Roll libjingle to r108:

* Add app/webrtc.
* Add examples/peerconnection.
* Bug fixes.

Roll webrtc to 1311.

BUG=110255
TEST=build and unit tests


Review URL: http://codereview.chromium.org/9174012

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@118031 0039d316-1c4b-4281-b951-d872f2087c98

4 files changed, 25 insertions, 1640 deletions

diff --git a/third_party/libjingle/README.chromium b/third_party/libjingle/README.chromium
index dd77c10..5b768df 100644
--- a/third_party/libjingle/README.chromium
+++ b/third_party/libjingle/README.chromium
@@ -1,8 +1,8 @@
 Name: Libjingle library. Used for p2p voice and video communication.
 Short Name: libjingle
 URL: http://code.google.com/p/libjingle/
-Version: 2.7.0
-Revision: 100
+Version: 0.6.10
+Revision: 108
 License: BSD
 License File: source/COPYING
 Security Critical: yes
diff --git a/third_party/libjingle/libjingle.gyp b/third_party/libjingle/libjingle.gyp
index 3f8aa60..916ca84 100644
--- a/third_party/libjingle/libjingle.gyp
+++ b/third_party/libjingle/libjingle.gyp
@@ -282,8 +282,10 @@
         'source/talk/base/taskrunner.h',
         'source/talk/base/thread.cc',
         'source/talk/base/thread.h',
-        'source/talk/base/time.cc',
-        'source/talk/base/time.h',
+        'source/talk/base/timeutils.cc',
+        'source/talk/base/timeutils.h',
+        'source/talk/base/timing.cc',
+        'source/talk/base/timing.h',
         'source/talk/base/urlencode.cc',
         'source/talk/base/urlencode.h',
         'source/talk/base/worker.cc',
@@ -309,8 +311,6 @@
         'source/talk/xmpp/jid.h',
         'source/talk/xmpp/plainsaslhandler.h',
         'source/talk/xmpp/prexmppauth.h',
-        'source/talk/xmpp/ratelimitmanager.cc',
-        'source/talk/xmpp/ratelimitmanager.h',
         'source/talk/xmpp/saslcookiemechanism.h',
         'source/talk/xmpp/saslhandler.h',
         'source/talk/xmpp/saslmechanism.cc',
@@ -458,17 +458,17 @@
       'target_name': 'libjingle_peerconnection',
       'type': 'static_library',
       'sources': [
-        'source/talk/app/webrtc/peerconnection.h',
-        'source/talk/app/webrtc/peerconnectionfactory.cc',
-        'source/talk/app/webrtc/peerconnectionfactory.h',
-        'source/talk/app/webrtc/peerconnectionimpl.cc',
-        'source/talk/app/webrtc/peerconnectionimpl.h',
-        'source/talk/app/webrtc/peerconnectionproxy.cc',
-        'source/talk/app/webrtc/peerconnectionproxy.h',
-        'source/talk/app/webrtc/webrtcsession.cc',
-        'source/talk/app/webrtc/webrtcsession.h',
-        'source/talk/app/webrtc/webrtcjson.cc',
-        'source/talk/app/webrtc/webrtcjson.h',
+        'source/talk/app/webrtcv1/peerconnection.h',
+        'source/talk/app/webrtcv1/peerconnectionfactory.cc',
+        'source/talk/app/webrtcv1/peerconnectionfactory.h',
+        'source/talk/app/webrtcv1/peerconnectionimpl.cc',
+        'source/talk/app/webrtcv1/peerconnectionimpl.h',
+        'source/talk/app/webrtcv1/peerconnectionproxy.cc',
+        'source/talk/app/webrtcv1/peerconnectionproxy.h',
+        'source/talk/app/webrtcv1/webrtcsession.cc',
+        'source/talk/app/webrtcv1/webrtcsession.h',
+        'source/talk/app/webrtcv1/webrtcjson.cc',
+        'source/talk/app/webrtcv1/webrtcjson.h',
         'source/talk/session/phone/audiomonitor.cc',
         'source/talk/session/phone/audiomonitor.h',
         'source/talk/session/phone/call.cc',
@@ -512,6 +512,7 @@
         'source/talk/session/phone/srtpfilter.h',
         'source/talk/session/phone/ssrcmuxfilter.cc',
         'source/talk/session/phone/ssrcmuxfilter.h',
+        'source/talk/session/phone/streamparams.cc',
         'source/talk/session/phone/videocapturer.cc',
         'source/talk/session/phone/videocapturer.h',
         'source/talk/session/phone/videocommon.cc',
@@ -523,9 +524,7 @@
         'source/talk/session/phone/webrtcpassthroughrender.cc',
         'source/talk/session/phone/webrtcvideocapturer.cc',
         'source/talk/session/phone/webrtcvideocapturer.h',
-        # TODO(ronghuawu): Fix this in libjingle and remove the override.
-        # See http://crbug.com/110255
-        'overrides/talk/session/phone/webrtcvideoengine.cc',
+        'source/talk/session/phone/webrtcvideoengine.cc',
         'source/talk/session/phone/webrtcvideoengine.h',
         'source/talk/session/phone/webrtcvideoframe.cc',
         'source/talk/session/phone/webrtcvideoframe.h',
diff --git a/third_party/libjingle/overrides/talk/base/basictypes.h b/third_party/libjingle/overrides/talk/base/basictypes.h
index 4ea193e..eeecc4b 100644
--- a/third_party/libjingle/overrides/talk/base/basictypes.h
+++ b/third_party/libjingle/overrides/talk/base/basictypes.h
@@ -1,4 +1,4 @@
-// Copyright (c) 2011 The Chromium Authors. All rights reserved.
+// Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
@@ -60,4 +60,9 @@ const int kForever = -1;
   (reinterpret_cast<uint8*>(((reinterpret_cast<uintptr_t>(p) + \
   ((t)-1)) & ~((t)-1))))
 
+// LIBJINGLE_DEFINE_STATIC_LOCAL() is a libjingle's copy
+// of CR_DEFINE_STATIC_LOCAL().
+#define LIBJINGLE_DEFINE_STATIC_LOCAL(type, name, arguments) \
+  CR_DEFINE_STATIC_LOCAL(type, name, arguments)
+
 #endif // OVERRIDES_TALK_BASE_BASICTYPES_H__
diff --git a/third_party/libjingle/overrides/talk/session/phone/webrtcvideoengine.cc b/third_party/libjingle/overrides/talk/session/phone/webrtcvideoengine.cc
deleted file mode 100644
index 829fac3..0000000
--- a/third_party/libjingle/overrides/talk/session/phone/webrtcvideoengine.cc
+++ /dev/null
@@ -1,1619 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2011, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#ifdef HAVE_CONFIG_H
-#include <config.h>
-#endif
-
-#ifdef HAVE_WEBRTC_VIDEO
-
-#include "talk/session/phone/webrtcvideoengine.h"
-
-#include "talk/base/basictypes.h"
-#include "talk/base/common.h"
-#include "talk/base/buffer.h"
-#include "talk/base/byteorder.h"
-#include "talk/base/logging.h"
-#include "talk/base/stringutils.h"
-#include "talk/session/phone/videorenderer.h"
-#include "talk/session/phone/webrtcpassthroughrender.h"
-#include "talk/session/phone/webrtcvoiceengine.h"
-#include "talk/session/phone/webrtcvideocapturer.h"
-#include "talk/session/phone/webrtcvideoframe.h"
-#include "talk/session/phone/webrtcvie.h"
-#include "talk/session/phone/webrtcvoe.h"
-
-// TODO Change video protection calls when WebRTC API has changed.
