diff --git a/talk/media/webrtc/webrtcvideoengine2.cc b/talk/media/webrtc/webrtcvideoengine2.cc
index f237b8fde6..6fbaf1b401 100644
--- a/talk/media/webrtc/webrtcvideoengine2.cc
+++ b/talk/media/webrtc/webrtcvideoengine2.cc
@@ -414,6 +414,11 @@ void* WebRtcVideoChannel2::WebRtcVideoSendStream::ConfigureVideoEncoderSettings(
     denoising = options.video_noise_reduction.value_or(false);
   }
 
+  if (CodecNamesEq(codec.name, kH264CodecName)) {
+    encoder_settings_.h264 = webrtc::VideoEncoder::GetDefaultH264Settings();
+    encoder_settings_.h264.frameDroppingOn = frame_dropping;
+    return &encoder_settings_.h264;
+  }
   if (CodecNamesEq(codec.name, kVp8CodecName)) {
     encoder_settings_.vp8 = webrtc::VideoEncoder::GetDefaultVp8Settings();
     encoder_settings_.vp8.automaticResizeOn = automatic_resize;
diff --git a/talk/media/webrtc/webrtcvideoengine2.h b/talk/media/webrtc/webrtcvideoengine2.h
index eb2480c4cb..6bace03c20 100644
--- a/talk/media/webrtc/webrtcvideoengine2.h
+++ b/talk/media/webrtc/webrtcvideoengine2.h
@@ -318,6 +318,7 @@ class WebRtcVideoChannel2 : public rtc::MessageHandler,
     };
 
     union VideoEncoderSettings {
+      webrtc::VideoCodecH264 h264;
       webrtc::VideoCodecVP8 vp8;
       webrtc::VideoCodecVP9 vp9;
     };
diff --git a/talk/media/webrtc/webrtcvideoengine2_unittest.cc b/talk/media/webrtc/webrtcvideoengine2_unittest.cc
index ac397975e6..8937eed251 100644
--- a/talk/media/webrtc/webrtcvideoengine2_unittest.cc
+++ b/talk/media/webrtc/webrtcvideoengine2_unittest.cc
@@ -716,10 +716,11 @@ TEST_F(WebRtcVideoEngine2Test, SimulcastDisabledForH264) {
   EXPECT_EQ(1u, encoder->GetCodecSettings().numberOfSimulcastStreams);
 }
 
-// Test external codec with be added to the end of the supported codec list.
+// Test that external codecs are added to the end of the supported codec list.
 TEST_F(WebRtcVideoEngine2Test, ReportSupportedExternalCodecs) {
   cricket::FakeWebRtcVideoEncoderFactory encoder_factory;
-  encoder_factory.AddSupportedVideoCodecType(webrtc::kVideoCodecH264, "H264");
+  encoder_factory.AddSupportedVideoCodecType(webrtc::kVideoCodecUnknown,
+                                             "FakeExternalCodec");
   engine_.SetExternalEncoderFactory(&encoder_factory);
   engine_.Init();
 
@@ -730,7 +731,7 @@ TEST_F(WebRtcVideoEngine2Test, ReportSupportedExternalCodecs) {
 
   // The external codec will appear at last.
   EXPECT_EQ("VP8", internal_codec.name);
-  EXPECT_EQ("H264", external_codec.name);
+  EXPECT_EQ("FakeExternalCodec", external_codec.name);
 }
 
 TEST_F(WebRtcVideoEngine2Test, RegisterExternalDecodersIfSupported) {
diff --git a/webrtc/build/common.gypi b/webrtc/build/common.gypi
index d483f24a53..a77420c070 100644
--- a/webrtc/build/common.gypi
+++ b/webrtc/build/common.gypi
@@ -144,7 +144,7 @@
     # compilation succeeds but |H264DecoderImpl| fails to initialize.
     # CHECK THE OPENH264, FFMPEG AND H.264 LICENSES/PATENTS BEFORE BUILDING.
     # http://www.openh264.org, https://www.ffmpeg.org/
-    'rtc_use_h264%': 0,  # TODO(hbos): enc/dec in follow up CL(s).
+    'rtc_use_h264%': 0,
 
     'conditions': [
       ['build_with_chromium==1', {
diff --git a/webrtc/build/webrtc.gni b/webrtc/build/webrtc.gni
index f76aa36508..f784ab1830 100644
--- a/webrtc/build/webrtc.gni
+++ b/webrtc/build/webrtc.gni
@@ -98,7 +98,7 @@ declare_args() {
   # compilation succeeds but |H264DecoderImpl| fails to initialize.
   # CHECK THE OPENH264, FFMPEG AND H.264 LICENSES/PATENTS BEFORE BUILDING.
   # http://www.openh264.org, https://www.ffmpeg.org/
-  rtc_use_h264 = false  # TODO(hbos): enc/dec in follow up CL(s).
+  rtc_use_h264 = false
 }
 
 # A second declare_args block, so that declarations within it can
diff --git a/webrtc/common_video/BUILD.gn b/webrtc/common_video/BUILD.gn
index 4ef968d60f..a1686a76e9 100644
--- a/webrtc/common_video/BUILD.gn
+++ b/webrtc/common_video/BUILD.gn
@@ -48,6 +48,7 @@ source_set("common_video") {
 
   deps = [
     "..:webrtc_common",
+    "../modules/video_coding:webrtc_h264",
     "../system_wrappers",
   ]
 
diff --git a/webrtc/common_video/common_video.gyp b/webrtc/common_video/common_video.gyp
index fe14da1d2e..c40885e8f8 100644
--- a/webrtc/common_video/common_video.gyp
+++ b/webrtc/common_video/common_video.gyp
@@ -19,6 +19,7 @@
       ],
       'dependencies': [
         '<(webrtc_root)/common.gyp:webrtc_common',
+        '<(webrtc_root)/modules/modules.gyp:webrtc_h264',
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
       ],
       'direct_dependent_settings': {
diff --git a/webrtc/common_video/video_frame.cc b/webrtc/common_video/video_frame.cc
index 3bf59a720c..86de823dc9 100644
--- a/webrtc/common_video/video_frame.cc
+++ b/webrtc/common_video/video_frame.cc
@@ -19,9 +19,9 @@
 
 namespace webrtc {
 
-// TODO(hbos): The FFmpeg video decoder will require up to 8 bytes, update this
-// when the FFmpeg decoder is added.
-const size_t EncodedImage::kBufferPaddingBytesH264 = 0;
+// FFmpeg's decoder, used by H264DecoderImpl, requires up to 8 bytes padding due
+// to optimized bitstream readers. See avcodec_decode_video2.
+const size_t EncodedImage::kBufferPaddingBytesH264 = 8;
 
 bool EqualPlane(const uint8_t* data1,
                 const uint8_t* data2,
diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp
index 9c50d267e9..b0b37113b8 100644
--- a/webrtc/modules/modules.gyp
+++ b/webrtc/modules/modules.gyp
@@ -45,6 +45,13 @@
           }, {
             'desktop_capture_supported%': 0,
           }],
+          ['rtc_use_h264==1', {
+            'videoprocessor_defines': [
+              'WEBRTC_VIDEOPROCESSOR_H264_TESTS',
+            ],
+          }, {
+            'videoprocessor_defines': [],
+          }],
         ],
       },
       'targets': [
@@ -69,6 +76,7 @@
           ],
           'defines': [
             '<@(audio_coding_defines)',
+            '<@(videoprocessor_defines)',
           ],
           'sources': [
             'audio_coding/test/APITest.cc',
diff --git a/webrtc/modules/video_coding/BUILD.gn b/webrtc/modules/video_coding/BUILD.gn
index f321fbc269..115ed3588b 100644
--- a/webrtc/modules/video_coding/BUILD.gn
+++ b/webrtc/modules/video_coding/BUILD.gn
@@ -133,15 +133,19 @@ source_set("webrtc_h264") {
     configs -= [ "//build/config/clang:find_bad_constructs" ]
   }
 
