                size_t payload_data_length) {
+  uint8_t nal_type = payload_data[0] & NalDefs::kTypeMask;
+  size_t offset = 0;
+  if (nal_type == Nalu::kFuA) {
+    // Fragmented NAL units (FU-A).
+    ParseFuaNalu(rtp_header, payload_data, payload_data_length, &offset);
+  } else {
+    // We handle STAP-A and single NALU's the same way here. The jitter buffer
+    // will depacketize the STAP-A into NAL units later.
+    ParseSingleNalu(rtp_header, payload_data, payload_data_length);
+  }
+  if (callback_->OnReceivedPayloadData(payload_data + offset,
+                                       payload_data_length - offset,
+                                       rtp_header) != 0) {
+    return false;
+  }
+  return true;
+}
+
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h
new file mode 100644
index 0000000..773fa18
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h
@@ -0,0 +1,102 @@
+/*
+ *  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_H264_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_
+
+#include <queue>
+
+#include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
+
+namespace webrtc {
+
+class RtpPacketizerH264 : public RtpPacketizer {
+ public:
+  // Initialize with payload from encoder.
+  // The payload_data must be exactly one encoded H264 frame.
+  explicit RtpPacketizerH264(size_t max_payload_len);
+
+  virtual ~RtpPacketizerH264();
+
+  virtual void SetPayloadData(
+      const uint8_t* payload_data,
+      size_t payload_size,
+      const RTPFragmentationHeader* fragmentation) OVERRIDE;
+
+  // Get the next payload with H264 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) OVERRIDE;
+
+ private:
+  struct Packet {
+    Packet(size_t offset,
+           size_t size,
+           bool first_fragment,
+           bool last_fragment,
+           bool aggregated,
+           uint8_t header)
+        : offset(offset),
+          size(size),
+          first_fragment(first_fragment),
+          last_fragment(last_fragment),
+          aggregated(aggregated),
+          header(header) {}
+
+    size_t offset;
+    size_t size;
+    bool first_fragment;
+    bool last_fragment;
+    bool aggregated;
+    uint8_t header;
+  };
+  typedef std::queue<Packet> PacketQueue;
+
+  void GeneratePackets();
+  void PacketizeFuA(size_t fragment_offset, size_t fragment_length);
+  int PacketizeStapA(size_t fragment_index,
+                     size_t fragment_offset,
+                     size_t fragment_length);
+  void NextAggregatePacket(uint8_t* buffer, size_t* bytes_to_send);
+  void NextFragmentPacket(uint8_t* buffer, size_t* bytes_to_send);
+
+  const uint8_t* payload_data_;
+  size_t payload_size_;
+  const size_t max_payload_len_;
+  RTPFragmentationHeader fragmentation_;
+  PacketQueue packets_;
+
+  DISALLOW_COPY_AND_ASSIGN(RtpPacketizerH264);
+};
+
+// Depacketizer for H264.
+class RtpDepacketizerH264 : public RtpDepacketizer {
+ public:
+  explicit RtpDepacketizerH264(RtpData* const callback);
+
+  virtual ~RtpDepacketizerH264() {}
+
+  virtual bool Parse(WebRtcRTPHeader* rtp_header,
+                     const uint8_t* payload_data,
+                     size_t payload_data_length) OVERRIDE;
+
+ private:
+  RtpData* const callback_;
+
+  DISALLOW_COPY_AND_ASSIGN(RtpDepacketizerH264);
+};
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
new file mode 100644
index 0000000..61c7f4c
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
@@ -0,0 +1,412 @@
+/*
+ *  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 <vector>
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+
+using ::testing::_;
+using ::testing::Args;
+using ::testing::ElementsAreArray;
+using ::testing::Return;
+using ::testing::SaveArgPointee;
+
+namespace webrtc {
+namespace {
+const size_t kMaxPayloadSize = 1200;
+const size_t kLengthFieldLength = 2;
+
+enum Nalu {
+  kSlice = 1,
+  kIdr = 5,
+  kSei = 6,
+  kSps = 7,
+  kPps = 8,
+  kStapA = 24,
+  kFuA = 28
+};
+
+static const size_t kNalHeaderSize = 1;
+static const size_t kFuAHeaderSize = 2;
+
+// Bit masks for FU (A and B) indicators.
+enum NalDefs { kFBit = 0x80, kNriMask = 0x60, kTypeMask = 0x1F };
+
+// Bit masks for FU (A and B) headers.
+enum FuDefs { kSBit = 0x80, kEBit = 0x40, kRBit = 0x20 };
+
+void VerifyFua(size_t fua_index,
+               const uint8_t* expected_payload,
+               int offset,
+               const uint8_t* packet,
+               size_t length,
+               const std::vector<size_t>& expected_sizes) {
+  ASSERT_EQ(expected_sizes[fua_index] + kFuAHeaderSize, length)
+      << "FUA index: " << fua_index;
+  const uint8_t kFuIndicator = 0x1C;  // F=0, NRI=0, Type=28.
+  EXPECT_EQ(kFuIndicator, packet[0]) << "FUA index: " << fua_index;
+  bool should_be_last_fua = (fua_index == expected_sizes.size() - 1);
+  uint8_t fu_header = 0;
+  if (fua_index == 0)
+    fu_header = 0x85;  // S=1, E=0, R=0, Type=5.
+  else if (should_be_last_fua)
+    fu_header = 0x45;  // S=0, E=1, R=0, Type=5.
+  else
+    fu_header = 0x05;  // S=0, E=0, R=0, Type=5.
+  EXPECT_EQ(fu_header, packet[1]) << "FUA index: " << fua_index;
+  std::vector<uint8_t> expected_packet_payload(
+      &expected_payload[offset],
+      &expected_payload[offset + expected_sizes[fua_index]]);
+  EXPECT_THAT(
+      expected_packet_payload,
+      ::testing::ElementsAreArray(&packet[2], expected_sizes[fua_index]))
+      << "FUA index: " << fua_index;
+}
+
+void TestFua(size_t frame_size,
+             size_t max_payload_size,
+             const std::vector<size_t>& expected_sizes) {
+  scoped_ptr<uint8_t[]> frame;
+  frame.reset(new uint8_t[frame_size]);
+  frame[0] = 0x05;  // F=0, NRI=0, Type=5.
+  for (size_t i = 0; i < frame_size - kNalHeaderSize; ++i) {
+    frame[i + kNalHeaderSize] = i;
+  }
+  RTPFragmentationHeader fragmentation;
+  fragmentation.VerifyAndAllocateFragmentationHeader(1);
+  fragmentation.fragmentationOffset[0] = 0;
+  fragmentation.fragmentationLength[0] = frame_size;
+  scoped_ptr<RtpPacketizer> packetizer(
+      RtpPacketizer::Create(kRtpVideoH264, max_payload_size));
+  packetizer->SetPayloadData(frame.get(), frame_size, &fragmentation);
+
+  scoped_ptr<uint8_t[]> packet(new uint8_t[max_payload_size]);
+  size_t length = 0;
+  bool last = false;
+  size_t offset = kNalHeaderSize;
+  for (size_t i = 0; i < expected_sizes.size(); ++i) {
+    ASSERT_TRUE(packetizer->NextPacket(packet.get(), &length, &last));
+    VerifyFua(i, frame.get(), offset, packet.get(), length, expected_sizes);
+    EXPECT_EQ(i == expected_sizes.size() - 1, last) << "FUA index: " << i;
+    offset += expected_sizes[i];
+  }
+
+  EXPECT_FALSE(packetizer->NextPacket(packet.get(), &length, &last));
+}
+
+size_t GetExpectedNaluOffset(const RTPFragmentationHeader& fragmentation,
+                             size_t start_index,
+                             size_t nalu_index) {
+  assert(nalu_index < fragmentation.fragmentationVectorSize);
+  size_t expected_nalu_offset = kNalHeaderSize;  // STAP-A header.
+  for (size_t i = start_index; i < nalu_index; ++i) {
+    expected_nalu_offset +=
+        kLengthFieldLength + fragmentation.fragmentationLength[i];
+  }
+  return expected_nalu_offset;
+}
+
+void VerifyStapAPayload(const RTPFragmentationHeader& fragmentation,
+                        size_t first_stapa_index,
+                        size_t nalu_index,
+                        const uint8_t* frame,
+                        size_t frame_length,
+                        const uint8_t* packet,
+                        size_t packet_length) {
+  size_t expected_payload_offset =
+      GetExpectedNaluOffset(fragmentation, first_stapa_index, nalu_index) +
+      kLengthFieldLength;
+  size_t offset = fragmentation.fragmentationOffset[nalu_index];
+  const uint8_t* expected_payload = &frame[offset];
+  size_t expected_payload_length =
+      fragmentation.fragmentationLength[nalu_index];
+  ASSERT_LE(offset + expected_payload_length, frame_length);
+  ASSERT_LE(expected_payload_offset + expected_payload_length, packet_length);
+  std::vector<uint8_t> expected_payload_vector(
+      expected_payload, &expected_payload[expected_payload_length]);
+  EXPECT_THAT(expected_payload_vector,
+              ::testing::ElementsAreArray(&packet[expected_payload_offset],
+                                          expected_payload_length));
+}
+}  // namespace
+
+TEST(RtpPacketizerH264Test, TestSingleNalu) {
+  const uint8_t frame[2] = {0x05, 0xFF};  // F=0, NRI=0, Type=5.
+  RTPFragmentationHeader fragmentation;
+  fragmentation.VerifyAndAllocateFragmentationHeader(1);
+  fragmentation.fragmentationOffset[0] = 0;
+  fragmentation.fragmentationLength[0] = sizeof(frame);
+  scoped_ptr<RtpPacketizer> packetizer(
+      RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize));
+  packetizer->SetPayloadData(frame, sizeof(frame), &fragmentation);
+  uint8_t packet[kMaxPayloadSize] = {0};
+  size_t length = 0;
+  bool last = false;
+  ASSERT_TRUE(packetizer->NextPacket(packet, &length, &last));
+  EXPECT_EQ(2u, length);
+  EXPECT_TRUE(last);
+  EXPECT_EQ(frame[0], packet[0]);
+  EXPECT_EQ(frame[1], packet[1]);
+  EXPECT_FALSE(packetizer->NextPacket(packet, &length, &last));
+}
+
+TEST(RtpPacketizerH264Test, TestStapA) {
+  const size_t kFrameSize = kMaxPayloadSize - 100;
+  uint8_t frame[kFrameSize] = {0x07, 0xFF,  // F=0, NRI=0, Type=7.
+                               0x08, 0xFF,  // F=0, NRI=0, Type=8.
+                               0x05};       // F=0, NRI=0, Type=5.
+  const size_t kPayloadOffset = 5;
+  for (size_t i = 0; i < kFrameSize - kPayloadOffset; ++i)
+    frame[i + kPayloadOffset] = i;
+  RTPFragmentationHeader fragmentation;
+  fragmentation.VerifyAndAllocateFragmentationHeader(3);
+  fragmentation.fragmentationOffset[0] = 0;
+  fragmentation.fragmentationLength[0] = 2;
+  fragmentation.fragmentationOffset[1] = 2;
+  fragmentation.fragmentationLength[1] = 2;
+  fragmentation.fragmentationOffset[2] = 4;
+  fragmentation.fragmentationLength[2] =