-#define WEBRTC_VIDEO_AVPF_NACK_ONLY
-
-namespace cricket {
-
-static const int kDefaultLogSeverity = talk_base::LS_WARNING;
-
-static const int kMinVideoBitrate = 100;
-static const int kStartVideoBitrate = 300;
-static const int kMaxVideoBitrate = 2000;
-
-static const int kVideoMtu = 1200;
-
-static const int kVideoRtpBufferSize = 65536;
-
-static const char kVp8PayloadName[] = "VP8";
-static const char kRedPayloadName[] = "red";
-static const char kFecPayloadName[] = "ulpfec";
-
-static const int kDefaultNumberOfTemporalLayers = 3;
-
-static void LogMultiline(talk_base::LoggingSeverity sev, char* text) {
-  const char* delim = "\r\n";
-  for (char* tok = strtok(text, delim); tok; tok = strtok(NULL, delim)) {
-    LOG_V(sev) << tok;
-  }
-}
-
-class WebRtcRenderAdapter : public webrtc::ExternalRenderer {
- public:
-  explicit WebRtcRenderAdapter(VideoRenderer* renderer)
-      : renderer_(renderer), width_(0), height_(0) {
-  }
-  virtual ~WebRtcRenderAdapter() {
-  }
-
-  void SetRenderer(VideoRenderer* renderer) {
-    talk_base::CritScope cs(&crit_);
-    renderer_ = renderer;
-  }
-  // Implementation of webrtc::ExternalRenderer.
-  virtual int FrameSizeChange(unsigned int width, unsigned int height,
-                              unsigned int /*number_of_streams*/) {
-    talk_base::CritScope cs(&crit_);
-    if (renderer_ == NULL) {
-      return 0;
-    }
-    width_ = width;
-    height_ = height;
-    return renderer_->SetSize(width_, height_, 0) ? 0 : -1;
-  }
-  virtual int DeliverFrame(unsigned char* buffer, int buffer_size,
-                           unsigned int time_stamp) {
-    talk_base::CritScope cs(&crit_);
-    frame_rate_tracker_.Update(1);
-    if (renderer_ == NULL) {
-      return 0;
-    }
-    WebRtcVideoFrame video_frame;
-    video_frame.Attach(buffer, buffer_size, width_, height_,
-                       1, 1, 0, time_stamp, 0);
-
-    int ret = renderer_->RenderFrame(&video_frame) ? 0 : -1;
-    uint8* buffer_temp;
-    size_t buffer_size_temp;
-    video_frame.Detach(&buffer_temp, &buffer_size_temp);
-    return ret;
-  }
-
-  unsigned int width() {
-   talk_base::CritScope cs(&crit_);
-   return width_;
-  }
-  unsigned int height() {
-   talk_base::CritScope cs(&crit_);
-   return height_;
-  }
-  int framerate() {
-    talk_base::CritScope cs(&crit_);
-    return frame_rate_tracker_.units_second();
-  }
-
- private:
-  talk_base::CriticalSection crit_;
-  VideoRenderer* renderer_;
-  unsigned int width_;
-  unsigned int height_;
-  talk_base::RateTracker frame_rate_tracker_;
-};
-
-class WebRtcDecoderObserver : public webrtc::ViEDecoderObserver {
- public:
-  WebRtcDecoderObserver(int video_channel)
-       : video_channel_(video_channel),
-         framerate_(0),
-         bitrate_(0),
-         firs_requested_(0) { }
-
-  // virtual functions from VieDecoderObserver.
-  virtual void IncomingCodecChanged(const int videoChannel,
-                                    const webrtc::VideoCodec& videoCodec) { }
-  virtual void IncomingRate(const int videoChannel,
-                            const unsigned int framerate,
-                            const unsigned int bitrate) {
-    ASSERT(video_channel_ == videoChannel);
-    framerate_ = framerate;
-    bitrate_ = bitrate;
-  }
-  virtual void RequestNewKeyFrame(const int videoChannel) {
-    ASSERT(video_channel_ == videoChannel);
-    ++firs_requested_;
-  }
-
-  int framerate() const { return framerate_; }
-  int bitrate() const { return bitrate_; }
-  int firs_requested() const { return firs_requested_; }
-
- private:
-  int video_channel_;
-  int framerate_;
-  int bitrate_;
-  int firs_requested_;
-};
-
-class WebRtcEncoderObserver : public webrtc::ViEEncoderObserver {
- public:
-  WebRtcEncoderObserver(int video_channel)
-       : video_channel_(video_channel), framerate_(0), bitrate_(0) { }
-
-  // virtual functions from VieEncoderObserver.
-  virtual void OutgoingRate(const int videoChannel,
-                            const unsigned int framerate,
-                            const unsigned int bitrate) {
-    ASSERT(video_channel_ == videoChannel);
-    framerate_ = framerate;
-    bitrate_ = bitrate;
-  }
-
-  int framerate() const { return framerate_; }
-  int bitrate() const { return bitrate_; }
-
- private:
-  int video_channel_;
-  int framerate_;
-  int bitrate_;
-};
-
-class LocalStreamInfo {
- public:
-  int width() {
-    talk_base::CritScope cs(&crit_);
-    return width_;
-  }
-  int height() {
-    talk_base::CritScope cs(&crit_);
-    return height_;
-  }
-  int framerate() {
-    talk_base::CritScope cs(&crit_);
-    return rate_tracker_.units_second();
-  }
-
-  void UpdateFrame(int width, int height) {
-    talk_base::CritScope cs(&crit_);
-    width_ = width;
-    height_ = height;
-    rate_tracker_.Update(1);
-  }
-
- private:
-  talk_base::CriticalSection crit_;
-  unsigned int width_;
-  unsigned int height_;
-  talk_base::RateTracker rate_tracker_;
-};
-
-const WebRtcVideoEngine::VideoCodecPref
-    WebRtcVideoEngine::kVideoCodecPrefs[] = {
-    {kVp8PayloadName, 100, 0},
-#ifndef WEBRTC_VIDEO_AVPF_NACK_ONLY
-    {kRedPayloadName, 101, 1},
-    {kFecPayloadName, 102, 2},
-#endif
-};
-
-// The formats are sorted by the descending order of width. We use the order to
-// find the next format for CPU and bandwidth adaptation.