+  defines = []
   deps = [
     "../../system_wrappers",
   ]
 
   if (rtc_use_h264) {
-    # Dependency for sake of compiling and so that variables use_openh264 and
-    # ffmpeg_branding are recognized build arguments (avoid "Build argument has
-    # no effect" error). The variables and dependencies will be used for real as
-    # soon as https://codereview.webrtc.org/1306813009/ lands.
+    defines += [ "WEBRTC_THIRD_PARTY_H264" ]
+    sources += [
+      "codecs/h264/h264_decoder_impl.cc",
+      "codecs/h264/h264_decoder_impl.h",
+      "codecs/h264/h264_encoder_impl.cc",
+      "codecs/h264/h264_encoder_impl.h",
+    ]
     deps += [
       "//third_party/ffmpeg:ffmpeg",
       "//third_party/openh264:encoder",
diff --git a/webrtc/modules/video_coding/codecs/h264/h264.cc b/webrtc/modules/video_coding/codecs/h264/h264.cc
index 645ed2cad7..6f7316b10a 100644
--- a/webrtc/modules/video_coding/codecs/h264/h264.cc
+++ b/webrtc/modules/video_coding/codecs/h264/h264.cc
@@ -11,12 +11,17 @@
 
 #include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
 
+#if defined(WEBRTC_THIRD_PARTY_H264)
+#include "webrtc/modules/video_coding/codecs/h264/h264_encoder_impl.h"
+#include "webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.h"
+#endif
 #if defined(WEBRTC_IOS)
 #include "webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_decoder.h"
 #include "webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.h"
 #endif
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
 
 namespace webrtc {
 
@@ -27,9 +32,15 @@ namespace webrtc {
 extern bool IsH264CodecSupportedObjC();
 #endif
 