+      kNalHeaderSize + kFrameSize - kPayloadOffset;
+  scoped_ptr<RtpPacketizer> packetizer(
+      RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize));
+  packetizer->SetPayloadData(frame, kFrameSize, &fragmentation);
+
+  uint8_t packet[kMaxPayloadSize] = {0};
+  size_t length = 0;
+  bool last = false;
+  ASSERT_TRUE(packetizer->NextPacket(packet, &length, &last));
+  size_t expected_packet_size =
+      kNalHeaderSize + 3 * kLengthFieldLength + kFrameSize;
+  ASSERT_EQ(expected_packet_size, length);
+  EXPECT_TRUE(last);
+  for (size_t i = 0; i < fragmentation.fragmentationVectorSize; ++i)
+    VerifyStapAPayload(fragmentation, 0, i, frame, kFrameSize, packet, length);
+
+  EXPECT_FALSE(packetizer->NextPacket(packet, &length, &last));
+}
+
+TEST(RtpPacketizerH264Test, TestMixedStapA_FUA) {
+  const size_t kFuaNaluSize = 2 * (kMaxPayloadSize - 100);
+  const size_t kStapANaluSize = 100;
+  RTPFragmentationHeader fragmentation;
+  fragmentation.VerifyAndAllocateFragmentationHeader(3);
+  fragmentation.fragmentationOffset[0] = 0;
+  fragmentation.fragmentationLength[0] = kFuaNaluSize;
+  fragmentation.fragmentationOffset[1] = kFuaNaluSize;
+  fragmentation.fragmentationLength[1] = kStapANaluSize;
+  fragmentation.fragmentationOffset[2] = kFuaNaluSize + kStapANaluSize;
+  fragmentation.fragmentationLength[2] = kStapANaluSize;
+  const size_t kFrameSize = kFuaNaluSize + 2 * kStapANaluSize;
+  uint8_t frame[kFrameSize];
+  size_t nalu_offset = 0;
+  for (size_t i = 0; i < fragmentation.fragmentationVectorSize; ++i) {
+    nalu_offset = fragmentation.fragmentationOffset[i];
+    frame[nalu_offset] = 0x05;  // F=0, NRI=0, Type=5.
+    for (size_t j = 1; j < fragmentation.fragmentationLength[i]; ++j) {
+      frame[nalu_offset + j] = i + j;
+    }
+  }
+  scoped_ptr<RtpPacketizer> packetizer(
+      RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize));
+  packetizer->SetPayloadData(frame, kFrameSize, &fragmentation);
+
+  // First expecting two FU-A packets.
+  std::vector<size_t> fua_sizes;
+  fua_sizes.push_back(1100);
+  fua_sizes.push_back(1099);
+  uint8_t packet[kMaxPayloadSize] = {0};
+  size_t length = 0;
+  bool last = false;
+  int fua_offset = kNalHeaderSize;
+  for (size_t i = 0; i < 2; ++i) {
+    ASSERT_TRUE(packetizer->NextPacket(packet, &length, &last));
+    VerifyFua(i, frame, fua_offset, packet, length, fua_sizes);
+    EXPECT_FALSE(last);
+    fua_offset += fua_sizes[i];
+  }
+  // Then expecting one STAP-A packet with two nal units.
+  ASSERT_TRUE(packetizer->NextPacket(packet, &length, &last));
+  size_t expected_packet_size =
+      kNalHeaderSize + 2 * kLengthFieldLength + 2 * kStapANaluSize;
+  ASSERT_EQ(expected_packet_size, length);
+  EXPECT_TRUE(last);
+  for (size_t i = 1; i < fragmentation.fragmentationVectorSize; ++i)
+    VerifyStapAPayload(fragmentation, 1, i, frame, kFrameSize, packet, length);
+
+  EXPECT_FALSE(packetizer->NextPacket(packet, &length, &last));
+}
+
+TEST(RtpPacketizerH264Test, TestFUAOddSize) {
+  const size_t kExpectedPayloadSizes[2] = {600, 600};
+  TestFua(
+      kMaxPayloadSize + 1,
+      kMaxPayloadSize,
+      std::vector<size_t>(kExpectedPayloadSizes,
+                          kExpectedPayloadSizes +
+                              sizeof(kExpectedPayloadSizes) / sizeof(size_t)));
+}
+
+TEST(RtpPacketizerH264Test, TestFUAEvenSize) {
+  const size_t kExpectedPayloadSizes[2] = {601, 600};
+  TestFua(
+      kMaxPayloadSize + 2,
+      kMaxPayloadSize,
+      std::vector<size_t>(kExpectedPayloadSizes,
+                          kExpectedPayloadSizes +
+                              sizeof(kExpectedPayloadSizes) / sizeof(size_t)));
+}
+
+TEST(RtpPacketizerH264Test, TestFUARounding) {
+  const size_t kExpectedPayloadSizes[8] = {1266, 1266, 1266, 1266,
+                                           1266, 1266, 1266, 1261};
+  TestFua(
+      10124,
+      1448,
+      std::vector<size_t>(kExpectedPayloadSizes,
+                          kExpectedPayloadSizes +
+                              sizeof(kExpectedPayloadSizes) / sizeof(size_t)));
+}
+
+TEST(RtpPacketizerH264Test, TestFUABig) {
+  const size_t kExpectedPayloadSizes[10] = {1198, 1198, 1198, 1198, 1198,
+                                            1198, 1198, 1198, 1198, 1198};
+  // Generate 10 full sized packets, leave room for FU-A headers minus the NALU
+  // header.
+  TestFua(
+      10 * (kMaxPayloadSize - kFuAHeaderSize) + kNalHeaderSize,
+      kMaxPayloadSize,
+      std::vector<size_t>(kExpectedPayloadSizes,
+                          kExpectedPayloadSizes +
+                              sizeof(kExpectedPayloadSizes) / sizeof(size_t)));
+}
+
+class RtpDepacketizerH264Test : public ::testing::Test {
+ protected:
+  RtpDepacketizerH264Test()
+      : callback_(),
+        depacketizer_(RtpDepacketizer::Create(kRtpVideoH264, &callback_)) {
+    memset(&last_header_, 0, sizeof(last_header_));
+  }
+
+  void ExpectPacket(const uint8_t* data, size_t length) {
+    EXPECT_CALL(callback_, OnReceivedPayloadData(_, length, _))
+        .With(Args<0, 1>(ElementsAreArray(data, length)))
+        .Times(1)
+        .WillRepeatedly(DoAll(SaveArgPointee<2>(&last_header_), Return(0)));
+  }
+
+  MockRtpData callback_;
+  scoped_ptr<RtpDepacketizer> depacketizer_;
+  WebRtcRTPHeader last_header_;
+};
+
+TEST_F(RtpDepacketizerH264Test, TestSingleNalu) {
+  uint8_t packet[2] = {0x05, 0xFF};  // F=0, NRI=0, Type=5.
+
+  WebRtcRTPHeader expected_header;
+  memset(&expected_header, 0, sizeof(expected_header));
+  ExpectPacket(packet, sizeof(packet));
+  EXPECT_TRUE(depacketizer_->Parse(&expected_header, packet, sizeof(packet)));
+  EXPECT_EQ(kVideoFrameKey, last_header_.frameType);
+  EXPECT_TRUE(last_header_.type.Video.isFirstPacket);
+  EXPECT_TRUE(last_header_.type.Video.codecHeader.H264.single_nalu);
+  EXPECT_FALSE(last_header_.type.Video.codecHeader.H264.stap_a);
+}
+
+TEST_F(RtpDepacketizerH264Test, TestStapAKey) {
+  uint8_t packet[16] = {Nalu::kStapA,  // F=0, NRI=0, Type=24.
+                        // Length, nal header, payload.
+                        0,            0x02,       Nalu::kIdr, 0xFF, 0,
+                        0x03,         Nalu::kIdr, 0xFF,       0x00, 0,
+                        0x04,         Nalu::kIdr, 0xFF,       0x00, 0x11};
+
+  WebRtcRTPHeader expected_header;
+  memset(&expected_header, 0, sizeof(expected_header));
+  ExpectPacket(packet, sizeof(packet));
+  EXPECT_TRUE(depacketizer_->Parse(&expected_header, packet, sizeof(packet)));
+  EXPECT_EQ(kVideoFrameKey, last_header_.frameType);
+  EXPECT_TRUE(last_header_.type.Video.isFirstPacket);
+  EXPECT_TRUE(last_header_.type.Video.codecHeader.H264.single_nalu);
+  EXPECT_TRUE(last_header_.type.Video.codecHeader.H264.stap_a);
+}
+
+TEST_F(RtpDepacketizerH264Test, TestStapADelta) {
+  uint8_t packet[16] = {Nalu::kStapA,  // F=0, NRI=0, Type=24.
+                        // Length, nal header, payload.
+                        0,            0x02,         Nalu::kSlice, 0xFF, 0,
+                        0x03,         Nalu::kSlice, 0xFF,         0x00, 0,
+                        0x04,         Nalu::kSlice, 0xFF,         0x00, 0x11};
+
+  WebRtcRTPHeader expected_header;
+  memset(&expected_header, 0, sizeof(expected_header));
+  ExpectPacket(packet, sizeof(packet));
+  EXPECT_TRUE(depacketizer_->Parse(&expected_header, packet, sizeof(packet)));
+  EXPECT_EQ(kVideoFrameDelta, last_header_.frameType);
+  EXPECT_TRUE(last_header_.type.Video.isFirstPacket);
+  EXPECT_TRUE(last_header_.type.Video.codecHeader.H264.single_nalu);
+  EXPECT_TRUE(last_header_.type.Video.codecHeader.H264.stap_a);
+}
+
+TEST_F(RtpDepacketizerH264Test, TestFuA) {
+  uint8_t packet1[3] = {
+      Nalu::kFuA,                  // F=0, NRI=0, Type=28.
+      FuDefs::kSBit | Nalu::kIdr,  // FU header.
+      0x01                         // Payload.
+  };
+  const uint8_t kExpected1[2] = {Nalu::kIdr, 0x01};
+
+  uint8_t packet2[3] = {
+      Nalu::kFuA,  // F=0, NRI=0, Type=28.
+      Nalu::kIdr,  // FU header.
+      0x02         // Payload.
+  };
+  const uint8_t kExpected2[1] = {0x02};
+
+  uint8_t packet3[3] = {
+      Nalu::kFuA,                  // F=0, NRI=0, Type=28.
+      FuDefs::kEBit | Nalu::kIdr,  // FU header.
+      0x03                         // Payload.
+  };
+  const uint8_t kExpected3[1] = {0x03};
+
+  WebRtcRTPHeader expected_header;
+  memset(&expected_header, 0, sizeof(expected_header));
+
+  // We expect that the first packet is one byte shorter since the FU-A header
+  // has been replaced by the original nal header.
+  ExpectPacket(kExpected1, sizeof(kExpected1));
+  EXPECT_TRUE(depacketizer_->Parse(&expected_header, packet1, sizeof(packet1)));
+  EXPECT_EQ(kVideoFrameKey, last_header_.frameType);
+  EXPECT_TRUE(last_header_.type.Video.isFirstPacket);
+  EXPECT_FALSE(last_header_.type.Video.codecHeader.H264.single_nalu);
+  EXPECT_FALSE(last_header_.type.Video.codecHeader.H264.stap_a);
+
+  // Following packets will be 2 bytes shorter since they will only be appended
+  // onto the first packet.
+  ExpectPacket(kExpected2, sizeof(kExpected2));
+  EXPECT_TRUE(depacketizer_->Parse(&expected_header, packet2, sizeof(packet2)));
+  EXPECT_EQ(kVideoFrameKey, last_header_.frameType);
+  EXPECT_FALSE(last_header_.type.Video.isFirstPacket);
+  EXPECT_FALSE(last_header_.type.Video.codecHeader.H264.single_nalu);
+  EXPECT_FALSE(last_header_.type.Video.codecHeader.H264.stap_a);
+
+  ExpectPacket(kExpected3, sizeof(kExpected3));
+  EXPECT_TRUE(depacketizer_->Parse(&expected_header, packet3, sizeof(packet3)));
+  EXPECT_EQ(kVideoFrameKey, last_header_.frameType);
+  EXPECT_FALSE(last_header_.type.Video.isFirstPacket);
+  EXPECT_FALSE(last_header_.type.Video.codecHeader.H264.single_nalu);
+  EXPECT_FALSE(last_header_.type.Video.codecHeader.H264.stap_a);
+}
+}  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc
index 9c04c25..049bd05 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc
@@ -21,57 +21,65 @@
 