-const VideoFormatPod WebRtcVideoEngine::kVideoFormats[] = {
-  {1280, 800, 30, FOURCC_ANY},
-  {1280, 720, 30, FOURCC_ANY},
-  {960, 600, 30, FOURCC_ANY},
-  {960, 540, 30, FOURCC_ANY},
-  {640, 400, 30, FOURCC_ANY},
-  {640, 360, 30, FOURCC_ANY},
-  {640, 480, 30, FOURCC_ANY},
-  {480, 300, 30, FOURCC_ANY},
-  {480, 270, 30, FOURCC_ANY},
-  {480, 360, 30, FOURCC_ANY},
-  {320, 200, 30, FOURCC_ANY},
-  {320, 180, 30, FOURCC_ANY},
-  {320, 240, 30, FOURCC_ANY},
-  {240, 150, 30, FOURCC_ANY},
-  {240, 135, 30, FOURCC_ANY},
-  {240, 180, 30, FOURCC_ANY},
-  {160, 100, 30, FOURCC_ANY},
-  {160, 90, 30, FOURCC_ANY},
-  {160, 120, 30, FOURCC_ANY},
-};
-
-const VideoFormatPod WebRtcVideoEngine::kDefaultVideoFormat =
-    {640, 400, 30, FOURCC_ANY};
-
-WebRtcVideoEngine::WebRtcVideoEngine() {
-  Construct(new ViEWrapper(), new ViETraceWrapper(), NULL);
-}
-
-WebRtcVideoEngine::WebRtcVideoEngine(WebRtcVoiceEngine* voice_engine,
-                                     ViEWrapper* vie_wrapper) {
-  Construct(vie_wrapper, new ViETraceWrapper(), voice_engine);
-}
-
-WebRtcVideoEngine::WebRtcVideoEngine(WebRtcVoiceEngine* voice_engine,
-                                     ViEWrapper* vie_wrapper,
-                                     ViETraceWrapper* tracing) {
-  Construct(vie_wrapper, tracing, voice_engine);
-}
-
-void WebRtcVideoEngine::Construct(ViEWrapper* vie_wrapper,
-                                  ViETraceWrapper* tracing,
-                                  WebRtcVoiceEngine* voice_engine) {
-  LOG(LS_INFO) << "WebRtcVideoEngine::WebRtcVideoEngine";
-  vie_wrapper_.reset(vie_wrapper);
-  vie_wrapper_base_initialized_ = false;
-  tracing_.reset(tracing);
-  voice_engine_ = voice_engine;
-  initialized_ = false;
-  log_level_ = kDefaultLogSeverity;
-  render_module_.reset(new WebRtcPassthroughRender());
-  local_renderer_w_ = local_renderer_h_ = 0;
-  local_renderer_ = NULL;
-  owns_capturer_ = false;
-  video_capturer_ = NULL;
-  capture_started_ = false;
-
-  ApplyLogging();
-  if (tracing_->SetTraceCallback(this) != 0) {
-    LOG_RTCERR1(SetTraceCallback, this);
-  }
-
-  // Set default quality levels for our supported codecs. We override them here
-  // if we know your cpu performance is low, and they can be updated explicitly
-  // by calling SetDefaultCodec.  For example by a flute preference setting, or
-  // by the server with a jec in response to our reported system info.
-  VideoCodec max_codec(kVideoCodecPrefs[0].payload_type,
-                       kVideoCodecPrefs[0].name,
-                       kDefaultVideoFormat.width,
-                       kDefaultVideoFormat.height,
-                       kDefaultVideoFormat.framerate,
-                       0);
-  if (!SetDefaultCodec(max_codec)) {
-    LOG(LS_ERROR) << "Failed to initialize list of supported codec types";
-  }
-}
-
-WebRtcVideoEngine::~WebRtcVideoEngine() {
-  ClearCapturer();
-  LOG(LS_INFO) << "WebRtcVideoEngine::~WebRtcVideoEngine";
-  if (initialized_) {
-    Terminate();
-  }
-  tracing_->SetTraceCallback(NULL);
-}
-
-bool WebRtcVideoEngine::Init() {
-  LOG(LS_INFO) << "WebRtcVideoEngine::Init";
-  bool result = InitVideoEngine();
-  if (result) {
-    LOG(LS_INFO) << "VideoEngine Init done";
-  } else {
-    LOG(LS_ERROR) << "VideoEngine Init failed, releasing";
-    Terminate();
-  }
-  return result;
-}
-
-bool WebRtcVideoEngine::InitVideoEngine() {
-  LOG(LS_INFO) << "WebRtcVideoEngine::InitVideoEngine";
-
-  // Init WebRTC VideoEngine.
-  if (!vie_wrapper_base_initialized_) {
-    if (vie_wrapper_->base()->Init() != 0) {
-      LOG_RTCERR0(Init);
-      return false;
-    }
-    vie_wrapper_base_initialized_ = true;
-  }
-
-  // Log the VoiceEngine version info.
-  char buffer[1024] = "";
-  if (vie_wrapper_->base()->GetVersion(buffer) != 0) {
-    LOG_RTCERR0(GetVersion);
-    return false;
-  }
-
-  LOG(LS_INFO) << "WebRtc VideoEngine Version:";
-  LogMultiline(talk_base::LS_INFO, buffer);
-
-  // Hook up to VoiceEngine for sync purposes, if supplied.
-  if (!voice_engine_) {
-    LOG(LS_WARNING) << "NULL voice engine";
-  } else if ((vie_wrapper_->base()->SetVoiceEngine(
-      voice_engine_->voe()->engine())) != 0) {
-    LOG_RTCERR0(SetVoiceEngine);
-    return false;
-  }
-
-  // Register for callbacks from the engine.
-  if ((vie_wrapper_->base()->RegisterObserver(*this)) != 0) {
-    LOG_RTCERR0(RegisterObserver);
-    return false;
-  }
-
-  // Register our custom render module.
-  if (vie_wrapper_->render()->RegisterVideoRenderModule(
-      *render_module_.get()) != 0) {
-    LOG_RTCERR0(RegisterVideoRenderModule);
-    return false;
-  }
-
-  initialized_ = true;
-  return true;
-}
-
-void WebRtcVideoEngine::Terminate() {
-  LOG(LS_INFO) << "WebRtcVideoEngine::Terminate";
-  initialized_ = false;
-  SetCapture(false);
-
-  if (vie_wrapper_->render()->DeRegisterVideoRenderModule(
-      *render_module_.get()) != 0) {
-    LOG_RTCERR0(DeRegisterVideoRenderModule);
-  }
-
-  if (vie_wrapper_->base()->DeregisterObserver() != 0) {
-    LOG_RTCERR0(DeregisterObserver);
-  }
-
-  if (vie_wrapper_->base()->SetVoiceEngine(NULL) != 0) {
-    LOG_RTCERR0(SetVoiceEngine);
-  }
-}
-
-int WebRtcVideoEngine::GetCapabilities() {
-  return VIDEO_RECV | VIDEO_SEND;
-}
-
-bool WebRtcVideoEngine::SetOptions(int options) {
-  return true;
-}
-
-bool WebRtcVideoEngine::SetDefaultEncoderConfig(
-    const VideoEncoderConfig& config) {
-  return SetDefaultCodec(config.max_codec);
-}
-
-// SetDefaultCodec may be called while the capturer is running. For example, a
-// test call is started in a page with QVGA default codec, and then a real call
-// is started in another page with VGA default codec. This is the corner case
-// and happens only when a session is started. We ignore this case currently.