+// If any H.264 codec is supported (iOS HW or OpenH264/FFmpeg).
 bool IsH264CodecSupported() {
 #if defined(WEBRTC_IOS)
-  return IsH264CodecSupportedObjC();
+  if (IsH264CodecSupportedObjC()) {
+    return true;
+  }
+#endif
+#if defined(WEBRTC_THIRD_PARTY_H264)
+  return true;
 #else
   return false;
 #endif
@@ -38,7 +49,14 @@ bool IsH264CodecSupported() {
 H264Encoder* H264Encoder::Create() {
   RTC_DCHECK(H264Encoder::IsSupported());
 #if defined(WEBRTC_IOS) && defined(WEBRTC_VIDEO_TOOLBOX_SUPPORTED)
-  return new H264VideoToolboxEncoder();
+  if (IsH264CodecSupportedObjC()) {
+    LOG(LS_INFO) << "Creating H264VideoToolboxEncoder.";
+    return new H264VideoToolboxEncoder();
+  }
+#endif
+#if defined(WEBRTC_THIRD_PARTY_H264)
+  LOG(LS_INFO) << "Creating H264EncoderImpl.";
+  return new H264EncoderImpl();
 #else
   RTC_NOTREACHED();
   return nullptr;
@@ -52,7 +70,14 @@ bool H264Encoder::IsSupported() {
 H264Decoder* H264Decoder::Create() {
   RTC_DCHECK(H264Decoder::IsSupported());
 #if defined(WEBRTC_IOS) && defined(WEBRTC_VIDEO_TOOLBOX_SUPPORTED)
-  return new H264VideoToolboxDecoder();
+  if (IsH264CodecSupportedObjC()) {
+    LOG(LS_INFO) << "Creating H264VideoToolboxDecoder.";
+    return new H264VideoToolboxDecoder();
+  }
+#endif
+#if defined(WEBRTC_THIRD_PARTY_H264)
+  LOG(LS_INFO) << "Creating H264DecoderImpl.";
+  return new H264DecoderImpl();
 #else
   RTC_NOTREACHED();
   return nullptr;
diff --git a/webrtc/modules/video_coding/codecs/h264/h264.gypi b/webrtc/modules/video_coding/codecs/h264/h264.gypi
index 78f0be37a7..0e80561bdf 100644
--- a/webrtc/modules/video_coding/codecs/h264/h264.gypi
+++ b/webrtc/modules/video_coding/codecs/h264/h264.gypi
@@ -24,12 +24,19 @@
           ],
         }],
         ['rtc_use_h264==1', {
-          # Dependency for sake of compiling The dependencies will be used for
-          # real as soon as https://codereview.webrtc.org/1306813009/ lands.
+          'defines': [
+            'WEBRTC_THIRD_PARTY_H264',
+          ],
           'dependencies': [
             '<(DEPTH)/third_party/ffmpeg/ffmpeg.gyp:ffmpeg',
             '<(DEPTH)/third_party/openh264/openh264.gyp:openh264_encoder',
           ],
+          'sources': [
+            'h264_decoder_impl.cc',
+            'h264_decoder_impl.h',
+            'h264_encoder_impl.cc',
+            'h264_encoder_impl.h',
+          ],
         }],
       ],
       'sources': [
diff --git a/webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc b/webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc
new file mode 100644
index 0000000000..75d2bfa732
--- /dev/null
+++ b/webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc
@@ -0,0 +1,362 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#include "webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.h"
+
+#include <algorithm>
+#include <limits>
+
+extern "C" {
+#include "third_party/ffmpeg/libavcodec/avcodec.h"
+#include "third_party/ffmpeg/libavformat/avformat.h"
+#include "third_party/ffmpeg/libavutil/imgutils.h"
+}  // extern "C"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/keep_ref_until_done.h"
+#include "webrtc/base/logging.h"
+
+namespace webrtc {
+
+namespace {
+
+const AVPixelFormat kPixelFormat = AV_PIX_FMT_YUV420P;
+const size_t kYPlaneIndex = 0;
+const size_t kUPlaneIndex = 1;
+const size_t kVPlaneIndex = 2;
+
+#if !defined(WEBRTC_CHROMIUM_BUILD)
+
+bool ffmpeg_initialized = false;
+
+// Called by FFmpeg to do mutex operations if initialized using
+// |InitializeFFmpeg|.
+int LockManagerOperation(void** lock, AVLockOp op)
+    EXCLUSIVE_LOCK_FUNCTION() UNLOCK_FUNCTION() {
+  switch (op) {
+    case AV_LOCK_CREATE:
+      *lock = new rtc::CriticalSection();
+      return 0;
+    case AV_LOCK_OBTAIN:
+      static_cast<rtc::CriticalSection*>(*lock)->Enter();
+      return 0;
+    case AV_LOCK_RELEASE:
+      static_cast<rtc::CriticalSection*>(*lock)->Leave();
+      return 0;
+    case AV_LOCK_DESTROY:
+      delete static_cast<rtc::CriticalSection*>(*lock);
+      *lock = nullptr;
+      return 0;
+  }
+  RTC_NOTREACHED() << "Unrecognized AVLockOp.";
+  return -1;
+}
+
+// TODO(hbos): Assumed to be called on a single thread. Should DCHECK that
+// InitializeFFmpeg is only called on one thread or make it thread safe.
+// See https://bugs.chromium.org/p/webrtc/issues/detail?id=5427.
+void InitializeFFmpeg() {
+  if (!ffmpeg_initialized) {
+    if (av_lockmgr_register(LockManagerOperation) < 0) {
+      RTC_NOTREACHED() << "av_lockmgr_register failed.";
+      return;
+    }
+    av_register_all();
+    ffmpeg_initialized = true;
+  }
+}
+
+#endif  // !defined(WEBRTC_CHROMIUM_BUILD)
+
+// Called by FFmpeg when it is done with a frame buffer, see AVGetBuffer2.
+void AVFreeBuffer2(void* opaque, uint8_t* data) {
+  VideoFrame* video_frame = static_cast<VideoFrame*>(opaque);
+  delete video_frame;
+}
+
+// Called by FFmpeg when it needs a frame buffer to store decoded frames in.
+// The VideoFrames returned by FFmpeg at |Decode| originate from here. They are
+// reference counted and freed by FFmpeg using |AVFreeBuffer2|.
+// TODO(hbos): Use a frame pool for better performance instead of create/free.
+// Could be owned by decoder, |static_cast<H264DecoderImpl*>(context->opaque)|.
+// Consider verifying that the buffer was allocated by us to avoid unsafe type
+// cast. See https://bugs.chromium.org/p/webrtc/issues/detail?id=5428.
+int AVGetBuffer2(AVCodecContext* context, AVFrame* av_frame, int flags) {
+  RTC_CHECK_EQ(context->pix_fmt, kPixelFormat);  // Same as in InitDecode.
+  // Necessary capability to be allowed to provide our own buffers.
+  RTC_CHECK(context->codec->capabilities | AV_CODEC_CAP_DR1);
+  // |av_frame->width| and |av_frame->height| are set by FFmpeg. These are the
+  // actual image's dimensions and may be different from |context->width| and
+  // |context->coded_width| due to reordering.
+  int width = av_frame->width;
+  int height = av_frame->height;
+  // See |lowres|, if used the decoder scales the image by 1/2^(lowres). This
+  // has implications on which resolutions are valid, but we don't use it.
+  RTC_CHECK_EQ(context->lowres, 0);
+  // Adjust the |width| and |height| to values acceptable by the decoder.
+  // Without this, FFmpeg may overflow the buffer. If modified, |width| and/or
+  // |height| are larger than the actual image and the image has to be cropped
+  // (top-left corner) after decoding to avoid visible borders to the right and
+  // bottom of the actual image.
+  avcodec_align_dimensions(context, &width, &height);
+
+  RTC_CHECK_GE(width, 0);
+  RTC_CHECK_GE(height, 0);
+  int ret = av_image_check_size(static_cast<unsigned int>(width),
+                                static_cast<unsigned int>(height), 0, nullptr);
+  if (ret < 0) {
+    LOG(LS_ERROR) << "Invalid picture size " << width << "x" << height;
+    return ret;
+  }
+
+  // The video frame is stored in |video_frame|. |av_frame| is FFmpeg's version
+  // of a video frame and will be set up to reference |video_frame|'s buffers.
+  VideoFrame* video_frame = new VideoFrame();
+  int stride_y = width;
+  int stride_uv = (width + 1) / 2;
+  RTC_CHECK_EQ(0, video_frame->CreateEmptyFrame(
+      width, height, stride_y, stride_uv, stride_uv));
+  int total_size = video_frame->allocated_size(kYPlane) +
+                   video_frame->allocated_size(kUPlane) +
+                   video_frame->allocated_size(kVPlane);
+  RTC_DCHECK_EQ(total_size, stride_y * height +
+                (stride_uv + stride_uv) * ((height + 1) / 2));
+
+  // FFmpeg expects the initial allocation to be zero-initialized according to
+  // http://crbug.com/390941.
+  // Using a single |av_frame->buf| - YUV is required to be a continuous blob of
+  // memory. We can zero-initialize with one memset operation for all planes.
+  RTC_DCHECK_EQ(video_frame->buffer(kUPlane),
+      video_frame->buffer(kYPlane) + video_frame->allocated_size(kYPlane));
+  RTC_DCHECK_EQ(video_frame->buffer(kVPlane),
+      video_frame->buffer(kUPlane) + video_frame->allocated_size(kUPlane));
+  memset(video_frame->buffer(kYPlane), 0, total_size);
+
+  av_frame->format = context->pix_fmt;
+  av_frame->reordered_opaque = context->reordered_opaque;
+
+  // Set |av_frame| members as required by FFmpeg.
+  av_frame->data[kYPlaneIndex] = video_frame->buffer(kYPlane);
+  av_frame->linesize[kYPlaneIndex] = video_frame->stride(kYPlane);
+  av_frame->data[kUPlaneIndex] = video_frame->buffer(kUPlane);
+  av_frame->linesize[kUPlaneIndex] = video_frame->stride(kUPlane);
+  av_frame->data[kVPlaneIndex] = video_frame->buffer(kVPlane);
+  av_frame->linesize[kVPlaneIndex] = video_frame->stride(kVPlane);
+  RTC_DCHECK_EQ(av_frame->extended_data, av_frame->data);
+
+  av_frame->buf[0] = av_buffer_create(av_frame->data[kYPlaneIndex],
+                                      total_size,
+                                      AVFreeBuffer2,
+                                      static_cast<void*>(video_frame),
+                                      0);
+  RTC_CHECK(av_frame->buf[0]);
+  return 0;
+}
+
+}  // namespace
+
+H264DecoderImpl::H264DecoderImpl()
+    : decoded_image_callback_(nullptr) {
+}
+
+H264DecoderImpl::~H264DecoderImpl() {
+  Release();
+}
+
+int32_t H264DecoderImpl::InitDecode(const VideoCodec* codec_settings,
+                                    int32_t number_of_cores) {
+  if (codec_settings &&
+      codec_settings->codecType != kVideoCodecH264) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+
+  // In Chromium FFmpeg will be initialized outside of WebRTC and we should not
+  // attempt to do so ourselves or it will be initialized twice.
+  // TODO(hbos): Put behind a different flag in case non-chromium project wants
+  // to initialize externally.
+  // See https://bugs.chromium.org/p/webrtc/issues/detail?id=5427.
+#if !defined(WEBRTC_CHROMIUM_BUILD)
+  // Make sure FFmpeg has been initialized.
+  InitializeFFmpeg();
+#endif
+
+  // Release necessary in case of re-initializing.
+  int32_t ret = Release();
+  if (ret != WEBRTC_VIDEO_CODEC_OK)
+    return ret;
+  RTC_DCHECK(!av_context_);
+
+  // Initialize AVCodecContext.
+  av_context_.reset(avcodec_alloc_context3(nullptr));
+
+  av_context_->codec_type = AVMEDIA_TYPE_VIDEO;
+  av_context_->codec_id = AV_CODEC_ID_H264;
+  if (codec_settings) {
+    av_context_->coded_width = codec_settings->width;
+    av_context_->coded_height = codec_settings->height;
+  }
+  av_context_->pix_fmt = kPixelFormat;
+  av_context_->extradata = nullptr;
+  av_context_->extradata_size = 0;
+
+  av_context_->thread_count = 1;
+  av_context_->thread_type = FF_THREAD_SLICE;
+
+  // FFmpeg will get video buffers from our AVGetBuffer2, memory managed by us.
+  av_context_->get_buffer2 = AVGetBuffer2;
+  // get_buffer2 is called with the context, there |opaque| can be used to get a
+  // pointer |this|.
+  av_context_->opaque = this;
+  // Use ref counted frames (av_frame_unref).
+  av_context_->refcounted_frames = 1;  // true
+
+  AVCodec* codec = avcodec_find_decoder(av_context_->codec_id);
+  if (!codec) {
+    // This is an indication that FFmpeg has not been initialized or it has not
+    // been compiled/initialized with the correct set of codecs.
+    LOG(LS_ERROR) << "FFmpeg H.264 decoder not found.";
+    Release();
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+  int res = avcodec_open2(av_context_.get(), codec, nullptr);
+  if (res < 0) {
+    LOG(LS_ERROR) << "avcodec_open2 error: " << res;
+    Release();
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+
+  av_frame_.reset(av_frame_alloc());
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t H264DecoderImpl::Release() {
+  av_context_.reset();
+  av_frame_.reset();
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t H264DecoderImpl::Reset() {
+  if (!IsInitialized())
+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+  InitDecode(nullptr, 1);
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t H264DecoderImpl::RegisterDecodeCompleteCallback(
+    DecodedImageCallback* callback) {
+  decoded_image_callback_ = callback;
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t H264DecoderImpl::Decode(const EncodedImage& input_image,
+                                bool /*missing_frames*/,
+                                const RTPFragmentationHeader* /*fragmentation*/,
+                                const CodecSpecificInfo* codec_specific_info,
+                                int64_t /*render_time_ms*/) {
+  if (!IsInitialized())