 // Define how the VP8PacketizerModes are implemented.
 // Modes are: kStrict, kAggregate, kEqualSize.
-const RtpFormatVp8::AggregationMode RtpFormatVp8::aggr_modes_[kNumModes] =
-    { kAggrNone, kAggrPartitions, kAggrFragments };
-const bool RtpFormatVp8::balance_modes_[kNumModes] =
-    { true, true, true };
-const bool RtpFormatVp8::separate_first_modes_[kNumModes] =
-    { true, false, false };
+const RtpPacketizerVp8::AggregationMode RtpPacketizerVp8::aggr_modes_
+    [kNumModes] = {kAggrNone, kAggrPartitions, kAggrFragments};
+const bool RtpPacketizerVp8::balance_modes_[kNumModes] = {true, true, true};
+const bool RtpPacketizerVp8::separate_first_modes_[kNumModes] = {true, false,
+                                                                 false};
 
-RtpFormatVp8::RtpFormatVp8(const uint8_t* payload_data,
-                           uint32_t payload_size,
-                           const RTPVideoHeaderVP8& hdr_info,
-                           int max_payload_len,
-                           const RTPFragmentationHeader& fragmentation,
-                           VP8PacketizerMode mode)
-    : payload_data_(payload_data),
-      payload_size_(static_cast<int>(payload_size)),
+RtpPacketizerVp8::RtpPacketizerVp8(const RTPVideoHeaderVP8& hdr_info,
+                                   int max_payload_len,
+                                   VP8PacketizerMode mode)
+    : payload_data_(NULL),
+      payload_size_(0),
       vp8_fixed_payload_descriptor_bytes_(1),
       aggr_mode_(aggr_modes_[mode]),
       balance_(balance_modes_[mode]),
       separate_first_(separate_first_modes_[mode]),
       hdr_info_(hdr_info),
-      num_partitions_(fragmentation.fragmentationVectorSize),
+      num_partitions_(0),
       max_payload_len_(max_payload_len),
       packets_calculated_(false) {
-  part_info_.CopyFrom(fragmentation);
 }
 
-RtpFormatVp8::RtpFormatVp8(const uint8_t* payload_data,
-                           uint32_t payload_size,
-                           const RTPVideoHeaderVP8& hdr_info,
-                           int max_payload_len)
-    : payload_data_(payload_data),
-      payload_size_(static_cast<int>(payload_size)),
+RtpPacketizerVp8::RtpPacketizerVp8(const RTPVideoHeaderVP8& hdr_info,
+                                   int max_payload_len)
+    : payload_data_(NULL),
+      payload_size_(0),
       part_info_(),
       vp8_fixed_payload_descriptor_bytes_(1),
       aggr_mode_(aggr_modes_[kEqualSize]),
       balance_(balance_modes_[kEqualSize]),
       separate_first_(separate_first_modes_[kEqualSize]),
       hdr_info_(hdr_info),
-      num_partitions_(1),
+      num_partitions_(0),
       max_payload_len_(max_payload_len),
       packets_calculated_(false) {
+}
+
+RtpPacketizerVp8::~RtpPacketizerVp8() {
+}
+
+void RtpPacketizerVp8::SetPayloadData(
+    const uint8_t* payload_data,
+    size_t payload_size,
+    const RTPFragmentationHeader* fragmentation) {
+  payload_data_ = payload_data;
+  payload_size_ = payload_size;
+  if (fragmentation) {
+    part_info_.CopyFrom(*fragmentation);
+    num_partitions_ = fragmentation->fragmentationVectorSize;
+  } else {
     part_info_.VerifyAndAllocateFragmentationHeader(1);
     part_info_.fragmentationLength[0] = payload_size;
     part_info_.fragmentationOffset[0] = 0;
+    num_partitions_ = part_info_.fragmentationVectorSize;
+  }
 }
 
-RtpFormatVp8::~RtpFormatVp8() {}
-
-int RtpFormatVp8::NextPacket(uint8_t* buffer,
-                             int* bytes_to_send,
-                             bool* last_packet) {
+bool RtpPacketizerVp8::NextPacket(uint8_t* buffer,
+                                  size_t* bytes_to_send,
+                                  bool* last_packet) {
   if (!packets_calculated_) {
     int ret = 0;
     if (aggr_mode_ == kAggrPartitions && balance_) {
@@ -80,26 +88,28 @@
       ret = GeneratePackets();
     }
     if (ret < 0) {
-      return ret;
+      return false;
     }
   }
   if (packets_.empty()) {
-    return -1;
+    return false;
   }
   InfoStruct packet_info = packets_.front();
   packets_.pop();
 
-  *bytes_to_send = WriteHeaderAndPayload(packet_info, buffer, max_payload_len_);
-  if (*bytes_to_send < 0) {
-    return -1;
+  int bytes = WriteHeaderAndPayload(packet_info, buffer, max_payload_len_);
+  if (bytes < 0) {
+    return false;
   }
+  *bytes_to_send = bytes;
 
   *last_packet = packets_.empty();
-  return packet_info.first_partition_ix;
+  return true;
 }
 
-int RtpFormatVp8::CalcNextSize(int max_payload_len, int remaining_bytes,
-                               bool split_payload) const {
+int RtpPacketizerVp8::CalcNextSize(int max_payload_len,
+                                   int remaining_bytes,
+                                   bool split_payload) const {
   if (max_payload_len == 0 || remaining_bytes == 0) {
     return 0;
   }
@@ -121,7 +131,7 @@
   }
 }
 
-int RtpFormatVp8::GeneratePackets() {
+int RtpPacketizerVp8::GeneratePackets() {
   if (max_payload_len_ < vp8_fixed_payload_descriptor_bytes_
       + PayloadDescriptorExtraLength() + 1) {
     // The provided payload length is not long enough for the payload
@@ -182,7 +192,7 @@
   return 0;
 }
 
-int RtpFormatVp8::GeneratePacketsBalancedAggregates() {
+int RtpPacketizerVp8::GeneratePacketsBalancedAggregates() {
   if (max_payload_len_ < vp8_fixed_payload_descriptor_bytes_
       + PayloadDescriptorExtraLength() + 1) {
     // The provided payload length is not long enough for the payload
@@ -241,9 +251,9 @@
   return 0;
 }
 
-void RtpFormatVp8::AggregateSmallPartitions(std::vector<int>* partition_vec,
-                                            int* min_size,
-                                            int* max_size) {
+void RtpPacketizerVp8::AggregateSmallPartitions(std::vector<int>* partition_vec,
+                                                int* min_size,
+                                                int* max_size) {
   assert(min_size && max_size);
   *min_size = -1;
   *max_size = -1;
@@ -285,10 +295,10 @@
   }
 }
 
-void RtpFormatVp8::QueuePacket(int start_pos,
-                               int packet_size,
-                               int first_partition_in_packet,
-                               bool start_on_new_fragment) {
+void RtpPacketizerVp8::QueuePacket(int start_pos,
+                                   int packet_size,
+                                   int first_partition_in_packet,
+                                   bool start_on_new_fragment) {
   // Write info to packet info struct and store in packet info queue.
   InfoStruct packet_info;
   packet_info.payload_start_pos = start_pos;
@@ -298,9 +308,9 @@
   packets_.push(packet_info);
 }
 
-int RtpFormatVp8::WriteHeaderAndPayload(const InfoStruct& packet_info,
-                                        uint8_t* buffer,
-                                        int buffer_length) const {
+int RtpPacketizerVp8::WriteHeaderAndPayload(const InfoStruct& packet_info,
+                                            uint8_t* buffer,
+                                            int buffer_length) const {
   // Write the VP8 payload descriptor.
   //       0
   //       0 1 2 3 4 5 6 7 8
@@ -333,8 +343,8 @@
       + extension_length;
 }
 
-int RtpFormatVp8::WriteExtensionFields(uint8_t* buffer,
-                                       int buffer_length) const {
+int RtpPacketizerVp8::WriteExtensionFields(uint8_t* buffer,
+                                           int buffer_length) const {
   int extension_length = 0;
   if (XFieldPresent()) {
     uint8_t* x_field = buffer + vp8_fixed_payload_descriptor_bytes_;
@@ -363,10 +373,10 @@
   return extension_length;
 }
 