-bool WebRtcVideoEngine::SetDefaultCodec(const VideoCodec& codec) {
-  if (!RebuildCodecList(codec)) {
-    LOG(LS_WARNING) << "Failed to RebuildCodecList";
-    return false;
-  }
-
-  default_codec_format_ = VideoFormat(
-      video_codecs_[0].width,
-      video_codecs_[0].height,
-      VideoFormat::FpsToInterval(video_codecs_[0].framerate),
-      FOURCC_ANY);
-  return true;
-}
-
-WebRtcVideoMediaChannel* WebRtcVideoEngine::CreateChannel(
-    VoiceMediaChannel* voice_channel) {
-  WebRtcVideoMediaChannel* channel =
-      new WebRtcVideoMediaChannel(this, voice_channel);
-  if (!channel->Init()) {
-    delete channel;
-    channel = NULL;
-  }
-  return channel;
-}
-
-bool WebRtcVideoEngine::SetCaptureDevice(const Device* device) {
-  if (!device) {
-    ClearCapturer();
-    LOG(LS_INFO) << "Camera set to NULL";
-    return true;
-  }
-  // No-op if the device hasn't changed.
-  if ((video_capturer_ != NULL) && video_capturer_->GetId() == device->id) {
-    return true;
-  }
-  // Create a new capturer for the specified device.
-  VideoCapturer* capturer = CreateVideoCapturer(*device);
-  if (!capturer) {
-    LOG(LS_ERROR) << "Failed to create camera '" << device->name << "', id='"
-                  << device->id << "'";
-    return false;
-  }
-  const bool owns_capturer = true;
-  if (!SetCapturer(capturer, owns_capturer)) {
-    return false;
-  }
-  LOG(LS_INFO) << "Camera set to '" << device->name << "', id='"
-               << device->id << "'";
-  return true;
-}
-
-bool WebRtcVideoEngine::SetCaptureModule(webrtc::VideoCaptureModule* vcm) {
-  if (!vcm) {
-    if ((video_capturer_ != NULL) && video_capturer_->IsRunning()) {
-      LOG(LS_WARNING) << "Failed to set camera to NULL when is running.";
-      return false;
-    } else {
-      ClearCapturer();
-      LOG(LS_INFO) << "Camera set to NULL";
-      return true;
-    }
-  }
-  // Create a new capturer for the specified device.
-  WebRtcVideoCapturer* capturer = new WebRtcVideoCapturer;
-  if (!capturer->Init(vcm)) {
-    LOG(LS_ERROR) << "Failed to create camera from VCM";
-    delete capturer;
-    return false;
-  }
-  const bool owns_capturer = true;
-  if (!SetCapturer(capturer, owns_capturer)) {
-    return false;
-  }
-  LOG(LS_INFO) << "Camera created with VCM";
-  CaptureResult ret = SetCapture(true);
-  if (ret != cricket::CR_SUCCESS && ret != cricket::CR_PENDING) {
-    return false;
-  }  
-  return true;
-}
-
-bool WebRtcVideoEngine::SetVideoCapturer(VideoCapturer* capturer,
-                                         uint32 /*ssrc*/) {
-  const bool capture = (capturer != NULL);
-  const bool owns_capturer = false;
-  CaptureResult res = CR_FAILURE;
-  if (capture) {
-    // Register the capturer before starting to capture.
-    if (!SetCapturer(capturer, owns_capturer)) {
-      return false;
-    }
-    const bool kEnableCapture = true;
-    res = SetCapture(kEnableCapture);
-  } else {
-    // Stop capturing before unregistering the capturer.
-    const bool kDisableCapture = false;
-    res = SetCapture(kDisableCapture);
-    if (!SetCapturer(capturer, owns_capturer)) {
-      return false;
-    }
-  }
-  return (res == CR_SUCCESS) || (res == CR_PENDING);
-}
-
-bool WebRtcVideoEngine::SetLocalRenderer(VideoRenderer* renderer) {
-  local_renderer_w_ = local_renderer_h_ = 0;
-  local_renderer_ = renderer;
-  return true;
-}
-
-CaptureResult WebRtcVideoEngine::SetCapture(bool capture) {
-  bool old_capture = capture_started_;
-  capture_started_ = capture;
-  CaptureResult res = UpdateCapturingState();
-  if (res != CR_SUCCESS && res != CR_PENDING) {
-    capture_started_ = old_capture;
-  }
-  return res;
-}
-
-VideoCapturer* WebRtcVideoEngine::CreateVideoCapturer(const Device& device) {
-  WebRtcVideoCapturer* capturer = new WebRtcVideoCapturer;
-  if (!capturer->Init(device)) {
-    delete capturer;
-    return NULL;
-  }
-  return capturer;
-}
-
-CaptureResult WebRtcVideoEngine::UpdateCapturingState() {
-  CaptureResult result = CR_SUCCESS;
-
-  bool capture = capture_started_;
-  if (!IsCapturing() && capture) {  // Start capturing.
-    if (video_capturer_ == NULL) {
-      return CR_NO_DEVICE;
-    }
-
-    VideoFormat capture_format;
-    if (!video_capturer_->GetBestCaptureFormat(default_codec_format_,
-                                               &capture_format)) {
-      LOG(LS_WARNING) << "Unsupported format:"
-                      << " width=" << default_codec_format_.width
-                      << " height=" << default_codec_format_.height
-                      << ". Supported formats are:";
-      const std::vector<VideoFormat>* formats =
-          video_capturer_->GetSupportedFormats();
-      if (formats) {
-        for (std::vector<VideoFormat>::const_iterator i = formats->begin();
-             i != formats->end(); ++i) {
-          const VideoFormat& format = *i;
-          LOG(LS_WARNING) << "  " << GetFourccName(format.fourcc) << ":"
-                          << format.width << "x" << format.height << "x"
-                          << format.framerate();
-        }
-      }
-      return CR_FAILURE;
-    }
-
-    // Start the video capturer.
-    result = video_capturer_->Start(capture_format);
-    if (CR_SUCCESS != result && CR_PENDING != result) {
-      LOG(LS_ERROR) << "Failed to start the video capturer";
-      return result;
-    }
-  } else if (IsCapturing() && !capture) {  // Stop capturing.
-    video_capturer_->Stop();
-  }
-
-  return result;
-}
-
-bool WebRtcVideoEngine::IsCapturing() const {
-  return (video_capturer_ != NULL) && video_capturer_->IsRunning();
-}
-
-void WebRtcVideoEngine::OnFrameCaptured(VideoCapturer* capturer,
-                                        const CapturedFrame* frame) {
-  // Force 16:10 for now. We'll be smarter with the capture refactor.
-  int cropped_height = frame->width * default_codec_format_.height
-      / default_codec_format_.width;
-  if (cropped_height > frame->height) {
-    // TODO: Once we support horizontal cropping, add cropped_width.
-    cropped_height = frame->height;
-  }
-
-  // This CapturedFrame* will already be in I420. In the future, when
-  // WebRtcVideoFrame has support for independent planes, we can just attach
-  // to it and update the pointers when cropping.