+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+  if (!decoded_image_callback_) {
+    LOG(LS_WARNING) << "InitDecode() has been called, but a callback function "
+        "has not been set with RegisterDecodeCompleteCallback()";
+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+  }
+  if (!input_image._buffer || !input_image._length)
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  if (codec_specific_info &&
+      codec_specific_info->codecType != kVideoCodecH264) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+
+  // FFmpeg requires padding due to some optimized bitstream readers reading 32
+  // or 64 bits at once and could read over the end. See avcodec_decode_video2.
+  RTC_CHECK_GE(input_image._size, input_image._length +
+                   EncodedImage::GetBufferPaddingBytes(kVideoCodecH264));
+  // "If the first 23 bits of the additional bytes are not 0, then damaged MPEG
+  // bitstreams could cause overread and segfault." See
+  // AV_INPUT_BUFFER_PADDING_SIZE. We'll zero the entire padding just in case.
+  memset(input_image._buffer + input_image._length,
+         0,
+         EncodedImage::GetBufferPaddingBytes(kVideoCodecH264));
+
+  AVPacket packet;
+  av_init_packet(&packet);
+  packet.data = input_image._buffer;
+  if (input_image._length >
+      static_cast<size_t>(std::numeric_limits<int>::max())) {
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+  packet.size = static_cast<int>(input_image._length);
+  av_context_->reordered_opaque = input_image.ntp_time_ms_ * 1000;  // ms -> μs
+
+  int frame_decoded = 0;
+  int result = avcodec_decode_video2(av_context_.get(),
+                                     av_frame_.get(),
+                                     &frame_decoded,
+                                     &packet);
+  if (result < 0) {
+    LOG(LS_ERROR) << "avcodec_decode_video2 error: " << result;
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+  // |result| is number of bytes used, which should be all of them.
+  if (result != packet.size) {
+    LOG(LS_ERROR) << "avcodec_decode_video2 consumed " << result << " bytes "
+        "when " << packet.size << " bytes were expected.";
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+
+  if (!frame_decoded) {
+    LOG(LS_WARNING) << "avcodec_decode_video2 successful but no frame was "
+        "decoded.";
+    return WEBRTC_VIDEO_CODEC_OK;
+  }
+
+  // Obtain the |video_frame| containing the decoded image.
+  VideoFrame* video_frame = static_cast<VideoFrame*>(
+      av_buffer_get_opaque(av_frame_->buf[0]));
+  RTC_DCHECK(video_frame);
+  RTC_CHECK_EQ(av_frame_->data[kYPlane], video_frame->buffer(kYPlane));
+  RTC_CHECK_EQ(av_frame_->data[kUPlane], video_frame->buffer(kUPlane));
+  RTC_CHECK_EQ(av_frame_->data[kVPlane], video_frame->buffer(kVPlane));
+  video_frame->set_timestamp(input_image._timeStamp);
+
+  // The decoded image may be larger than what is supposed to be visible, see
+  // |AVGetBuffer2|'s use of |avcodec_align_dimensions|. This crops the image
+  // without copying the underlying buffer.
+  rtc::scoped_refptr<VideoFrameBuffer> buf = video_frame->video_frame_buffer();
+  if (av_frame_->width != buf->width() || av_frame_->height != buf->height()) {
+    video_frame->set_video_frame_buffer(
+        new rtc::RefCountedObject<WrappedI420Buffer>(
+            av_frame_->width, av_frame_->height,
+            buf->data(kYPlane), buf->stride(kYPlane),
+            buf->data(kUPlane), buf->stride(kUPlane),
+            buf->data(kVPlane), buf->stride(kVPlane),
+            rtc::KeepRefUntilDone(buf)));
+  }
+
+  // Return decoded frame.
+  int32_t ret = decoded_image_callback_->Decoded(*video_frame);
+  // Stop referencing it, possibly freeing |video_frame|.
+  av_frame_unref(av_frame_.get());
+  video_frame = nullptr;
+
+  if (ret) {
+    LOG(LS_WARNING) << "DecodedImageCallback::Decoded returned " << ret;
+    return ret;
+  }
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+bool H264DecoderImpl::IsInitialized() const {
+  return av_context_ != nullptr;
+}
+
+}  // namespace webrtc
diff --git a/webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.h b/webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.h
new file mode 100644
index 0000000000..f5924fccc1
--- /dev/null
+++ b/webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.h
@@ -0,0 +1,65 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_DECODER_IMPL_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_DECODER_IMPL_H_
+
+#include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
+
+extern "C" {
+#include "third_party/ffmpeg/libavcodec/avcodec.h"
+}  // extern "C"
+
+#include "webrtc/base/scoped_ptr.h"
+
+namespace webrtc {
+
+struct AVCodecContextDeleter {
+  void operator()(AVCodecContext* ptr) const { avcodec_free_context(&ptr); }
+};
+struct AVFrameDeleter {
+  void operator()(AVFrame* ptr) const { av_frame_free(&ptr); }
+};
+
+class H264DecoderImpl : public H264Decoder {
+ public:
+  H264DecoderImpl();
+  ~H264DecoderImpl() override;
+
+  // If |codec_settings| is NULL it is ignored. If it is not NULL,
+  // |codec_settings->codecType| must be |kVideoCodecH264|.
+  int32_t InitDecode(const VideoCodec* codec_settings,
+                     int32_t number_of_cores) override;
+  int32_t Release() override;
+  int32_t Reset() override;
+
+  int32_t RegisterDecodeCompleteCallback(
+      DecodedImageCallback* callback) override;
+
+  // |missing_frames|, |fragmentation| and |render_time_ms| are ignored.
+  int32_t Decode(const EncodedImage& input_image,
+                 bool /*missing_frames*/,
+                 const RTPFragmentationHeader* /*fragmentation*/,
+                 const CodecSpecificInfo* codec_specific_info = nullptr,
+                 int64_t render_time_ms = -1) override;
+
+ private:
+  bool IsInitialized() const;
+
+  rtc::scoped_ptr<AVCodecContext, AVCodecContextDeleter> av_context_;
+  rtc::scoped_ptr<AVFrame, AVFrameDeleter> av_frame_;
+
+  DecodedImageCallback* decoded_image_callback_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_DECODER_IMPL_H_
diff --git a/webrtc/modules/video_coding/codecs/h264/h264_encoder_impl.cc b/webrtc/modules/video_coding/codecs/h264/h264_encoder_impl.cc
new file mode 100644
index 0000000000..281eb3d55a
--- /dev/null
+++ b/webrtc/modules/video_coding/codecs/h264/h264_encoder_impl.cc
@@ -0,0 +1,401 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#include "webrtc/modules/video_coding/codecs/h264/h264_encoder_impl.h"
+
+#include <limits>
+
+#include "third_party/openh264/src/codec/api/svc/codec_api.h"
+#include "third_party/openh264/src/codec/api/svc/codec_app_def.h"
+#include "third_party/openh264/src/codec/api/svc/codec_def.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
+
+namespace webrtc {
+
+namespace {
+
+const bool kOpenH264EncoderDetailedLogging = false;
+
+int NumberOfThreads(int width, int height, int number_of_cores) {
+  if (width * height >= 1920 * 1080 && number_of_cores > 8) {
+    return 8;  // 8 threads for 1080p on high perf machines.
+  } else if (width * height > 1280 * 960 && number_of_cores >= 6) {
+    return 3;  // 3 threads for 1080p.
+  } else if (width * height > 640 * 480 && number_of_cores >= 3) {
+    return 2;  // 2 threads for qHD/HD.
+  } else {
+    return 1;  // 1 thread for VGA or less.
+  }
+}
+
+}  // namespace
+
+static FrameType EVideoFrameType_to_FrameType(EVideoFrameType type) {
+  switch (type) {
+    case videoFrameTypeInvalid:
+      return kEmptyFrame;
+    case videoFrameTypeIDR:
+      return kVideoFrameKey;
+    case videoFrameTypeSkip:
+    case videoFrameTypeI:
+    case videoFrameTypeP:
+    case videoFrameTypeIPMixed:
+      return kVideoFrameDelta;
+    default:
+      LOG(LS_WARNING) << "Unknown EVideoFrameType: " << type;
+      return kVideoFrameDelta;
+  }
+}
+
+// Helper method used by H264EncoderImpl::Encode.
+// Copies the encoded bytes from |info| to |encoded_image| and updates the
+// fragmentation information of |frag_header|. The |encoded_image->_buffer| may
+// be deleted and reallocated if a bigger buffer is required.
+//
+// After OpenH264 encoding, the encoded bytes are stored in |info| spread out
+// over a number of layers and "NAL units". Each NAL unit is a fragment starting
+// with the four-byte start code {0,0,0,1}. All of this data (including the
+// start codes) is copied to the |encoded_image->_buffer| and the |frag_header|
+// is updated to point to each fragment, with offsets and lengths set as to
+// exclude the start codes.
+static void RtpFragmentize(EncodedImage* encoded_image,
+                           rtc::scoped_ptr<uint8_t[]>* encoded_image_buffer,
+                           const VideoFrame& frame,
+                           SFrameBSInfo* info,
+                           RTPFragmentationHeader* frag_header) {
+  // Calculate minimum buffer size required to hold encoded data.
+  size_t required_size = 0;
+  size_t fragments_count = 0;
+  for (int layer = 0; layer < info->iLayerNum; ++layer) {
+    const SLayerBSInfo& layerInfo = info->sLayerInfo[layer];
+    for (int nal = 0; nal < layerInfo.iNalCount; ++nal, ++fragments_count) {
+      RTC_CHECK_GE(layerInfo.pNalLengthInByte[nal], 0);
+      // Ensure |required_size| will not overflow.
+      RTC_CHECK_LE(static_cast<size_t>(layerInfo.pNalLengthInByte[nal]),
+                   std::numeric_limits<size_t>::max() - required_size);
+      required_size += layerInfo.pNalLengthInByte[nal];
+    }
+  }
+  if (encoded_image->_size < required_size) {
+    // Increase buffer size. Allocate enough to hold an unencoded image, this
+    // should be more than enough to hold any encoded data of future frames of
+    // the same size (avoiding possible future reallocation due to variations in
+    // required size).
+    encoded_image->_size = CalcBufferSize(kI420, frame.width(), frame.height());
+    if (encoded_image->_size < required_size) {
+      // Encoded data > unencoded data. Allocate required bytes.
+      LOG(LS_WARNING) << "Encoding produced more bytes than the original image "
+                      << "data! Original bytes: " << encoded_image->_size
+                      << ", encoded bytes: " << required_size << ".";
+      encoded_image->_size = required_size;
+    }
+    encoded_image->_buffer = new uint8_t[encoded_image->_size];
+    encoded_image_buffer->reset(encoded_image->_buffer);
+  }
+
+  // Iterate layers and NAL units, note each NAL unit as a fragment and copy
+  // the data to |encoded_image->_buffer|.
+  const uint8_t start_code[4] = {0, 0, 0, 1};
+  frag_header->VerifyAndAllocateFragmentationHeader(fragments_count);
+  size_t frag = 0;
+  encoded_image->_length = 0;
+  for (int layer = 0; layer < info->iLayerNum; ++layer) {
+    const SLayerBSInfo& layerInfo = info->sLayerInfo[layer];
+    // Iterate NAL units making up this layer, noting fragments.
+    size_t layer_len = 0;
+    for (int nal = 0; nal < layerInfo.iNalCount; ++nal, ++frag) {
+      // Because the sum of all layer lengths, |required_size|, fits in a
+      // |size_t|, we know that any indices in-between will not overflow.
+      RTC_DCHECK_GE(layerInfo.pNalLengthInByte[nal], 4);
+      RTC_DCHECK_EQ(layerInfo.pBsBuf[layer_len+0], start_code[0]);
+      RTC_DCHECK_EQ(layerInfo.pBsBuf[layer_len+1], start_code[1]);
+      RTC_DCHECK_EQ(layerInfo.pBsBuf[layer_len+2], start_code[2]);
+      RTC_DCHECK_EQ(layerInfo.pBsBuf[layer_len+3], start_code[3]);
+      frag_header->fragmentationOffset[frag] =
+          encoded_image->_length + layer_len + sizeof(start_code);
+      frag_header->fragmentationLength[frag] =
+          layerInfo.pNalLengthInByte[nal] - sizeof(start_code);
+      layer_len += layerInfo.pNalLengthInByte[nal];
+    }
+    // Copy the entire layer's data (including start codes).
+    memcpy(encoded_image->_buffer + encoded_image->_length,
+           layerInfo.pBsBuf,
+           layer_len);
+    encoded_image->_length += layer_len;
+  }
+}
+
+H264EncoderImpl::H264EncoderImpl()
+    : openh264_encoder_(nullptr),
+      encoded_image_callback_(nullptr) {
+}
+
+H264EncoderImpl::~H264EncoderImpl() {
+  Release();
+}
+
+int32_t H264EncoderImpl::InitEncode(const VideoCodec* codec_settings,
+                                    int32_t number_of_cores,
+                                    size_t /*max_payload_size*/) {
+  if (!codec_settings ||
+      codec_settings->codecType != kVideoCodecH264) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+  if (codec_settings->maxFramerate == 0)