-int RtpFormatVp8::WritePictureIDFields(uint8_t* x_field,
-                                       uint8_t* buffer,
-                                       int buffer_length,
-                                       int* extension_length) const {
+int RtpPacketizerVp8::WritePictureIDFields(uint8_t* x_field,
+                                           uint8_t* buffer,
+                                           int buffer_length,
+                                           int* extension_length) const {
   *x_field |= kIBit;
   const int pic_id_length = WritePictureID(
       buffer + vp8_fixed_payload_descriptor_bytes_ + *extension_length,
@@ -377,8 +387,7 @@
   return 0;
 }
 
-int RtpFormatVp8::WritePictureID(uint8_t* buffer,
-                                 int buffer_length) const {
+int RtpPacketizerVp8::WritePictureID(uint8_t* buffer, int buffer_length) const {
   const uint16_t pic_id =
       static_cast<uint16_t> (hdr_info_.pictureId);
   int picture_id_len = PictureIdLength();
@@ -392,10 +401,10 @@
   return picture_id_len;
 }
 
-int RtpFormatVp8::WriteTl0PicIdxFields(uint8_t* x_field,
-                                       uint8_t* buffer,
-                                       int buffer_length,
-                                       int* extension_length) const {
+int RtpPacketizerVp8::WriteTl0PicIdxFields(uint8_t* x_field,
+                                           uint8_t* buffer,
+                                           int buffer_length,
+                                           int* extension_length) const {
   if (buffer_length < vp8_fixed_payload_descriptor_bytes_ + *extension_length
       + 1) {
     return -1;
@@ -407,10 +416,10 @@
   return 0;
 }
 
-int RtpFormatVp8::WriteTIDAndKeyIdxFields(uint8_t* x_field,
-                                          uint8_t* buffer,
-                                          int buffer_length,
-                                          int* extension_length) const {
+int RtpPacketizerVp8::WriteTIDAndKeyIdxFields(uint8_t* x_field,
+                                              uint8_t* buffer,
+                                              int buffer_length,
+                                              int* extension_length) const {
   if (buffer_length < vp8_fixed_payload_descriptor_bytes_ + *extension_length
       + 1) {
     return -1;
@@ -432,7 +441,7 @@
   return 0;
 }
 
-int RtpFormatVp8::PayloadDescriptorExtraLength() const {
+int RtpPacketizerVp8::PayloadDescriptorExtraLength() const {
   int length_bytes = PictureIdLength();
   if (TL0PicIdxFieldPresent()) ++length_bytes;
   if (TIDFieldPresent() || KeyIdxFieldPresent()) ++length_bytes;
@@ -440,7 +449,7 @@
   return length_bytes;
 }
 
-int RtpFormatVp8::PictureIdLength() const {
+int RtpPacketizerVp8::PictureIdLength() const {
   if (hdr_info_.pictureId == kNoPictureId) {
     return 0;
   }
@@ -450,22 +459,22 @@
   return 2;
 }
 
-bool RtpFormatVp8::XFieldPresent() const {
+bool RtpPacketizerVp8::XFieldPresent() const {
   return (TIDFieldPresent() || TL0PicIdxFieldPresent() || PictureIdPresent()
       || KeyIdxFieldPresent());
 }
 
-bool RtpFormatVp8::TIDFieldPresent() const {
+bool RtpPacketizerVp8::TIDFieldPresent() const {
   assert((hdr_info_.layerSync == false) ||
          (hdr_info_.temporalIdx != kNoTemporalIdx));
   return (hdr_info_.temporalIdx != kNoTemporalIdx);
 }
 
-bool RtpFormatVp8::KeyIdxFieldPresent() const {
+bool RtpPacketizerVp8::KeyIdxFieldPresent() const {
   return (hdr_info_.keyIdx != kNoKeyIdx);
 }
 
-bool RtpFormatVp8::TL0PicIdxFieldPresent() const {
+bool RtpPacketizerVp8::TL0PicIdxFieldPresent() const {
   return (hdr_info_.tl0PicIdx != kNoTl0PicIdx);
 }
 }  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h b/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h
index e4d3dc0..755d19e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h
@@ -30,6 +30,7 @@
 
 #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 {
@@ -44,25 +45,24 @@
 };
 
 // Packetizer for VP8.
-class RtpFormatVp8 {
+class RtpPacketizerVp8 : public RtpPacketizer {
  public:
   // Initialize with payload from encoder and fragmentation info.
   // The payload_data must be exactly one encoded VP8 frame.
-  RtpFormatVp8(const uint8_t* payload_data,
-               uint32_t payload_size,
-               const RTPVideoHeaderVP8& hdr_info,
-               int max_payload_len,
-               const RTPFragmentationHeader& fragmentation,
-               VP8PacketizerMode mode);
+  RtpPacketizerVp8(const RTPVideoHeaderVP8& hdr_info,
+                   int max_payload_len,
+                   VP8PacketizerMode mode);
 
   // Initialize without fragmentation info. Mode kEqualSize will be used.
   // The payload_data must be exactly one encoded VP8 frame.
-  RtpFormatVp8(const uint8_t* payload_data,
-               uint32_t payload_size,
-               const RTPVideoHeaderVP8& hdr_info,
-               int max_payload_len);
+  RtpPacketizerVp8(const RTPVideoHeaderVP8& hdr_info, int max_payload_len);
 
-  ~RtpFormatVp8();
+  virtual ~RtpPacketizerVp8();
+
+  virtual void SetPayloadData(
+      const uint8_t* payload_data,
+      size_t payload_size,
+      const RTPFragmentationHeader* fragmentation) OVERRIDE;
 
   // Get the next payload with VP8 payload header.
   // max_payload_len limits the sum length of payload and VP8 payload header.
@@ -75,9 +75,9 @@
   // the first payload byte in the packet is taken, with the first partition
   // having index 0; returns negative on error.
   // For the kEqualSize mode: returns 0 on success, return negative on error.
-  int NextPacket(uint8_t* buffer,
-                 int* bytes_to_send,
-                 bool* last_packet);
+  virtual bool NextPacket(uint8_t* buffer,
+                          size_t* bytes_to_send,
+                          bool* last_packet) OVERRIDE;
 
  private:
   typedef struct {
@@ -187,7 +187,7 @@
   bool PictureIdPresent() const { return (PictureIdLength() > 0); }
 
   const uint8_t* payload_data_;
-  const int payload_size_;
+  int payload_size_;
   RTPFragmentationHeader part_info_;
   const int vp8_fixed_payload_descriptor_bytes_;  // Length of VP8 payload
                                                   // descriptors's fixed part.
@@ -195,14 +195,12 @@
   const bool balance_;
   const bool separate_first_;
   const RTPVideoHeaderVP8 hdr_info_;
-  const int num_partitions_;
+  int num_partitions_;
   const int max_payload_len_;
   InfoQueue packets_;
   bool packets_calculated_;
 
-  DISALLOW_COPY_AND_ASSIGN(RtpFormatVp8);
+  DISALLOW_COPY_AND_ASSIGN(RtpPacketizerVp8);
 };
-
 }  // namespace
-
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_H_
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.cc
index a121a69..549512b 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.cc
@@ -64,20 +64,19 @@
 }
 
 void RtpFormatVp8TestHelper::GetAllPacketsAndCheck(
-    RtpFormatVp8* packetizer,
+    RtpPacketizerVp8* packetizer,
     const int* expected_sizes,
     const int* expected_part,
     const bool* expected_frag_start,
     int expected_num_packets) {
   ASSERT_TRUE(inited_);
-  int send_bytes = 0;
+  size_t send_bytes = 0;
   bool last = false;
   for (int i = 0; i < expected_num_packets; ++i) {
     std::ostringstream ss;
     ss << "Checking packet " << i;
     SCOPED_TRACE(ss.str());
-    EXPECT_EQ(expected_part[i],
-              packetizer->NextPacket(buffer_, &send_bytes, &last));
+    EXPECT_TRUE(packetizer->NextPacket(buffer_, &send_bytes, &last));
     CheckPacket(send_bytes, expected_sizes[i], last,
                 expected_frag_start[i]);
   }
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h b/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h
index e146492..22e6803 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h
@@ -32,7 +32,7 @@
   explicit RtpFormatVp8TestHelper(const RTPVideoHeaderVP8* hdr);
   ~RtpFormatVp8TestHelper();
   bool Init(const int* partition_sizes, int num_partitions);
-  void GetAllPacketsAndCheck(RtpFormatVp8* packetizer,
+  void GetAllPacketsAndCheck(RtpPacketizerVp8* packetizer,
                              const int* expected_sizes,
                              const int* expected_part,
                              const bool* expected_frag_start,
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc
index 1d8b2be..8dfc27a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc
@@ -24,9 +24,9 @@
 
 namespace webrtc {
 
-class RtpFormatVp8Test : public ::testing::Test {
+class RtpPacketizerVp8Test : public ::testing::Test {
  protected:
-  RtpFormatVp8Test() : helper_(NULL) {}
+  RtpPacketizerVp8Test() : helper_(NULL) {}
   virtual void TearDown() { delete helper_; }
   bool Init(const int* partition_sizes, int num_partitions) {
     hdr_info_.pictureId = kNoPictureId;
@@ -44,19 +44,17 @@
   test::RtpFormatVp8TestHelper* helper_;
 };
 
-TEST_F(RtpFormatVp8Test, TestStrictMode) {
+TEST_F(RtpPacketizerVp8Test, TestStrictMode) {
   const int kSizeVector[] = {10, 8, 27};
   const int kNumPartitions = sizeof(kSizeVector) / sizeof(kSizeVector[0]);
   ASSERT_TRUE(Init(kSizeVector, kNumPartitions));
 
   hdr_info_.pictureId = 200;  // > 0x7F should produce 2-byte PictureID.
   const int kMaxSize = 13;
-  RtpFormatVp8 packetizer(helper_->payload_data(),
-                          helper_->payload_size(),
-                          hdr_info_,
-                          kMaxSize,
-                          *(helper_->fragmentation()),
-                          kStrict);
+  RtpPacketizerVp8 packetizer(hdr_info_, kMaxSize, kStrict);
+  packetizer.SetPayloadData(helper_->payload_data(),
+                            helper_->payload_size(),
+                            helper_->fragmentation());
 
   // The expected sizes are obtained by running a verified good implementation.
   const int kExpectedSizes[] = {9, 9, 12, 11, 11, 11, 10};
@@ -71,19 +69,17 @@
                                  kExpectedFragStart, kExpectedNum);
 }
 