-  WebRtcVideoFrame i420_frame;
-  if (!i420_frame.Init(frame, frame->width, cropped_height)) {
-    LOG(LS_ERROR) << "Couldn't convert to I420! "
-                  << frame->width << " x " << cropped_height;
-    return;
-  }
-
-  // Send I420 frame to the local renderer.
-  if (local_renderer_) {
-    if (local_renderer_w_ != static_cast<int>(i420_frame.GetWidth()) ||
-        local_renderer_h_ != static_cast<int>(i420_frame.GetHeight())) {
-      local_renderer_->SetSize(local_renderer_w_ = i420_frame.GetWidth(),
-                               local_renderer_h_ = i420_frame.GetHeight(), 0);
-    }
-    local_renderer_->RenderFrame(&i420_frame);
-  }
-
-  // Send I420 frame to the registered senders.
-  talk_base::CritScope cs(&channels_crit_);
-  for (VideoChannels::iterator it = channels_.begin();
-      it != channels_.end(); ++it) {
-    if ((*it)->sending()) (*it)->SendFrame(0, &i420_frame);
-  }
-}
-
-const std::vector<VideoCodec>& WebRtcVideoEngine::codecs() const {
-  return video_codecs_;
-}
-
-void WebRtcVideoEngine::SetLogging(int min_sev, const char* filter) {
-  log_level_ = min_sev;
-  ApplyLogging();
-}
-
-int WebRtcVideoEngine::GetLastEngineError() {
-  return vie_wrapper_->error();
-}
-
-// Checks to see whether we comprehend and could receive a particular codec
-bool WebRtcVideoEngine::FindCodec(const VideoCodec& in) {
-  for (int i = 0; i < ARRAY_SIZE(kVideoFormats); ++i) {
-    const VideoFormat fmt(kVideoFormats[i]);
-    if ((in.width == 0 && in.height == 0) ||
-        (fmt.width == in.width && fmt.height == in.height)) {
-      for (int j = 0; j < ARRAY_SIZE(kVideoCodecPrefs); ++j) {
-        VideoCodec codec(kVideoCodecPrefs[j].payload_type,
-                         kVideoCodecPrefs[j].name, 0, 0, 0, 0);
-        if (codec.Matches(in)) {
-          return true;
-        }
-      }
-    }
-  }
-  return false;
-}
-
-// Given the requested codec, returns true if we can send that codec type and
-// updates out with the best quality we could send for that codec. If current is
-// not empty, we constrain out so that its aspect ratio matches current's.
-bool WebRtcVideoEngine::CanSendCodec(const VideoCodec& requested,
-                                     const VideoCodec& current,
-                                     VideoCodec* out) {
-  if (!out) {
-    return false;
-  }
-
-  std::vector<VideoCodec>::const_iterator local_max;
-  for (local_max = video_codecs_.begin();
-       local_max < video_codecs_.end();
-       ++local_max) {
-    // First match codecs by payload type
-    if (!requested.Matches(local_max->id, local_max->name)) {
-      continue;
-    }
-
-    out->id = requested.id;
-    out->name = requested.name;
-    out->preference = requested.preference;
-    out->framerate = talk_base::_min(requested.framerate, local_max->framerate);
-    out->width = 0;
-    out->height = 0;
-
-    if (0 == requested.width && 0 == requested.height) {
-      // Special case with resolution 0. The channel should not send frames.
-      return true;
-    } else if (0 == requested.width || 0 == requested.height) {
-      // 0xn and nx0 are invalid resolutions.
-      return false;
-    }
-
-    // Pick the best quality that is within their and our bounds and has the
-    // correct aspect ratio.
-    for (int j = 0; j < ARRAY_SIZE(kVideoFormats); ++j) {
-      const VideoFormat format(kVideoFormats[j]);
-
-      // Skip any format that is larger than the local or remote maximums, or
-      // smaller than the current best match
-      if (format.width > requested.width || format.height > requested.height ||
-          format.width > local_max->width ||
-          (format.width < out->width && format.height < out->height)) {
-        continue;
-      }
-
-      bool better = false;
-
-      // Check any further constraints on this prospective format
-      if (!out->width || !out->height) {
-        // If we don't have any matches yet, this is the best so far.
-        better = true;
-      } else if (current.width && current.height) {
-        // current is set so format must match its ratio exactly.
-        better =
-            (format.width * current.height == format.height * current.width);
-      } else {
-        // Prefer closer aspect ratios i.e
-        // format.aspect - requested.aspect < out.aspect - requested.aspect
-        better = abs(format.width * requested.height * out->height -
-                     requested.width * format.height * out->height) <
-                 abs(out->width * format.height * requested.height -
-                     requested.width * format.height * out->height);
-      }
-
-      if (better) {
-        out->width = format.width;
-        out->height = format.height;
-      }
-    }
-    if (out->width > 0) {
-      return true;
-    }
-  }
-  return false;
-}
-
-void WebRtcVideoEngine::ConvertToCricketVideoCodec(
-    const webrtc::VideoCodec& in_codec, VideoCodec& out_codec) {
-  out_codec.id = in_codec.plType;
-  out_codec.name = in_codec.plName;
-  out_codec.width = in_codec.width;
-  out_codec.height = in_codec.height;
-  out_codec.framerate = in_codec.maxFramerate;
-}
-
-bool WebRtcVideoEngine::ConvertFromCricketVideoCodec(
-    const VideoCodec& in_codec, webrtc::VideoCodec& out_codec) {
-  bool found = false;
-  int ncodecs = vie_wrapper_->codec()->NumberOfCodecs();
-  for (int i = 0; i < ncodecs; ++i) {
-    if (vie_wrapper_->codec()->GetCodec(i, out_codec) == 0 &&
-      in_codec.name == out_codec.plName) {
-      found = true;
-      break;
-    }
-  }
-
-  if (!found) {
-    LOG(LS_ERROR) << "invalid codec type";
-    return false;
-  }
-
-  if (in_codec.id != 0)
-    out_codec.plType = in_codec.id;
-
-  if (in_codec.width != 0)
-    out_codec.width = in_codec.width;
-
-  if (in_codec.height != 0)
-    out_codec.height = in_codec.height;
-
-  if (in_codec.framerate != 0)
-    out_codec.maxFramerate = in_codec.framerate;
-
-  // Init the codec with the default bandwidth options.