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  if (codec_settings->width < 1 || codec_settings->height < 1)
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+
+  int32_t release_ret = Release();
+  if (release_ret != WEBRTC_VIDEO_CODEC_OK)
+    return release_ret;
+  RTC_DCHECK(!openh264_encoder_);
+
+  // Create encoder.
+  if (WelsCreateSVCEncoder(&openh264_encoder_) != 0) {
+    // Failed to create encoder.
+    LOG(LS_ERROR) << "Failed to create OpenH264 encoder";
+    RTC_DCHECK(!openh264_encoder_);
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+  RTC_DCHECK(openh264_encoder_);
+  if (kOpenH264EncoderDetailedLogging) {
+    int trace_level = WELS_LOG_DETAIL;
+    openh264_encoder_->SetOption(ENCODER_OPTION_TRACE_LEVEL,
+                                 &trace_level);
+  }
+  // else WELS_LOG_DEFAULT is used by default.
+
+  codec_settings_ = *codec_settings;
+  if (codec_settings_.targetBitrate == 0)
+    codec_settings_.targetBitrate = codec_settings_.startBitrate;
+
+  // Initialization parameters.
+  // There are two ways to initialize. There is SEncParamBase (cleared with
+  // memset(&p, 0, sizeof(SEncParamBase)) used in Initialize, and SEncParamExt
+  // which is a superset of SEncParamBase (cleared with GetDefaultParams) used
+  // in InitializeExt.
+  SEncParamExt init_params;
+  openh264_encoder_->GetDefaultParams(&init_params);
+  if (codec_settings_.mode == kRealtimeVideo) {
+    init_params.iUsageType = CAMERA_VIDEO_REAL_TIME;
+  } else if (codec_settings_.mode == kScreensharing) {
+    init_params.iUsageType = SCREEN_CONTENT_REAL_TIME;
+  } else {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+  init_params.iPicWidth = codec_settings_.width;
+  init_params.iPicHeight = codec_settings_.height;
+  // |init_params| uses bit/s, |codec_settings_| uses kbit/s.
+  init_params.iTargetBitrate = codec_settings_.targetBitrate * 1000;
+  init_params.iMaxBitrate = codec_settings_.maxBitrate * 1000;
+  // Rate Control mode
+  init_params.iRCMode = RC_BITRATE_MODE;
+  init_params.fMaxFrameRate = static_cast<float>(codec_settings_.maxFramerate);
+
+  // The following parameters are extension parameters (they're in SEncParamExt,
+  // not in SEncParamBase).
+  init_params.bEnableFrameSkip =
+      codec_settings_.codecSpecific.H264.frameDroppingOn;
+  // |uiIntraPeriod|    - multiple of GOP size
+  // |keyFrameInterval| - number of frames
+  init_params.uiIntraPeriod =
+      codec_settings_.codecSpecific.H264.keyFrameInterval;
+  init_params.uiMaxNalSize = 0;
+  // Threading model: use auto.
+  //  0: auto (dynamic imp. internal encoder)
+  //  1: single thread (default value)
+  // >1: number of threads
+  init_params.iMultipleThreadIdc = NumberOfThreads(init_params.iPicWidth,
+                                                   init_params.iPicHeight,
+                                                   number_of_cores);
+  // The base spatial layer 0 is the only one we use.
+  init_params.sSpatialLayers[0].iVideoWidth        = init_params.iPicWidth;
+  init_params.sSpatialLayers[0].iVideoHeight       = init_params.iPicHeight;
+  init_params.sSpatialLayers[0].fFrameRate         = init_params.fMaxFrameRate;
+  init_params.sSpatialLayers[0].iSpatialBitrate    = init_params.iTargetBitrate;
+  init_params.sSpatialLayers[0].iMaxSpatialBitrate = init_params.iMaxBitrate;
+  // Slice num according to number of threads.
+  init_params.sSpatialLayers[0].sSliceCfg.uiSliceMode = SM_AUTO_SLICE;
+
+  // Initialize.
+  if (openh264_encoder_->InitializeExt(&init_params) != 0) {
+    LOG(LS_ERROR) << "Failed to initialize OpenH264 encoder";
+    Release();
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+  int video_format = EVideoFormatType::videoFormatI420;
+  openh264_encoder_->SetOption(ENCODER_OPTION_DATAFORMAT,
+                               &video_format);
+
+  // Initialize encoded image. Default buffer size: size of unencoded data.
+  encoded_image_._size = CalcBufferSize(
+      kI420, codec_settings_.width, codec_settings_.height);
+  encoded_image_._buffer = new uint8_t[encoded_image_._size];
+  encoded_image_buffer_.reset(encoded_image_._buffer);
+  encoded_image_._completeFrame = true;
+  encoded_image_._encodedWidth = 0;
+  encoded_image_._encodedHeight = 0;
+  encoded_image_._length = 0;
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t H264EncoderImpl::Release() {
+  if (openh264_encoder_) {
+    int uninit_ret = openh264_encoder_->Uninitialize();
+    if (uninit_ret != 0) {
+      LOG(LS_WARNING) << "OpenH264 encoder's Uninitialize() returned "
+                      << "unsuccessful: " << uninit_ret;
+    }
+    WelsDestroySVCEncoder(openh264_encoder_);
+    openh264_encoder_ = nullptr;
+  }
+  if (encoded_image_._buffer != nullptr) {
+    encoded_image_._buffer = nullptr;
+    encoded_image_buffer_.reset();
+  }
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t H264EncoderImpl::RegisterEncodeCompleteCallback(
+    EncodedImageCallback* callback) {
+  encoded_image_callback_ = callback;
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t H264EncoderImpl::SetRates(uint32_t bitrate, uint32_t framerate) {
+  if (bitrate <= 0 || framerate <= 0) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+  codec_settings_.targetBitrate = bitrate;
+  codec_settings_.maxFramerate = framerate;
+
+  SBitrateInfo target_bitrate;
+  memset(&target_bitrate, 0, sizeof(SBitrateInfo));
+  target_bitrate.iLayer = SPATIAL_LAYER_ALL,
+  target_bitrate.iBitrate = codec_settings_.targetBitrate * 1000;
+  openh264_encoder_->SetOption(ENCODER_OPTION_BITRATE,
+                               &target_bitrate);
+  float max_framerate = static_cast<float>(codec_settings_.maxFramerate);
+  openh264_encoder_->SetOption(ENCODER_OPTION_FRAME_RATE,
+                               &max_framerate);
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t H264EncoderImpl::Encode(
+    const VideoFrame& frame, const CodecSpecificInfo* codec_specific_info,
+    const std::vector<FrameType>* frame_types) {
+  if (!IsInitialized())
+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+  if (frame.IsZeroSize())
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  if (!encoded_image_callback_) {
+    LOG(LS_WARNING) << "InitEncode() has been called, but a callback function "
+                    << "has not been set with RegisterEncodeCompleteCallback()";
+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+  }
+  if (frame.width()  != codec_settings_.width ||
+      frame.height() != codec_settings_.height) {
+    LOG(LS_WARNING) << "Encoder initialized for " << codec_settings_.width
+                    << "x" << codec_settings_.height << " but trying to encode "
+                    << frame.width() << "x" << frame.height() << " frame.";
+    return WEBRTC_VIDEO_CODEC_ERR_SIZE;
+  }
+
+  bool force_key_frame = false;
+  if (frame_types != nullptr) {
+    // We only support a single stream.
+    RTC_DCHECK_EQ(frame_types->size(), static_cast<size_t>(1));
+    // Skip frame?
+    if ((*frame_types)[0] == kEmptyFrame) {
+      return WEBRTC_VIDEO_CODEC_OK;
+    }
+    // Force key frame?
+    force_key_frame = (*frame_types)[0] == kVideoFrameKey;
+  }
+  if (force_key_frame) {
+    // API doc says ForceIntraFrame(false) does nothing, but calling this
+    // function forces a key frame regardless of the |bIDR| argument's value.
+    // (If every frame is a key frame we get lag/delays.)
+    openh264_encoder_->ForceIntraFrame(true);
+  }
+
+  // EncodeFrame input.
+  SSourcePicture picture;
+  memset(&picture, 0, sizeof(SSourcePicture));
+  picture.iPicWidth = frame.width();
+  picture.iPicHeight = frame.height();
+  picture.iColorFormat = EVideoFormatType::videoFormatI420;
+  picture.uiTimeStamp = frame.ntp_time_ms();
+  picture.iStride[0] = frame.stride(kYPlane);
+  picture.iStride[1] = frame.stride(kUPlane);
+  picture.iStride[2] = frame.stride(kVPlane);
+  picture.pData[0] = const_cast<uint8_t*>(frame.buffer(kYPlane));
+  picture.pData[1] = const_cast<uint8_t*>(frame.buffer(kUPlane));
+  picture.pData[2] = const_cast<uint8_t*>(frame.buffer(kVPlane));
+
+  // EncodeFrame output.
+  SFrameBSInfo info;
+  memset(&info, 0, sizeof(SFrameBSInfo));
+
+  // Encode!
+  int enc_ret = openh264_encoder_->EncodeFrame(&picture, &info);
+  if (enc_ret != 0) {
+    LOG(LS_ERROR) << "OpenH264 frame encoding failed, EncodeFrame returned "
+                  << enc_ret << ".";
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+
+  encoded_image_._encodedWidth = frame.width();
+  encoded_image_._encodedHeight = frame.height();
+  encoded_image_._timeStamp = frame.timestamp();
+  encoded_image_.ntp_time_ms_ = frame.ntp_time_ms();
+  encoded_image_.capture_time_ms_ = frame.render_time_ms();
+  encoded_image_._frameType = EVideoFrameType_to_FrameType(info.eFrameType);
+
+  // Split encoded image up into fragments. This also updates |encoded_image_|.
+  RTPFragmentationHeader frag_header;
+  RtpFragmentize(&encoded_image_, &encoded_image_buffer_, frame, &info,
+                 &frag_header);
+
+  // Encoder can skip frames to save bandwidth in which case
+  // |encoded_image_._length| == 0.
+  if (encoded_image_._length > 0) {
+    // Deliver encoded image.
+    CodecSpecificInfo codec_specific;
+    codec_specific.codecType = kVideoCodecH264;
+    encoded_image_callback_->Encoded(encoded_image_,
+                                     &codec_specific,
+                                     &frag_header);
+  }
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+bool H264EncoderImpl::IsInitialized() const {
+  return openh264_encoder_ != nullptr;
+}
+
+int32_t H264EncoderImpl::SetChannelParameters(
+    uint32_t packet_loss, int64_t rtt) {
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t H264EncoderImpl::SetPeriodicKeyFrames(bool enable) {
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+void H264EncoderImpl::OnDroppedFrame() {
+}
+
+}  // namespace webrtc
diff --git a/webrtc/modules/video_coding/codecs/h264/h264_encoder_impl.h b/webrtc/modules/video_coding/codecs/h264/h264_encoder_impl.h
new file mode 100644
index 0000000000..0bf06ac404
--- /dev/null
+++ b/webrtc/modules/video_coding/codecs/h264/h264_encoder_impl.h
@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_ENCODER_IMPL_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_ENCODER_IMPL_H_
+
+#include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
+
+#include <vector>
+
+#include "webrtc/base/scoped_ptr.h"
+
+class ISVCEncoder;
+
+namespace webrtc {
+
+class H264EncoderImpl : public H264Encoder {
+ public:
+  H264EncoderImpl();
+  ~H264EncoderImpl() override;
+
+  // |max_payload_size| is ignored.
+  // The following members of |codec_settings| are used. The rest are ignored.
+  // - codecType (must be kVideoCodecH264)
+  // - targetBitrate
+  // - maxFramerate
+  // - width
+  // - height
+  int32_t InitEncode(const VideoCodec* codec_settings,
+                     int32_t number_of_cores,
+                     size_t /*max_payload_size*/) override;
+  int32_t Release() override;
+
+  int32_t RegisterEncodeCompleteCallback(
+      EncodedImageCallback* callback) override;
+  int32_t SetRates(uint32_t bitrate, uint32_t framerate) override;
+
+  // The result of encoding - an EncodedImage and RTPFragmentationHeader - are
+  // passed to the encode complete callback.
+  int32_t Encode(const VideoFrame& frame,
+                 const CodecSpecificInfo* codec_specific_info,
+                 const std::vector<FrameType>* frame_types) override;
+
+  // Unsupported / Do nothing.
+  int32_t SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
+  int32_t SetPeriodicKeyFrames(bool enable) override;
+  void OnDroppedFrame() override;
+
+ private:
+  bool IsInitialized() const;
+
+  ISVCEncoder* openh264_encoder_;
+  VideoCodec codec_settings_;
+
+  EncodedImage encoded_image_;
+  rtc::scoped_ptr<uint8_t[]> encoded_image_buffer_;
+  EncodedImageCallback* encoded_image_callback_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_ENCODER_IMPL_H_
diff --git a/webrtc/modules/video_coding/codecs/h264/include/h264.h b/webrtc/modules/video_coding/codecs/h264/include/h264.h
index 50ca57c1c9..7f0bbf042b 100644
--- a/webrtc/modules/video_coding/codecs/h264/include/h264.h
+++ b/webrtc/modules/video_coding/codecs/h264/include/h264.h
@@ -30,6 +30,7 @@ namespace webrtc {
 class H264Encoder : public VideoEncoder {
  public:
   static H264Encoder* Create();
+  // If H.264 is supported (any implementation).
   static bool IsSupported();
 