-TEST_F(RtpFormatVp8Test, TestAggregateMode) {
+TEST_F(RtpPacketizerVp8Test, TestAggregateMode) {
   const int kSizeVector[] = {60, 10, 10};
   const int kNumPartitions = sizeof(kSizeVector) / sizeof(kSizeVector[0]);
   ASSERT_TRUE(Init(kSizeVector, kNumPartitions));
 
   hdr_info_.pictureId = 20;  // <= 0x7F should produce 1-byte PictureID.
   const int kMaxSize = 25;
-  RtpFormatVp8 packetizer(helper_->payload_data(),
-                          helper_->payload_size(),
-                          hdr_info_,
-                          kMaxSize,
-                          *(helper_->fragmentation()),
-                          kAggregate);
+  RtpPacketizerVp8 packetizer(hdr_info_, kMaxSize, kAggregate);
+  packetizer.SetPayloadData(helper_->payload_data(),
+                            helper_->payload_size(),
+                            helper_->fragmentation());
 
   // The expected sizes are obtained by running a verified good implementation.
   const int kExpectedSizes[] = {23, 23, 23, 23};
@@ -97,19 +93,17 @@
                                  kExpectedFragStart, kExpectedNum);
 }
 
-TEST_F(RtpFormatVp8Test, TestAggregateModeManyPartitions1) {
+TEST_F(RtpPacketizerVp8Test, TestAggregateModeManyPartitions1) {
   const int kSizeVector[] = {1600, 200, 200, 200, 200, 200, 200, 200, 200};
   const int kNumPartitions = sizeof(kSizeVector) / sizeof(kSizeVector[0]);
   ASSERT_TRUE(Init(kSizeVector, kNumPartitions));
 
   hdr_info_.pictureId = 20;  // <= 0x7F should produce 1-byte PictureID.
   const int kMaxSize = 1500;
-  RtpFormatVp8 packetizer(helper_->payload_data(),
-                          helper_->payload_size(),
-                          hdr_info_,
-                          kMaxSize,
-                          *(helper_->fragmentation()),
-                          kAggregate);
+  RtpPacketizerVp8 packetizer(hdr_info_, kMaxSize, kAggregate);
+  packetizer.SetPayloadData(helper_->payload_data(),
+                            helper_->payload_size(),
+                            helper_->fragmentation());
 
   // The expected sizes are obtained by running a verified good implementation.
   const int kExpectedSizes[] = {803, 803, 803, 803};
@@ -123,19 +117,17 @@
                                  kExpectedFragStart, kExpectedNum);
 }
 
-TEST_F(RtpFormatVp8Test, TestAggregateModeManyPartitions2) {
+TEST_F(RtpPacketizerVp8Test, TestAggregateModeManyPartitions2) {
   const int kSizeVector[] = {1599, 200, 200, 200, 1600, 200, 200, 200, 200};
   const int kNumPartitions = sizeof(kSizeVector) / sizeof(kSizeVector[0]);
   ASSERT_TRUE(Init(kSizeVector, kNumPartitions));
 
   hdr_info_.pictureId = 20;  // <= 0x7F should produce 1-byte PictureID.
   const int kMaxSize = 1500;
-  RtpFormatVp8 packetizer(helper_->payload_data(),
-                          helper_->payload_size(),
-                          hdr_info_,
-                          kMaxSize,
-                          *(helper_->fragmentation()),
-                          kAggregate);
+  RtpPacketizerVp8 packetizer(hdr_info_, kMaxSize, kAggregate);
+  packetizer.SetPayloadData(helper_->payload_data(),
+                            helper_->payload_size(),
+                            helper_->fragmentation());
 
   // The expected sizes are obtained by running a verified good implementation.
   const int kExpectedSizes[] = {803, 802, 603, 803, 803, 803};
@@ -149,19 +141,17 @@
                                  kExpectedFragStart, kExpectedNum);
 }
 
-TEST_F(RtpFormatVp8Test, TestAggregateModeTwoLargePartitions) {
+TEST_F(RtpPacketizerVp8Test, TestAggregateModeTwoLargePartitions) {
   const int kSizeVector[] = {1654, 2268};
   const int kNumPartitions = sizeof(kSizeVector) / sizeof(kSizeVector[0]);
   ASSERT_TRUE(Init(kSizeVector, kNumPartitions));
 
   hdr_info_.pictureId = 20;  // <= 0x7F should produce 1-byte PictureID.
   const int kMaxSize = 1460;
-  RtpFormatVp8 packetizer(helper_->payload_data(),
-                          helper_->payload_size(),
-                          hdr_info_,
-                          kMaxSize,
-                          *(helper_->fragmentation()),
-                          kAggregate);
+  RtpPacketizerVp8 packetizer(hdr_info_, kMaxSize, kAggregate);
+  packetizer.SetPayloadData(helper_->payload_data(),
+                            helper_->payload_size(),
+                            helper_->fragmentation());
 
   // The expected sizes are obtained by running a verified good implementation.
   const int kExpectedSizes[] = {830, 830, 1137, 1137};
@@ -176,17 +166,16 @@
 }
 
 // Verify that EqualSize mode is forced if fragmentation info is missing.
-TEST_F(RtpFormatVp8Test, TestEqualSizeModeFallback) {
+TEST_F(RtpPacketizerVp8Test, TestEqualSizeModeFallback) {
   const int kSizeVector[] = {10, 10, 10};
   const int kNumPartitions = sizeof(kSizeVector) / sizeof(kSizeVector[0]);
   ASSERT_TRUE(Init(kSizeVector, kNumPartitions));
 
   hdr_info_.pictureId = 200;  // > 0x7F should produce 2-byte PictureID
   const int kMaxSize = 12;  // Small enough to produce 4 packets.
-  RtpFormatVp8 packetizer(helper_->payload_data(),
-                          helper_->payload_size(),
-                          hdr_info_,
-                          kMaxSize);
+  RtpPacketizerVp8 packetizer(hdr_info_, kMaxSize);
+  packetizer.SetPayloadData(
+      helper_->payload_data(), helper_->payload_size(), NULL);
 
   // Expecting three full packets, and one with the remainder.
   const int kExpectedSizes[] = {12, 11, 12, 11};
@@ -203,17 +192,16 @@
 }
 
 // Verify that non-reference bit is set. EqualSize mode fallback is expected.
-TEST_F(RtpFormatVp8Test, TestNonReferenceBit) {
+TEST_F(RtpPacketizerVp8Test, TestNonReferenceBit) {
   const int kSizeVector[] = {10, 10, 10};
   const int kNumPartitions = sizeof(kSizeVector) / sizeof(kSizeVector[0]);
   ASSERT_TRUE(Init(kSizeVector, kNumPartitions));
 
   hdr_info_.nonReference = true;
   const int kMaxSize = 25;  // Small enough to produce two packets.
-  RtpFormatVp8 packetizer(helper_->payload_data(),
-                          helper_->payload_size(),
-                          hdr_info_,
-                          kMaxSize);
+  RtpPacketizerVp8 packetizer(hdr_info_, kMaxSize);
+  packetizer.SetPayloadData(
+      helper_->payload_data(), helper_->payload_size(), NULL);
 
   // EqualSize mode => First packet full; other not.
   const int kExpectedSizes[] = {16, 16};
@@ -230,7 +218,7 @@
 }
 
 // Verify Tl0PicIdx and TID fields, and layerSync bit.
-TEST_F(RtpFormatVp8Test, TestTl0PicIdxAndTID) {
+TEST_F(RtpPacketizerVp8Test, TestTl0PicIdxAndTID) {
   const int kSizeVector[] = {10, 10, 10};
   const int kNumPartitions = sizeof(kSizeVector) / sizeof(kSizeVector[0]);
   ASSERT_TRUE(Init(kSizeVector, kNumPartitions));
@@ -240,12 +228,10 @@
   hdr_info_.layerSync = true;
   // kMaxSize is only limited by allocated buffer size.
   const int kMaxSize = helper_->buffer_size();
-  RtpFormatVp8 packetizer(helper_->payload_data(),
-                          helper_->payload_size(),
-                          hdr_info_,
-                          kMaxSize,
-                          *(helper_->fragmentation()),
-                          kAggregate);
+  RtpPacketizerVp8 packetizer(hdr_info_, kMaxSize, kAggregate);
+  packetizer.SetPayloadData(helper_->payload_data(),
+                            helper_->payload_size(),
+                            helper_->fragmentation());
 
   // Expect one single packet of payload_size() + 4 bytes header.
   const int kExpectedSizes[1] = {helper_->payload_size() + 4};
@@ -260,7 +246,7 @@
 }
 
 // Verify KeyIdx field.
-TEST_F(RtpFormatVp8Test, TestKeyIdx) {
+TEST_F(RtpPacketizerVp8Test, TestKeyIdx) {
   const int kSizeVector[] = {10, 10, 10};
   const int kNumPartitions = sizeof(kSizeVector) / sizeof(kSizeVector[0]);
   ASSERT_TRUE(Init(kSizeVector, kNumPartitions));
@@ -268,12 +254,10 @@
   hdr_info_.keyIdx = 17;
   // kMaxSize is only limited by allocated buffer size.
   const int kMaxSize = helper_->buffer_size();
-  RtpFormatVp8 packetizer(helper_->payload_data(),
-                          helper_->payload_size(),
-                          hdr_info_,
-                          kMaxSize,
-                          *(helper_->fragmentation()),
-                          kAggregate);
+  RtpPacketizerVp8 packetizer(hdr_info_, kMaxSize, kAggregate);
+  packetizer.SetPayloadData(helper_->payload_data(),
+                            helper_->payload_size(),
+                            helper_->fragmentation());
 
   // Expect one single packet of payload_size() + 3 bytes header.
   const int kExpectedSizes[1] = {helper_->payload_size() + 3};
@@ -288,7 +272,7 @@
 }
 
 // Verify TID field and KeyIdx field in combination.
-TEST_F(RtpFormatVp8Test, TestTIDAndKeyIdx) {
+TEST_F(RtpPacketizerVp8Test, TestTIDAndKeyIdx) {
   const int kSizeVector[] = {10, 10, 10};
   const int kNumPartitions = sizeof(kSizeVector) / sizeof(kSizeVector[0]);
   ASSERT_TRUE(Init(kSizeVector, kNumPartitions));
@@ -297,12 +281,10 @@
   hdr_info_.keyIdx = 5;
   // kMaxSize is only limited by allocated buffer size.
   const int kMaxSize = helper_->buffer_size();
-  RtpFormatVp8 packetizer(helper_->payload_data(),
-                          helper_->payload_size(),
-                          hdr_info_,
-                          kMaxSize,
-                          *(helper_->fragmentation()),
-                          kAggregate);
+  RtpPacketizerVp8 packetizer(hdr_info_, kMaxSize, kAggregate);
+  packetizer.SetPayloadData(helper_->payload_data(),
+                            helper_->payload_size(),
+                            helper_->fragmentation());
 