-  out_codec.minBitrate = kMinVideoBitrate;
-  out_codec.startBitrate = kStartVideoBitrate;
-  out_codec.maxBitrate = kMaxVideoBitrate;
-
-  return true;
-}
-
-void WebRtcVideoEngine::RegisterChannel(WebRtcVideoMediaChannel *channel) {
-  talk_base::CritScope cs(&channels_crit_);
-  channels_.push_back(channel);
-}
-
-void WebRtcVideoEngine::UnregisterChannel(WebRtcVideoMediaChannel *channel) {
-  talk_base::CritScope cs(&channels_crit_);
-  channels_.erase(std::remove(channels_.begin(), channels_.end(), channel),
-                  channels_.end());
-}
-
-bool WebRtcVideoEngine::SetVoiceEngine(WebRtcVoiceEngine* voice_engine) {
-  if (initialized_) {
-    LOG(LS_WARNING) << "SetVoiceEngine can not be called after Init.";
-    return false;
-  }
-  voice_engine_ = voice_engine;
-  return true;
-}
-
-bool WebRtcVideoEngine::EnableTimedRender() {
-  if (initialized_) {
-    LOG(LS_WARNING) << "EnableTimedRender can not be called after Init.";
-    return false;
-  }
-  render_module_.reset(webrtc::VideoRender::CreateVideoRender(0, NULL,
-      false, webrtc::kRenderExternal));
-  return true;
-}
-
-void WebRtcVideoEngine::ApplyLogging() {
-  int filter = 0;
-  switch (log_level_) {
-    case talk_base::LS_VERBOSE: filter |= webrtc::kTraceAll;
-    case talk_base::LS_INFO: filter |= webrtc::kTraceStateInfo;
-    case talk_base::LS_WARNING: filter |= webrtc::kTraceWarning;
-    case talk_base::LS_ERROR: filter |=
-        webrtc::kTraceError | webrtc::kTraceCritical;
-  }
-  tracing_->SetTraceFilter(filter);
-}
-
-// Rebuilds the codec list to be only those that are less intensive
-// than the specified codec.
-bool WebRtcVideoEngine::RebuildCodecList(const VideoCodec& in_codec) {
-  if (!FindCodec(in_codec))
-    return false;
-
-  video_codecs_.clear();
-
-  bool found = false;
-  for (size_t i = 0; i < ARRAY_SIZE(kVideoCodecPrefs); ++i) {
-    const VideoCodecPref& pref(kVideoCodecPrefs[i]);
-    if (!found)
-      found = (in_codec.name == pref.name);
-    if (found) {
-      VideoCodec codec(pref.payload_type, pref.name,
-                       in_codec.width, in_codec.height, in_codec.framerate,
-                       ARRAY_SIZE(kVideoCodecPrefs) - i);
-      video_codecs_.push_back(codec);
-    }
-  }
-  ASSERT(found);
-  return true;
-}
-
-bool WebRtcVideoEngine::SetCapturer(VideoCapturer* capturer,
-                                    bool own_capturer) {
-  if (capturer == NULL) {
-    ClearCapturer();
-    return true;
-  }
-  // Hook up signals and install the supplied capturer.
-  SignalCaptureResult.repeat(capturer->SignalStartResult);
-  capturer->SignalFrameCaptured.connect(this,
-      &WebRtcVideoEngine::OnFrameCaptured);
-  ClearCapturer();
-  video_capturer_ = capturer;
-  owns_capturer_ = own_capturer;
-  // Possibly restart the capturer if it is supposed to be running.
-  CaptureResult result = UpdateCapturingState();
-  if (result != CR_SUCCESS && result != CR_PENDING) {
-    LOG(LS_WARNING) << "Camera failed to restart";
-    return false;
-  }
-  return true;
-}
-
-void WebRtcVideoEngine::PerformanceAlarm(const unsigned int cpu_load) {
-  LOG(LS_INFO) << "WebRtcVideoEngine::PerformanceAlarm";
-}
-
-// Ignore spammy trace messages, mostly from the stats API when we haven't
-// gotten RTCP info yet from the remote side.
-bool WebRtcVideoEngine::ShouldIgnoreTrace(const std::string& trace) {
-  static const char* const kTracesToIgnore[] = {
-    NULL
-  };
-  for (const char* const* p = kTracesToIgnore; *p; ++p) {
-    if (trace.find(*p) == 0) {
-      return true;
-    }
-  }
-  return false;
-}
-
-int WebRtcVideoEngine::GetNumOfChannels() {
-  talk_base::CritScope cs(&channels_crit_);
-  return channels_.size();
-}
-
-void WebRtcVideoEngine::Print(const webrtc::TraceLevel level,
-                              const char* trace, const int length) {
-  talk_base::LoggingSeverity sev = talk_base::LS_VERBOSE;
-  if (level == webrtc::kTraceError || level == webrtc::kTraceCritical)
-    sev = talk_base::LS_ERROR;
-  else if (level == webrtc::kTraceWarning)
-    sev = talk_base::LS_WARNING;
-  else if (level == webrtc::kTraceStateInfo || level == webrtc::kTraceInfo)
-    sev = talk_base::LS_INFO;
-
-  if (sev >= log_level_) {
-    // Skip past boilerplate prefix text
-    if (length < 72) {
-      std::string msg(trace, length);
-      LOG(LS_ERROR) << "Malformed webrtc log message: ";
-      LOG_V(sev) << msg;
-    } else {
-      std::string msg(trace + 71, length - 72);
-      if (!ShouldIgnoreTrace(msg) &&
-          (!voice_engine_ || !voice_engine_->ShouldIgnoreTrace(msg))) {
-        LOG_V(sev) << "WebRtc:" << msg;
-      }
-    }
-  }
-}
-
-// TODO: stubs for now
-bool WebRtcVideoEngine::RegisterProcessor(
-    VideoProcessor* video_processor) {
-  return true;
-}
-bool WebRtcVideoEngine::UnregisterProcessor(
-    VideoProcessor* video_processor) {
-  return true;
-}
-
-void WebRtcVideoEngine::ClearCapturer() {
-  if (owns_capturer_) {
-    delete video_capturer_;
-  }
-  video_capturer_ = NULL;
-}
-
-// WebRtcVideoMediaChannel
-
-WebRtcVideoMediaChannel::WebRtcVideoMediaChannel(
-    WebRtcVideoEngine* engine, VoiceMediaChannel* channel)
-    : engine_(engine),
-      voice_channel_(channel),
-      vie_channel_(-1),
-      vie_capture_(-1),
-      external_capture_(NULL),
-      sending_(false),
-      render_started_(false),
-      muted_(false),
-      send_min_bitrate_(kMinVideoBitrate),
-      send_start_bitrate_(kStartVideoBitrate),
-      send_max_bitrate_(kMaxVideoBitrate),
-      local_stream_info_(new LocalStreamInfo()) {
-  engine->RegisterChannel(this);
-}
-
-bool WebRtcVideoMediaChannel::Init() {
-  if (engine_->vie()->base()->CreateChannel(vie_channel_) != 0) {
-    LOG_RTCERR1(CreateChannel, vie_channel_);
-    return false;
-  }
-
-  LOG(LS_INFO) << "WebRtcVideoMediaChannel::Init "
-               << "vie_channel " << vie_channel_ << " created";
-
-  // Connect the voice channel, if there is one.
-  if (voice_channel_) {
-    WebRtcVoiceMediaChannel* channel =
-        static_cast<WebRtcVoiceMediaChannel*>(voice_channel_);
-    if (engine_->vie()->base()->ConnectAudioChannel(
-        vie_channel_, channel->voe_channel()) != 0) {
-      LOG_RTCERR2(ConnectAudioChannel, vie_channel_, channel->voe_channel());
-      LOG(LS_WARNING) << "A/V not synchronized";
-      // Not a fatal error.