   ~H264Encoder() override {}
diff --git a/webrtc/modules/video_coding/codecs/test/videoprocessor.cc b/webrtc/modules/video_coding/codecs/test/videoprocessor.cc
index c047c51df2..5b33e8265f 100644
--- a/webrtc/modules/video_coding/codecs/test/videoprocessor.cc
+++ b/webrtc/modules/video_coding/codecs/test/videoprocessor.cc
@@ -225,8 +225,10 @@ bool VideoProcessorImpl::ProcessFrame(int frame_number) {
   }
 }
 
-void VideoProcessorImpl::FrameEncoded(webrtc::VideoCodecType codec,
-                                      const EncodedImage& encoded_image) {
+void VideoProcessorImpl::FrameEncoded(
+    webrtc::VideoCodecType codec,
+    const EncodedImage& encoded_image,
+    const webrtc::RTPFragmentationHeader* fragmentation) {
   // Timestamp is frame number, so this gives us #dropped frames.
   int num_dropped_from_prev_encode =
       encoded_image._timeStamp - prev_time_stamp_ - 1;
@@ -277,14 +279,18 @@ void VideoProcessorImpl::FrameEncoded(webrtc::VideoCodecType codec,
         assert(false);
     }
   }
+
+  // Make a raw copy of the |encoded_image| buffer.
   size_t copied_buffer_size = encoded_image._length +
                               EncodedImage::GetBufferPaddingBytes(codec);
   rtc::scoped_ptr<uint8_t[]> copied_buffer(new uint8_t[copied_buffer_size]);
   memcpy(copied_buffer.get(), encoded_image._buffer, encoded_image._length);
+  // The image to feed to the decoder.
   EncodedImage copied_image;
   memcpy(&copied_image, &encoded_image, sizeof(copied_image));
   copied_image._size = copied_buffer_size;
   copied_image._buffer = copied_buffer.get();
+
   if (!exclude_this_frame) {
     stat.packets_dropped =
         packet_manipulator_->ManipulatePackets(&copied_image);
@@ -415,14 +421,17 @@ int32_t VideoProcessorImpl::VideoProcessorEncodeCompleteCallback::Encoded(
     const EncodedImage& encoded_image,
     const webrtc::CodecSpecificInfo* codec_specific_info,
     const webrtc::RTPFragmentationHeader* fragmentation) {
+  // Forward to parent class.
   RTC_CHECK(codec_specific_info);
   video_processor_->FrameEncoded(codec_specific_info->codecType,
-                                 encoded_image);  // Forward to parent class.
+                                 encoded_image,
+                                 fragmentation);
   return 0;
 }
 int32_t VideoProcessorImpl::VideoProcessorDecodeCompleteCallback::Decoded(
     VideoFrame& image) {
-  video_processor_->FrameDecoded(image);  // Forward to parent class.
+  // Forward to parent class.
+  video_processor_->FrameDecoded(image);
   return 0;
 }
 