   // Expect one single packet of payload_size() + 3 bytes header.
   const int kExpectedSizes[1] = {helper_->payload_size() + 3};
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
index c058ed5..5eb5710 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
@@ -14,6 +14,7 @@
 #include <string.h>
 
 #include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
@@ -113,8 +114,13 @@
       return ReceiveGenericCodec(rtp_header, payload_data, payload_data_length);
     case kRtpVideoVp8:
       return ReceiveVp8Codec(rtp_header, payload_data, payload_data_length);
-    case kRtpVideoH264:
-      assert(false);  // Not yet supported.
+    case kRtpVideoH264: {
+      scoped_ptr<RtpDepacketizer> depacketizer(RtpDepacketizer::Create(
+          rtp_header->type.Video.codec, data_callback_));
+      return depacketizer->Parse(rtp_header, payload_data, payload_data_length)
+                 ? 0
+                 : -1;
+    }
     case kRtpVideoNone:
       break;
   }
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
index 4d81cb3..365e4d7 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
@@ -60,7 +60,7 @@
 
   void SetPacketOverHead(uint16_t packet_over_head);
 
- protected:
+ private:
   int32_t ReceiveGenericCodec(WebRtcRTPHeader* rtp_header,
                               const uint8_t* payload_data,
                               uint16_t payload_data_length);
@@ -69,10 +69,13 @@
                           const uint8_t* payload_data,
                           uint16_t payload_data_length);
 
+  int32_t ReceiveH264Codec(WebRtcRTPHeader* rtp_header,
+                           const uint8_t* payload_data,
+                           uint16_t payload_data_length);
+
   int32_t BuildRTPheader(const WebRtcRTPHeader* rtp_header,
                          uint8_t* data_buffer) const;
 
- private:
   int32_t ParseVideoCodecSpecific(
       WebRtcRTPHeader* rtp_header,
       const uint8_t* payload_data,
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp.gypi b/webrtc/modules/rtp_rtcp/source/rtp_rtcp.gypi
index dcd6598..05741a0 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp.gypi
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp.gypi
@@ -86,6 +86,10 @@
         'rtp_sender_video.cc',
         'rtp_sender_video.h',
         'video_codec_information.h',
+        'rtp_format.cc',
+        'rtp_format.h',
+        'rtp_format_h264.cc',
+        'rtp_format_h264.h',
         'rtp_format_vp8.cc',
         'rtp_format_vp8.h',
         'rtp_format_video_generic.h',
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
index c53dd21..95e8f97 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -275,61 +275,63 @@
   return 0;
 }
 
-int32_t
-RTPSenderVideo::SendVideo(const RtpVideoCodecTypes videoType,
-                          const FrameType frameType,
-                          const int8_t payloadType,
-                          const uint32_t captureTimeStamp,
-                          int64_t capture_time_ms,
-                          const uint8_t* payloadData,
-                          const uint32_t payloadSize,
-                          const RTPFragmentationHeader* fragmentation,
-                          VideoCodecInformation* codecInfo,
-                          const RTPVideoTypeHeader* rtpTypeHdr)
-{
-    if( payloadSize == 0)
-    {
-        return -1;
-    }
+int32_t RTPSenderVideo::SendVideo(const RtpVideoCodecTypes videoType,
+                                  const FrameType frameType,
+                                  const int8_t payloadType,
+                                  const uint32_t captureTimeStamp,
+                                  int64_t capture_time_ms,
+                                  const uint8_t* payloadData,
+                                  const uint32_t payloadSize,
+                                  const RTPFragmentationHeader* fragmentation,
+                                  VideoCodecInformation* codecInfo,
+                                  const RTPVideoTypeHeader* rtpTypeHdr) {
+  if (payloadSize == 0) {
+    return -1;
+  }
 
-    if (frameType == kVideoFrameKey) {
-      producer_fec_.SetFecParameters(&key_fec_params_,
-                                     _numberFirstPartition);
-    } else {
-      producer_fec_.SetFecParameters(&delta_fec_params_,
-                                     _numberFirstPartition);
-    }
+  if (frameType == kVideoFrameKey) {
+    producer_fec_.SetFecParameters(&key_fec_params_, _numberFirstPartition);
+  } else {
+    producer_fec_.SetFecParameters(&delta_fec_params_, _numberFirstPartition);
+  }
 
-    // Default setting for number of first partition packets:
-    // Will be extracted in SendVP8 for VP8 codec; other codecs use 0
-    _numberFirstPartition = 0;
+  // Default setting for number of first partition packets:
+  // Will be extracted in SendVP8 for VP8 codec; other codecs use 0
+  _numberFirstPartition = 0;
 
-    int32_t retVal = -1;
-    switch(videoType)
-    {
+  switch (videoType) {
     case kRtpVideoGeneric:
-        retVal = SendGeneric(frameType, payloadType, captureTimeStamp,
-                             capture_time_ms, payloadData, payloadSize);
-        break;
-    case kRtpVideoVp8:
-        retVal = SendVP8(frameType,
+      return SendGeneric(frameType,
                          payloadType,
                          captureTimeStamp,
                          capture_time_ms,
                          payloadData,
-                         payloadSize,
-                         fragmentation,
-                         rtpTypeHdr);
-        break;
+                         payloadSize);
+    case kRtpVideoVp8:
+      return SendVP8(frameType,
+                     payloadType,
+                     captureTimeStamp,
+                     capture_time_ms,
+                     payloadData,
+                     payloadSize,
+                     fragmentation,
+                     rtpTypeHdr);
+    case kRtpVideoH264:
+      return SendH264(frameType,
+                      payloadType,
+                      captureTimeStamp,
+                      capture_time_ms,
+                      payloadData,
+                      payloadSize,
+                      fragmentation,
+                      rtpTypeHdr)
+                 ? 0
+                 : -1;
     default:
-        assert(false);
-        break;
-    }
-    if(retVal <= 0)
-    {
-        return retVal;
-    }
-    return 0;
+      assert(false);
+      break;
+  }
+  return 0;
 }
 
 int32_t RTPSenderVideo::SendGeneric(const FrameType frame_type,
@@ -427,8 +429,8 @@
     assert(rtpTypeHdr);
     // Initialize disregarding partition boundaries: this will use kEqualSize
     // packetization mode, which produces ~equal size packets for each frame.
-    RtpFormatVp8 packetizer(data, payloadBytesToSend, rtpTypeHdr->VP8,
-                            maxPayloadLengthVP8);
+    RtpPacketizerVp8 packetizer(rtpTypeHdr->VP8, maxPayloadLengthVP8);
+    packetizer.SetPayloadData(data, payloadBytesToSend, NULL);
 
     StorageType storage = kAllowRetransmission;
     if (rtpTypeHdr->VP8.temporalIdx == 0 &&
@@ -450,22 +452,10 @@
     {
         // Write VP8 Payload Descriptor and VP8 payload.
         uint8_t dataBuffer[IP_PACKET_SIZE] = {0};
-        int payloadBytesInPacket = 0;
-        int packetStartPartition =
-            packetizer.NextPacket(&dataBuffer[rtpHeaderLength],
-                                  &payloadBytesInPacket, &last);
-        // TODO(holmer): Temporarily disable first partition packet counting
-        // to avoid a bug in ProducerFec which doesn't properly handle
-        // important packets.
-        // if (packetStartPartition == 0)
-        // {
-        //     ++_numberFirstPartition;
-        // }
-        // else
-        if (packetStartPartition < 0)
-        {
-            return -1;
-        }
+        size_t payloadBytesInPacket = 0;
+        if (!packetizer.NextPacket(
+                &dataBuffer[rtpHeaderLength], &payloadBytesInPacket, &last))
+          return -1;
 
         // Write RTP header.
         // Set marker bit true if this is the last packet in frame.
@@ -485,6 +475,55 @@
     return 0;
 }
 
+bool RTPSenderVideo::SendH264(const FrameType frameType,
+                              const int8_t payloadType,
+                              const uint32_t captureTimeStamp,
+                              int64_t capture_time_ms,
+                              const uint8_t* payloadData,
+                              const uint32_t payloadSize,
+                              const RTPFragmentationHeader* fragmentation,
+                              const RTPVideoTypeHeader* rtpTypeHdr) {
+  size_t rtp_header_length = _rtpSender.RTPHeaderLength();
+  int32_t payload_bytes_to_send = payloadSize;
+  const uint8_t* data = payloadData;
+  size_t max_payload_length = _rtpSender.MaxDataPayloadLength();
+
+  scoped_ptr<RtpPacketizer> packetizer(
+      RtpPacketizer::Create(kRtpVideoH264, max_payload_length));
+  packetizer->SetPayloadData(data, payload_bytes_to_send, fragmentation);
+
+  StorageType storage = kAllowRetransmission;
+  bool protect = (frameType == kVideoFrameKey);
+  bool last = false;
+
+  while (!last) {
+    // Write H264 payload.
+    uint8_t dataBuffer[IP_PACKET_SIZE] = {0};
+    size_t payload_bytes_in_packet = 0;
+    if (!packetizer->NextPacket(
+            &dataBuffer[rtp_header_length], &payload_bytes_in_packet, &last)) {
+      return false;
+    }
+
+    // Write RTP header.
+    // Set marker bit true if this is the last packet in frame.
+    _rtpSender.BuildRTPheader(
+        dataBuffer, payloadType, last, captureTimeStamp, capture_time_ms);
+    if (SendVideoPacket(dataBuffer,
+                        payload_bytes_in_packet,
+                        rtp_header_length,
+                        captureTimeStamp,
+                        capture_time_ms,
+                        storage,
+                        protect)) {
+      LOG(LS_WARNING)
+          << "RTPSenderVideo::SendH264 failed to send packet number "
+          << _rtpSender.SequenceNumber();
+    }
+  }
+  return true;
+}
+
 void RTPSenderVideo::ProcessBitrate() {
   _videoBitrate.Process();
   _fecOverheadRate.Process();
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
index 82bd1de..77b91c9 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
@@ -110,6 +110,15 @@
                     const RTPFragmentationHeader* fragmentation,
                     const RTPVideoTypeHeader* rtpTypeHdr);
 