-    }
-  }
-
-  // Register external transport.
-  if (engine_->vie()->network()->RegisterSendTransport(
-      vie_channel_, *this) != 0) {
-    LOG_RTCERR1(RegisterSendTransport, vie_channel_);
-    return false;
-  }
-
-  // Set MTU.
-  if (engine_->vie()->network()->SetMTU(vie_channel_, kVideoMtu) != 0) {
-    LOG_RTCERR2(SetMTU, vie_channel_, kVideoMtu);
-    return false;
-  }
-
-  // Register external capture.
-  if (engine()->vie()->capture()->AllocateExternalCaptureDevice(
-      vie_capture_, external_capture_) != 0) {
-    LOG_RTCERR0(AllocateExternalCaptureDevice);
-    return false;
-  }
-
-  // Connect external capture.
-  if (engine()->vie()->capture()->ConnectCaptureDevice(
-      vie_capture_, vie_channel_) != 0) {
-    LOG_RTCERR2(ConnectCaptureDevice, vie_capture_, vie_channel_);
-    return false;
-  }
-
-  // Install render adapter.
-  remote_renderer_.reset(new WebRtcRenderAdapter(NULL));
-  if (engine_->vie()->render()->AddRenderer(vie_channel_,
-      webrtc::kVideoI420, remote_renderer_.get()) != 0) {
-    LOG_RTCERR3(AddRenderer, vie_channel_, webrtc::kVideoI420,
-                remote_renderer_.get());
-    remote_renderer_.reset();
-    return false;
-  }
-
-  // Register decoder observer for incoming framerate and bitrate.
-  decoder_observer_.reset(new WebRtcDecoderObserver(vie_channel_));
-  if (engine()->vie()->codec()->RegisterDecoderObserver(
-      vie_channel_, *decoder_observer_) != 0) {
-    LOG_RTCERR1(RegisterDecoderObserver, decoder_observer_.get());
-    return false;
-  }
-
-  // Register encoder observer for outgoing framerate and bitrate.
-  encoder_observer_.reset(new WebRtcEncoderObserver(vie_channel_));
-  if (engine()->vie()->codec()->RegisterEncoderObserver(
-      vie_channel_, *encoder_observer_) != 0) {
-    LOG_RTCERR1(RegisterEncoderObserver, encoder_observer_.get());
-    return false;
-  }
-
-  // Turn on RTCP and loss feedback reporting.
-  if (!EnableRtcp() ||
-      !EnablePli()) {
-    return false;
-  }
-
-#ifdef WEBRTC_VIDEO_AVPF_NACK_ONLY
-  // Turn on NACK-only loss handling.
-  if (!EnableNack())
-    return false;
-#endif
-
-  // Turn on TMMBR-based BWE reporting.
-  if (!EnableTmmbr()) {
-    return false;
-  }
-
-  return true;
-}
-
-WebRtcVideoMediaChannel::~WebRtcVideoMediaChannel() {
-  if (vie_channel_ != -1) {
-    // Stop sending.
-    SetSend(false);
-    if (engine()->vie()->codec()->DeregisterEncoderObserver(
-        vie_channel_) != 0) {
-      LOG_RTCERR1(DeregisterEncoderObserver, vie_channel_);
-    }
-
-    // Stop the renderer.
-    SetRender(false);
-    if (engine()->vie()->codec()->DeregisterDecoderObserver(
-        vie_channel_) != 0) {
-      LOG_RTCERR1(DeregisterDecoderObserver, vie_channel_);
-    }
-    if (remote_renderer_.get() &&
-        engine()->vie()->render()->RemoveRenderer(vie_channel_) != 0) {
-      LOG_RTCERR1(RemoveRenderer, vie_channel_);
-    }
-
-    // Destroy the external capture interface.
-    if (vie_capture_ != -1) {
-      if (engine()->vie()->capture()->DisconnectCaptureDevice(
-          vie_channel_) != 0) {
-        LOG_RTCERR1(DisconnectCaptureDevice, vie_channel_);
-      }
-      if (engine()->vie()->capture()->ReleaseCaptureDevice(
-          vie_capture_) != 0) {
-        LOG_RTCERR1(ReleaseCaptureDevice, vie_capture_);
-      }
-    }
-
-    // Deregister external transport.
-    if (engine()->vie()->network()->DeregisterSendTransport(
-        vie_channel_) != 0) {
-      LOG_RTCERR1(DeregisterSendTransport, vie_channel_);
-    }
-
-    // Delete the VideoEngine channel.
-    if (engine()->vie()->base()->DeleteChannel(vie_channel_) != 0) {
-      LOG_RTCERR1(DeleteChannel, vie_channel_);
-    }
-  }
-
-  // Unregister the channel from the engine.
-  engine()->UnregisterChannel(this);
-}
-
-bool WebRtcVideoMediaChannel::SetRecvCodecs(
-    const std::vector<VideoCodec>& codecs) {
-  bool ret = true;
-  for (std::vector<VideoCodec>::const_iterator iter = codecs.begin();
-      iter != codecs.end(); ++iter) {
-    if (engine()->FindCodec(*iter)) {
-      webrtc::VideoCodec wcodec;
-      if (engine()->ConvertFromCricketVideoCodec(*iter, wcodec)) {
-        if (engine()->vie()->codec()->SetReceiveCodec(
-            vie_channel_, wcodec) != 0) {
-          LOG_RTCERR2(SetReceiveCodec, vie_channel_, wcodec.plName);
-          ret = false;
-        }
-      }
-    } else {
-      LOG(LS_INFO) << "Unknown codec " << iter->name;
-      ret = false;
-    }
-  }
-
-  // make channel ready to receive packets
-  if (ret) {
-    if (engine()->vie()->base()->StartReceive(vie_channel_) != 0) {
-      LOG_RTCERR1(StartReceive, vie_channel_);
-      ret = false;
-    }
-  }
-  return ret;
-}
-
-bool WebRtcVideoMediaChannel::SetSendCodecs(
-    const std::vector<VideoCodec>& codecs) {
-  // Match with local video codec list.
-  std::vector<webrtc::VideoCodec> send_codecs;
-  int red_type = -1, fec_type = -1;
-  VideoCodec checked_codec;
-  VideoCodec current;  // defaults to 0x0
-  if (sending_) {
-    engine()->ConvertToCricketVideoCodec(*send_codec_, current);
-  }
-  for (std::vector<VideoCodec>::const_iterator iter = codecs.begin();
-      iter != codecs.end(); ++iter) {
-    if (_stricmp(iter->name.c_str(), kRedPayloadName) == 0) {
-      red_type = iter->id;
-    } else if (_stricmp(iter->name.c_str(), kFecPayloadName) == 0) {
-      fec_type = iter->id;
-    } else if (engine()->CanSendCodec(*iter, current, &checked_codec)) {
-      webrtc::VideoCodec wcodec;
-      if (engine()->ConvertFromCricketVideoCodec(checked_codec, wcodec)) {
-        send_codecs.push_back(wcodec);
-      }
-    } else {
-      LOG(LS_WARNING) << "Unknown codec " << iter->name;
-    }
-  }
-
-  // Fail if we don't have a match.