diff --git a/webrtc/modules/video_coding/codecs/test/videoprocessor.h b/webrtc/modules/video_coding/codecs/test/videoprocessor.h
index 7c83de265e..cd1c7b9d62 100644
--- a/webrtc/modules/video_coding/codecs/test/videoprocessor.h
+++ b/webrtc/modules/video_coding/codecs/test/videoprocessor.h
@@ -173,7 +173,8 @@ class VideoProcessorImpl : public VideoProcessor {
  private:
   // Invoked by the callback when a frame has completed encoding.
   void FrameEncoded(webrtc::VideoCodecType codec,
-                    const webrtc::EncodedImage& encodedImage);
+                    const webrtc::EncodedImage& encodedImage,
+                    const webrtc::RTPFragmentationHeader* fragmentation);
   // Invoked by the callback when a frame has completed decoding.
   void FrameDecoded(const webrtc::VideoFrame& image);
   // Used for getting a 32-bit integer representing time
diff --git a/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc b/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
index 7b92616e1b..79d75c903b 100644
--- a/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
+++ b/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
@@ -15,9 +15,10 @@
 #include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/modules/video_coding/codecs/test/packet_manipulator.h"
 #include "webrtc/modules/video_coding/codecs/test/videoprocessor.h"
+#include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
-#include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h"
+#include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
 #include "webrtc/modules/video_coding/include/video_coding.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/testsupport/frame_reader.h"
@@ -150,7 +151,11 @@ class VideoProcessorIntegrationTest : public testing::Test {
   virtual ~VideoProcessorIntegrationTest() {}
 