+    bool SendH264(const FrameType frameType,
+                  const int8_t payloadType,
+                  const uint32_t captureTimeStamp,
+                  int64_t capture_time_ms,
+                  const uint8_t* payloadData,
+                  const uint32_t payloadSize,
+                  const RTPFragmentationHeader* fragmentation,
+                  const RTPVideoTypeHeader* rtpTypeHdr);
+
 private:
     RTPSenderInterface&        _rtpSender;
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_utility_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_utility_unittest.cc
index 27170d5..82ec953 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_utility_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_utility_unittest.cc
@@ -251,10 +251,11 @@
   inputHeader.layerSync = false;
   inputHeader.tl0PicIdx = kNoTl0PicIdx;  // Disable.
   inputHeader.keyIdx = 31;
-  RtpFormatVp8 packetizer(payload, 10, inputHeader, 20);
+  RtpPacketizerVp8 packetizer(inputHeader, 20);
+  packetizer.SetPayloadData(payload, 10, NULL);
   bool last;
-  int send_bytes;
-  ASSERT_EQ(0, packetizer.NextPacket(packet, &send_bytes, &last));
+  size_t send_bytes;
+  ASSERT_TRUE(packetizer.NextPacket(packet, &send_bytes, &last));
   ASSERT_TRUE(last);
 
   RTPPayloadParser rtpPayloadParser(kRtpVideoVp8, packet, send_bytes);
diff --git a/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h b/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h
index 9776479..e6b1d09 100644
--- a/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h
+++ b/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h
@@ -28,29 +28,30 @@
 
 // Note: if any pointers are added to this struct, it must be fitted
 // with a copy-constructor. See below.
-struct CodecSpecificInfoVP8
-{
-    bool             hasReceivedSLI;
-    uint8_t    pictureIdSLI;
-    bool             hasReceivedRPSI;
-    uint64_t   pictureIdRPSI;
-    int16_t    pictureId;         // negative value to skip pictureId
-    bool             nonReference;
-    uint8_t    simulcastIdx;
-    uint8_t    temporalIdx;
-    bool             layerSync;
-    int              tl0PicIdx;         // Negative value to skip tl0PicIdx
-    int8_t     keyIdx;            // negative value to skip keyIdx
+struct CodecSpecificInfoVP8 {
+  bool hasReceivedSLI;
+  uint8_t pictureIdSLI;
+  bool hasReceivedRPSI;
+  uint64_t pictureIdRPSI;
+  int16_t pictureId;  // Negative value to skip pictureId.
+  bool nonReference;
+  uint8_t simulcastIdx;
+  uint8_t temporalIdx;
+  bool layerSync;
+  int tl0PicIdx;  // Negative value to skip tl0PicIdx.
+  int8_t keyIdx;  // Negative value to skip keyIdx.
 };
 
 struct CodecSpecificInfoGeneric {
   uint8_t simulcast_idx;
 };
 
-union CodecSpecificInfoUnion
-{
-    CodecSpecificInfoGeneric   generic;
-    CodecSpecificInfoVP8       VP8;
+struct CodecSpecificInfoH264 {};
+
+union CodecSpecificInfoUnion {
+  CodecSpecificInfoGeneric generic;
+  CodecSpecificInfoVP8 VP8;
+  CodecSpecificInfoH264 H264;
 };
 
 // Note: if any pointers are added to this struct or its sub-structs, it
diff --git a/webrtc/modules/video_coding/main/source/encoded_frame.cc b/webrtc/modules/video_coding/main/source/encoded_frame.cc
index 3ccf0b0..0d07955 100644
--- a/webrtc/modules/video_coding/main/source/encoded_frame.cc
+++ b/webrtc/modules/video_coding/main/source/encoded_frame.cc
@@ -100,49 +100,45 @@
 
 void VCMEncodedFrame::CopyCodecSpecific(const RTPVideoHeader* header)
 {
-    if (header)
-    {
-        switch (header->codec)
-        {
-            case kRtpVideoVp8:
-            {
-                if (_codecSpecificInfo.codecType != kVideoCodecVP8)
-                {
-                    // This is the first packet for this frame.
-                    _codecSpecificInfo.codecSpecific.VP8.pictureId = -1;
-                    _codecSpecificInfo.codecSpecific.VP8.temporalIdx = 0;
-                    _codecSpecificInfo.codecSpecific.VP8.layerSync = false;
-                    _codecSpecificInfo.codecSpecific.VP8.keyIdx = -1;
-                    _codecSpecificInfo.codecType = kVideoCodecVP8;
-                }
-                _codecSpecificInfo.codecSpecific.VP8.nonReference =
-                    header->codecHeader.VP8.nonReference;
-                if (header->codecHeader.VP8.pictureId != kNoPictureId)
-                {
-                    _codecSpecificInfo.codecSpecific.VP8.pictureId =
-                        header->codecHeader.VP8.pictureId;
-                }
-                if (header->codecHeader.VP8.temporalIdx != kNoTemporalIdx)
-                {
-                    _codecSpecificInfo.codecSpecific.VP8.temporalIdx =
-                        header->codecHeader.VP8.temporalIdx;
-                    _codecSpecificInfo.codecSpecific.VP8.layerSync =
-                        header->codecHeader.VP8.layerSync;
-                }
-                if (header->codecHeader.VP8.keyIdx != kNoKeyIdx)
-                {
-                    _codecSpecificInfo.codecSpecific.VP8.keyIdx =
-                        header->codecHeader.VP8.keyIdx;
-                }
-                break;
-            }
-            default:
-            {
-                _codecSpecificInfo.codecType = kVideoCodecUnknown;
-                break;
-            }
+  if (header) {
+    switch (header->codec) {
+      case kRtpVideoVp8: {
+        if (_codecSpecificInfo.codecType != kVideoCodecVP8) {
+          // This is the first packet for this frame.
+          _codecSpecificInfo.codecSpecific.VP8.pictureId = -1;
+          _codecSpecificInfo.codecSpecific.VP8.temporalIdx = 0;
+          _codecSpecificInfo.codecSpecific.VP8.layerSync = false;
+          _codecSpecificInfo.codecSpecific.VP8.keyIdx = -1;
+          _codecSpecificInfo.codecType = kVideoCodecVP8;
         }
+        _codecSpecificInfo.codecSpecific.VP8.nonReference =
+            header->codecHeader.VP8.nonReference;
+        if (header->codecHeader.VP8.pictureId != kNoPictureId) {
+          _codecSpecificInfo.codecSpecific.VP8.pictureId =
+              header->codecHeader.VP8.pictureId;
+        }
+        if (header->codecHeader.VP8.temporalIdx != kNoTemporalIdx) {
+          _codecSpecificInfo.codecSpecific.VP8.temporalIdx =
+              header->codecHeader.VP8.temporalIdx;
+          _codecSpecificInfo.codecSpecific.VP8.layerSync =
+              header->codecHeader.VP8.layerSync;
+        }
+        if (header->codecHeader.VP8.keyIdx != kNoKeyIdx) {
+          _codecSpecificInfo.codecSpecific.VP8.keyIdx =
+              header->codecHeader.VP8.keyIdx;
+        }
+        break;
+      }
+      case kRtpVideoH264: {
+        _codecSpecificInfo.codecType = kVideoCodecH264;
+        break;
+      }
+      default: {
+        _codecSpecificInfo.codecType = kVideoCodecUnknown;
+        break;
+      }
     }
+  }
 }
 
 const RTPFragmentationHeader* VCMEncodedFrame::FragmentationHeader() const {
diff --git a/webrtc/modules/video_coding/main/source/generic_encoder.cc b/webrtc/modules/video_coding/main/source/generic_encoder.cc
index 6fb2c9f..655f7ac 100644
--- a/webrtc/modules/video_coding/main/source/generic_encoder.cc
+++ b/webrtc/modules/video_coding/main/source/generic_encoder.cc
@@ -38,6 +38,9 @@
       (*rtp)->simulcastIdx = info->codecSpecific.VP8.simulcastIdx;
       return;
     }
+    case kVideoCodecH264:
+      (*rtp)->codec = kRtpVideoH264;
+      return;
     case kVideoCodecGeneric:
       (*rtp)->codec = kRtpVideoGeneric;
       (*rtp)->simulcastIdx = info->codecSpecific.generic.simulcast_idx;
diff --git a/webrtc/modules/video_coding/main/source/jitter_buffer.cc b/webrtc/modules/video_coding/main/source/jitter_buffer.cc
index d8792f2..9aa3409 100644
--- a/webrtc/modules/video_coding/main/source/jitter_buffer.cc
+++ b/webrtc/modules/video_coding/main/source/jitter_buffer.cc
@@ -482,7 +482,6 @@
 
 VCMEncodedFrame* VCMJitterBuffer::ExtractAndSetDecode(uint32_t timestamp) {
   CriticalSectionScoped cs(crit_sect_);
-
   if (!running_) {
     return NULL;
   }
@@ -611,7 +610,6 @@
   if (error != kNoError && frame == NULL) {
     return error;
   }
-
   int64_t now_ms = clock_->TimeInMilliseconds();
   // We are keeping track of the first and latest seq numbers, and
   // the number of wraps to be able to calculate how many packets we expect.
diff --git a/webrtc/modules/video_coding/main/source/packet.cc b/webrtc/modules/video_coding/main/source/packet.cc
index c1f1a04..63dcd63 100644
--- a/webrtc/modules/video_coding/main/source/packet.cc
+++ b/webrtc/modules/video_coding/main/source/packet.cc
@@ -94,33 +94,44 @@
   memset(&codecSpecificHeader, 0, sizeof(RTPVideoHeader));
 }
 
-void VCMPacket::CopyCodecSpecifics(const RTPVideoHeader& videoHeader)
-{
-    switch(videoHeader.codec)
-    {
-        case kRtpVideoVp8:
-            {
-                // Handle all packets within a frame as depending on the previous packet
-                // TODO(holmer): This should be changed to make fragments independent
-                // when the VP8 RTP receiver supports fragments.
-                if (isFirstPacket && markerBit)
-                    completeNALU = kNaluComplete;
-                else if (isFirstPacket)
-                    completeNALU = kNaluStart;
-                else if (markerBit)
-                    completeNALU = kNaluEnd;
-                else
-                    completeNALU = kNaluIncomplete;
+void VCMPacket::CopyCodecSpecifics(const RTPVideoHeader& videoHeader) {
+  switch (videoHeader.codec) {
+    case kRtpVideoVp8:
+      // Handle all packets within a frame as depending on the previous packet
+      // TODO(holmer): This should be changed to make fragments independent
+      // when the VP8 RTP receiver supports fragments.
+      if (isFirstPacket && markerBit)
+        completeNALU = kNaluComplete;
+      else if (isFirstPacket)
+        completeNALU = kNaluStart;
+      else if (markerBit)
+        completeNALU = kNaluEnd;
+      else
+        completeNALU = kNaluIncomplete;
 