   void SetUpCodecConfig() {
-    if (codec_type_ == kVideoCodecVP8) {
+    if (codec_type_ == kVideoCodecH264) {
+      encoder_ = H264Encoder::Create();
+      decoder_ = H264Decoder::Create();
+      VideoCodingModule::Codec(kVideoCodecH264, &codec_settings_);
+    } else if (codec_type_ == kVideoCodecVP8) {
       encoder_ = VP8Encoder::Create();
       decoder_ = VP8Decoder::Create();
       VideoCodingModule::Codec(kVideoCodecVP8, &codec_settings_);
@@ -184,6 +189,12 @@ class VideoProcessorIntegrationTest : public testing::Test {
 
     // These features may be set depending on the test.
     switch (config_.codec_settings->codecType) {
+      case kVideoCodecH264:
+        config_.codec_settings->codecSpecific.H264.frameDroppingOn =
+            frame_dropper_on_;
+        config_.codec_settings->codecSpecific.H264.keyFrameInterval =
+            kBaseKeyFrameInterval;
+        break;
       case kVideoCodecVP8:
         config_.codec_settings->codecSpecific.VP8.errorConcealmentOn =
             error_concealment_on_;
@@ -579,6 +590,37 @@ void SetRateControlMetrics(RateControlMetrics* rc_metrics,
   rc_metrics[update_index].num_key_frames = num_key_frames;
 }
 
+#if defined(WEBRTC_VIDEOPROCESSOR_H264_TESTS)
+
+// H264: Run with no packet loss and fixed bitrate. Quality should be very high.
+// Note(hbos): The PacketManipulatorImpl code used to simulate packet loss in
+// these unittests appears to drop "packets" in a way that is not compatible
+// with H264. Therefore ProcessXPercentPacketLossH264, X != 0, unittests have
+// not been added.
+TEST_F(VideoProcessorIntegrationTest, Process0PercentPacketLossH264) {
+  // Bitrate and frame rate profile.
+  RateProfile rate_profile;
+  SetRateProfilePars(&rate_profile, 0, 500, 30, 0);
+  rate_profile.frame_index_rate_update[1] = kNbrFramesShort + 1;
+  rate_profile.num_frames = kNbrFramesShort;
+  // Codec/network settings.
+  CodecConfigPars process_settings;
+  SetCodecParameters(&process_settings, kVideoCodecH264, 0.0f, -1, 1, false,
+                     false, true, false);
+  // Metrics for expected quality.
+  QualityMetrics quality_metrics;
+  SetQualityMetrics(&quality_metrics, 35.0, 25.0, 0.93, 0.70);
+  // Metrics for rate control.
+  RateControlMetrics rc_metrics[1];
+  SetRateControlMetrics(rc_metrics, 0, 2, 60, 20, 10, 20, 0, 1);
+  ProcessFramesAndVerify(quality_metrics,
+                         rate_profile,
+                         process_settings,
+                         rc_metrics);
+}
+
+#endif  // defined(WEBRTC_VIDEOPROCESSOR_H264_TESTS)
+
 // VP9: Run with no packet loss and fixed bitrate. Quality should be very high.
 // One key frame (first frame only) in sequence. Setting |key_frame_interval|
 // to -1 below means no periodic key frames in test.
diff --git a/webrtc/test/encoder_settings.cc b/webrtc/test/encoder_settings.cc
index bae13505c0..64339df4aa 100644
--- a/webrtc/test/encoder_settings.cc
+++ b/webrtc/test/encoder_settings.cc
@@ -58,7 +58,9 @@ VideoReceiveStream::Decoder CreateMatchingDecoder(
   VideoReceiveStream::Decoder decoder;
   decoder.payload_type = encoder_settings.payload_type;
   decoder.payload_name = encoder_settings.payload_name;
-  if (encoder_settings.payload_name == "VP8") {
+  if (encoder_settings.payload_name == "H264") {
+    decoder.decoder = VideoDecoder::Create(VideoDecoder::kH264);
+  } else if (encoder_settings.payload_name == "VP8") {
     decoder.decoder = VideoDecoder::Create(VideoDecoder::kVp8);
   } else if (encoder_settings.payload_name == "VP9") {
     decoder.decoder = VideoDecoder::Create(VideoDecoder::kVp9);
diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index d036474c13..6201e92220 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -23,6 +23,7 @@
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+#include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
 #include "webrtc/modules/video_coding/include/video_coding_defines.h"
@@ -286,12 +287,15 @@ TEST_F(EndToEndTest, SendsAndReceivesVP9) {
   RunBaseTest(&test);
 }
 
+#if defined(WEBRTC_END_TO_END_H264_TESTS)
+
 TEST_F(EndToEndTest, SendsAndReceivesH264) {
   class H264Observer : public test::EndToEndTest, public VideoRenderer {
    public:
     H264Observer()
         : EndToEndTest(2 * kDefaultTimeoutMs),
-          fake_encoder_(Clock::GetRealTimeClock()),
+          encoder_(VideoEncoder::Create(VideoEncoder::kH264)),
+          decoder_(H264Decoder::Create()),
           frame_counter_(0) {}
 
     void PerformTest() override {
@@ -305,9 +309,9 @@ TEST_F(EndToEndTest, SendsAndReceivesH264) {
         VideoEncoderConfig* encoder_config) override {
       send_config->rtp.nack.rtp_history_ms =
           (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
-      send_config->encoder_settings.encoder = &fake_encoder_;
+      send_config->encoder_settings.encoder = encoder_.get();
       send_config->encoder_settings.payload_name = "H264";
-      send_config->encoder_settings.payload_type = kFakeVideoSendPayloadType;
+      send_config->encoder_settings.payload_type = 126;
       encoder_config->streams[0].min_bitrate_bps = 50000;
       encoder_config->streams[0].target_bitrate_bps =
           encoder_config->streams[0].max_bitrate_bps = 2000000;
@@ -318,7 +322,7 @@ TEST_F(EndToEndTest, SendsAndReceivesH264) {
           send_config->encoder_settings.payload_type;
       (*receive_configs)[0].decoders[0].payload_name =
           send_config->encoder_settings.payload_name;
-      (*receive_configs)[0].decoders[0].decoder = &fake_decoder_;
+      (*receive_configs)[0].decoders[0].decoder = decoder_.get();
     }
 
     void RenderFrame(const VideoFrame& video_frame,
@@ -331,14 +335,16 @@ TEST_F(EndToEndTest, SendsAndReceivesH264) {
     bool IsTextureSupported() const override { return false; }
 
    private:
-    test::FakeH264Decoder fake_decoder_;
-    test::FakeH264Encoder fake_encoder_;
+    rtc::scoped_ptr<webrtc::VideoEncoder> encoder_;
+    rtc::scoped_ptr<webrtc::VideoDecoder> decoder_;
     int frame_counter_;
   } test;
 
   RunBaseTest(&test);
 }
 
+#endif  // defined(WEBRTC_END_TO_END_H264_TESTS)
+
 TEST_F(EndToEndTest, ReceiverUsesLocalSsrc) {
   class SyncRtcpObserver : public test::EndToEndTest {
    public:
diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc
index 6e8c9902c6..5a8c4db731 100644
--- a/webrtc/video/video_quality_test.cc
+++ b/webrtc/video/video_quality_test.cc
@@ -37,6 +37,7 @@
 namespace webrtc {
 
 static const int kSendStatsPollingIntervalMs = 1000;
+static const int kPayloadTypeH264 = 122;
 static const int kPayloadTypeVP8 = 123;
 static const int kPayloadTypeVP9 = 124;
 
@@ -780,7 +781,10 @@ void VideoQualityTest::SetupCommon(Transport* send_transport,
   CreateSendConfig(num_streams, 0, send_transport);
 
   int payload_type;
-  if (params_.common.codec == "VP8") {
+  if (params_.common.codec == "H264") {
+    encoder_.reset(VideoEncoder::Create(VideoEncoder::kH264));
+    payload_type = kPayloadTypeH264;
+  } else if (params_.common.codec == "VP8") {
     encoder_.reset(VideoEncoder::Create(VideoEncoder::kVp8));
     payload_type = kPayloadTypeVP8;
   } else if (params_.common.codec == "VP9") {
diff --git a/webrtc/webrtc_tests.gypi b/webrtc/webrtc_tests.gypi
index 998c4e0bde..94c9731e8b 100644
--- a/webrtc/webrtc_tests.gypi
+++ b/webrtc/webrtc_tests.gypi
@@ -194,6 +194,11 @@
         'webrtc',
       ],
       'conditions': [
+        ['rtc_use_h264==1', {
+          'defines': [
+            'WEBRTC_END_TO_END_H264_TESTS',
+          ],
+        }],
         ['OS=="android"', {
           'dependencies': [
             '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',