-                codec = kVideoCodecVP8;
-                break;
-            }
-        default:
-            {
-                codec = kVideoCodecUnknown;
-                break;
-            }
-    }
+      codec = kVideoCodecVP8;
+      return;
+    case kRtpVideoH264:
+      isFirstPacket = videoHeader.isFirstPacket;
+      if (isFirstPacket)
+        insertStartCode = true;
+
+      if (videoHeader.codecHeader.H264.single_nalu) {
+        completeNALU = kNaluComplete;
+      } else if (isFirstPacket) {
+        completeNALU = kNaluStart;
+      } else if (markerBit) {
+        completeNALU = kNaluEnd;
+      } else {
+        completeNALU = kNaluIncomplete;
+      }
+      codec = kVideoCodecH264;
+      return;
+    case kRtpVideoGeneric:
+    case kRtpVideoNone:
+      codec = kVideoCodecUnknown;
+      return;
+  }
 }
 
-}
+}  // namespace webrtc
diff --git a/webrtc/modules/video_coding/main/source/session_info.cc b/webrtc/modules/video_coding/main/source/session_info.cc
index dab3da1..b50a01a 100644
--- a/webrtc/modules/video_coding/main/source/session_info.cc
+++ b/webrtc/modules/video_coding/main/source/session_info.cc
@@ -14,7 +14,7 @@
 #include "webrtc/system_wrappers/interface/logging.h"
 
 namespace webrtc {
-
+namespace {
 // Used in determining whether a frame is decodable.
 enum {kRttThreshold = 100};  // Not decodable if Rtt is lower than this.
 
@@ -23,6 +23,11 @@
 static const float kLowPacketPercentageThreshold = 0.2f;
 static const float kHighPacketPercentageThreshold = 0.8f;
 
+uint16_t BufferToUWord16(const uint8_t* dataBuffer) {
+  return (dataBuffer[0] << 8) | dataBuffer[1];
+}
+}  // namespace
+
 VCMSessionInfo::VCMSessionInfo()
     : session_nack_(false),
       complete_(false),
@@ -121,9 +126,6 @@
   VCMPacket& packet = *packet_it;
   PacketIterator it;
 
-  int packet_size = packet.sizeBytes;
-  packet_size += (packet.insertStartCode ? kH264StartCodeLengthBytes : 0);
-
   // Calculate the offset into the frame buffer for this packet.
   int offset = 0;
   for (it = packets_.begin(); it != packet_it; ++it)
@@ -131,23 +133,63 @@
 
   // Set the data pointer to pointing to the start of this packet in the
   // frame buffer.
-  const uint8_t* data = packet.dataPtr;
+  const uint8_t* packet_buffer = packet.dataPtr;
   packet.dataPtr = frame_buffer + offset;
-  packet.sizeBytes = packet_size;
 
-  ShiftSubsequentPackets(packet_it, packet_size);
-
-  const unsigned char startCode[] = {0, 0, 0, 1};
-  if (packet.insertStartCode) {
-    memcpy(const_cast<uint8_t*>(packet.dataPtr), startCode,
-           kH264StartCodeLengthBytes);
+  // We handle H.264 STAP-A packets in a special way as we need to remove the
+  // two length bytes between each NAL unit, and potentially add start codes.
+  const size_t kH264NALHeaderLengthInBytes = 1;
+  const size_t kLengthFieldLength = 2;
+  if (packet.codecSpecificHeader.codecHeader.H264.stap_a) {
+    size_t required_length = 0;
+    const uint8_t* nalu_ptr = packet_buffer + kH264NALHeaderLengthInBytes;
+    while (nalu_ptr < packet_buffer + packet.sizeBytes) {
+      uint32_t length = BufferToUWord16(nalu_ptr);
+      required_length +=
+          length + (packet.insertStartCode ? kH264StartCodeLengthBytes : 0);
+      nalu_ptr += kLengthFieldLength + length;
+    }
+    ShiftSubsequentPackets(packet_it, required_length);
+    nalu_ptr = packet_buffer + kH264NALHeaderLengthInBytes;
+    uint8_t* frame_buffer_ptr = frame_buffer + offset;
+    while (nalu_ptr < packet_buffer + packet.sizeBytes) {
+      uint32_t length = BufferToUWord16(nalu_ptr);
+      nalu_ptr += kLengthFieldLength;
+      frame_buffer_ptr += Insert(nalu_ptr,
+                                 length,
+                                 packet.insertStartCode,
+                                 const_cast<uint8_t*>(frame_buffer_ptr));
+      nalu_ptr += length;
+    }
+    packet.sizeBytes = required_length;
+    return packet.sizeBytes;
   }
-  memcpy(const_cast<uint8_t*>(packet.dataPtr
-      + (packet.insertStartCode ? kH264StartCodeLengthBytes : 0)),
-      data,
-      packet.sizeBytes);
+  ShiftSubsequentPackets(
+      packet_it,
+      packet.sizeBytes +
+          (packet.insertStartCode ? kH264StartCodeLengthBytes : 0));
 
-  return packet_size;
+  packet.sizeBytes = Insert(packet_buffer,
+                            packet.sizeBytes,
+                            packet.insertStartCode,
+                            const_cast<uint8_t*>(packet.dataPtr));
+  return packet.sizeBytes;
+}
+
+size_t VCMSessionInfo::Insert(const uint8_t* buffer,
+                              size_t length,
+                              bool insert_start_code,
+                              uint8_t* frame_buffer) {
+  if (insert_start_code) {
+    const unsigned char startCode[] = {0, 0, 0, 1};
+    memcpy(frame_buffer, startCode, kH264StartCodeLengthBytes);
+  }
+  memcpy(frame_buffer + (insert_start_code ? kH264StartCodeLengthBytes : 0),
+         buffer,
+         length);
+  length += (insert_start_code ? kH264StartCodeLengthBytes : 0);
+
+  return length;
 }
 
 void VCMSessionInfo::ShiftSubsequentPackets(PacketIterator it,
@@ -420,34 +462,49 @@
       (*rit).seqNum == packet.seqNum && (*rit).sizeBytes > 0)
     return -2;
 
-  // Only insert media packets between first and last packets (when available).
-  // Placing check here, as to properly account for duplicate packets.
-  // Check if this is first packet (only valid for some codecs)
-  // Should only be set for one packet per session.
-  if (packet.isFirstPacket && first_packet_seq_num_ == -1) {
-    // The first packet in a frame signals the frame type.
+  if (packet.codec == kVideoCodecH264) {
     frame_type_ = packet.frameType;
-    // Store the sequence number for the first packet.
-    first_packet_seq_num_ = static_cast<int>(packet.seqNum);
-  } else if (first_packet_seq_num_ != -1 &&
-        !IsNewerSequenceNumber(packet.seqNum, first_packet_seq_num_)) {
-    LOG(LS_WARNING) << "Received packet with a sequence number which is out of"
-                       "frame boundaries";
-    return -3;
-  } else if (frame_type_ == kFrameEmpty && packet.frameType != kFrameEmpty) {
-    // Update the frame type with the type of the first media packet.
-    // TODO(mikhal): Can this trigger?
-    frame_type_ = packet.frameType;
-  }
+    if (packet.isFirstPacket &&
+        (first_packet_seq_num_ == -1 ||
+         IsNewerSequenceNumber(first_packet_seq_num_, packet.seqNum))) {
+      first_packet_seq_num_ = packet.seqNum;
+    }
+    if (packet.markerBit &&
+        (last_packet_seq_num_ == -1 ||
+         IsNewerSequenceNumber(packet.seqNum, last_packet_seq_num_))) {
+      last_packet_seq_num_ = packet.seqNum;
+    }
+  } else {
+    // Only insert media packets between first and last packets (when
+    // available).
+    // Placing check here, as to properly account for duplicate packets.
+    // Check if this is first packet (only valid for some codecs)
+    // Should only be set for one packet per session.
+    if (packet.isFirstPacket && first_packet_seq_num_ == -1) {
+      // The first packet in a frame signals the frame type.
+      frame_type_ = packet.frameType;
+      // Store the sequence number for the first packet.
+      first_packet_seq_num_ = static_cast<int>(packet.seqNum);
+    } else if (first_packet_seq_num_ != -1 &&
+               !IsNewerSequenceNumber(packet.seqNum, first_packet_seq_num_)) {
+      LOG(LS_WARNING) << "Received packet with a sequence number which is out "
+                         "of frame boundaries";
+      return -3;
+    } else if (frame_type_ == kFrameEmpty && packet.frameType != kFrameEmpty) {
+      // Update the frame type with the type of the first media packet.
+      // TODO(mikhal): Can this trigger?
+      frame_type_ = packet.frameType;
+    }
 
-  // Track the marker bit, should only be set for one packet per session.
-  if (packet.markerBit && last_packet_seq_num_ == -1) {
-    last_packet_seq_num_ = static_cast<int>(packet.seqNum);
-  } else if (last_packet_seq_num_ != -1 &&
-      IsNewerSequenceNumber(packet.seqNum, last_packet_seq_num_)) {
-    LOG(LS_WARNING) << "Received packet with a sequence number which is out of"
-                       "frame boundaries";
-    return -3;
+    // Track the marker bit, should only be set for one packet per session.
+    if (packet.markerBit && last_packet_seq_num_ == -1) {
+      last_packet_seq_num_ = static_cast<int>(packet.seqNum);
+    } else if (last_packet_seq_num_ != -1 &&
+               IsNewerSequenceNumber(packet.seqNum, last_packet_seq_num_)) {
+      LOG(LS_WARNING) << "Received packet with a sequence number which is out "
+                         "of frame boundaries";
+      return -3;
+    }
   }
 
   // The insert operation invalidates the iterator |rit|.
diff --git a/webrtc/modules/video_coding/main/source/session_info.h b/webrtc/modules/video_coding/main/source/session_info.h
index cae3ee1..25216c7 100644
--- a/webrtc/modules/video_coding/main/source/session_info.h
+++ b/webrtc/modules/video_coding/main/source/session_info.h
@@ -116,6 +116,10 @@
                          const PacketIterator& prev_it);
   int InsertBuffer(uint8_t* frame_buffer,
                    PacketIterator packetIterator);
+  size_t Insert(const uint8_t* buffer,
+                size_t length,
+                bool insert_start_code,
+                uint8_t* frame_buffer);
   void ShiftSubsequentPackets(PacketIterator it, int steps_to_shift);
   PacketIterator FindNaluEnd(PacketIterator packet_iter) const;
   // Deletes the data of all packets between |start| and |end|, inclusively.