-  if (send_codecs.empty()) {
-    LOG(LS_WARNING) << "No matching codecs avilable";
-    return false;
-  }
-
-#ifndef WEBRTC_VIDEO_AVPF_NACK_ONLY
-  // Configure FEC if enabled.
-  if (!SetNackFec(red_type, fec_type)) {
-    return false;
-  }
-#endif
-
-  // Select the first matched codec.
-  webrtc::VideoCodec& codec(send_codecs[0]);
-
-  // Set the default number of temporal layers for VP8.
-  if (webrtc::kVideoCodecVP8 == codec.codecType) {
-    codec.codecSpecific.VP8.numberOfTemporalLayers =
-        kDefaultNumberOfTemporalLayers;
-  }
-
-  if (!SetSendCodec(
-      codec, send_min_bitrate_, send_start_bitrate_, send_max_bitrate_)) {
-    return false;
-  }
-
-  LOG(LS_INFO) << "Selected video codec " << send_codec_->plName << "/"
-               << send_codec_->width << "x" << send_codec_->height << "x"
-               << static_cast<int>(send_codec_->maxFramerate);
-  if (webrtc::kVideoCodecVP8 == codec.codecType) {
-    LOG(LS_INFO) << "VP8 number of layers: "
-                 << static_cast<int>(
-                    send_codec_->codecSpecific.VP8.numberOfTemporalLayers);
-  }
-  return true;
-}
-
-bool WebRtcVideoMediaChannel::SetRender(bool render) {
-  if (render == render_started_) {
-    return true;  // no action required
-  }
-
-  bool ret = true;
-  if (render) {
-    if (engine()->vie()->render()->StartRender(vie_channel_) != 0) {
-      LOG_RTCERR1(StartRender, vie_channel_);
-      ret = false;
-    }
-  } else {
-    if (engine()->vie()->render()->StopRender(vie_channel_) != 0) {
-      LOG_RTCERR1(StopRender, vie_channel_);
-      ret = false;
-    }
-  }
-  if (ret) {
-    render_started_ = render;
-  }
-
-  return ret;
-}
-
-bool WebRtcVideoMediaChannel::SetSend(bool send) {
-  if (send == sending()) {
-    return true;  // no action required
-  }
-
-  if (send) {
-    // We've been asked to start sending.
-    // SetSendCodecs must have been called already.
-    if (!send_codec_.get()) {
-      return false;
-    }
-
-    if (engine()->vie()->base()->StartSend(vie_channel_) != 0) {
-      LOG_RTCERR1(StartSend, vie_channel_);
-      return false;
-    }
-  } else {
-    // We've been asked to stop sending.
-    if (engine()->vie()->base()->StopSend(vie_channel_) != 0) {
-      LOG_RTCERR1(StopSend, vie_channel_);
-      return false;
-    }
-  }
-
-  sending_ = send;
-  return true;
-}
-
-bool WebRtcVideoMediaChannel::AddStream(uint32 ssrc, uint32 voice_ssrc) {
-  return false;
-}
-
-bool WebRtcVideoMediaChannel::RemoveStream(uint32 ssrc) {
-  return false;
-}
-
-bool WebRtcVideoMediaChannel::SetRenderer(
-    uint32 ssrc, VideoRenderer* renderer) {
-  if (ssrc != 0)
-    return false;
-
-  remote_renderer_->SetRenderer(renderer);
-  return true;
-}
-
-bool WebRtcVideoMediaChannel::GetStats(VideoMediaInfo* info) {
-  // Get basic statistics.
-  unsigned int bytes_sent, packets_sent, bytes_recv, packets_recv;
-  unsigned int ssrc;
-  if (engine_->vie()->rtp()->GetRTPStatistics(vie_channel_,
-          bytes_sent, packets_sent, bytes_recv, packets_recv) != 0) {
-    LOG_RTCERR1(GetRTPStatistics, vie_channel_);
-    return false;
-  }
-
-  // Get sender statistics and build VideoSenderInfo.
-  if (engine_->vie()->rtp()->GetLocalSSRC(vie_channel_, ssrc) == 0) {
-    VideoSenderInfo sinfo;
-    sinfo.ssrc = ssrc;
-    sinfo.codec_name = send_codec_.get() ? send_codec_->plName : "";
-    sinfo.bytes_sent = bytes_sent;
-    sinfo.packets_sent = packets_sent;
-    sinfo.packets_cached = -1;
-    sinfo.packets_lost = -1;
-    sinfo.fraction_lost = -1;
-    sinfo.firs_rcvd = -1;
-    sinfo.nacks_rcvd = -1;
-    sinfo.rtt_ms = -1;
-    sinfo.frame_width = local_stream_info_->width();
-    sinfo.frame_height = local_stream_info_->height();
-    sinfo.framerate_input = local_stream_info_->framerate();
-    sinfo.framerate_sent = encoder_observer_->framerate();
-    sinfo.nominal_bitrate = encoder_observer_->bitrate();
-    sinfo.preferred_bitrate = kMaxVideoBitrate;
-
-    // Get received RTCP statistics for the sender, if available.
-    // It's not a fatal error if we can't, since RTCP may not have arrived yet.
-    uint16 r_fraction_lost;
-    unsigned int r_cumulative_lost;
-    unsigned int r_extended_max;
-    unsigned int r_jitter;
-    int r_rtt_ms;
-    if (engine_->vie()->rtp()->GetReceivedRTCPStatistics(vie_channel_,
-            r_fraction_lost, r_cumulative_lost, r_extended_max,
-            r_jitter, r_rtt_ms) == 0) {
-      // Convert Q8 to float.
-      sinfo.packets_lost = r_cumulative_lost;
-      sinfo.fraction_lost = static_cast<float>(r_fraction_lost) / (1 << 8);
-      sinfo.rtt_ms = r_rtt_ms;
-    }
-    info->senders.push_back(sinfo);
-  } else {
-    LOG_RTCERR1(GetLocalSSRC, vie_channel_);
-  }
-
-  // Get receiver statistics and build VideoReceiverInfo, if we have data.
-  if (engine_->vie()->rtp()->GetRemoteSSRC(vie_channel_, ssrc) == 0) {
-    VideoReceiverInfo rinfo;
-    rinfo.ssrc = ssrc;
-    rinfo.bytes_rcvd = bytes_recv;
-    rinfo.packets_rcvd = packets_recv;
-    rinfo.packets_lost = -1;
-    rinfo.packets_concealed = -1;
-    rinfo.fraction_lost = -1;  // from SentRTCP
-    rinfo.firs_sent = decoder_observer_->firs_requested();
-    rinfo.nacks_sent = -1;
-    rinfo.frame_width = remote_renderer_->width();
-    rinfo.frame_height = remote_renderer_->height();
-    rinfo.framerate_rcvd = decoder_observer_->framerate();
-    int fps = remote_renderer_->framerate();
-    rinfo.framerate_decoded = fps;
-    rinfo.framerate_output = fps;
-
-    // Get sent RTCP statistics.
-    uint16 s_fraction_lost;
-    unsigned int s_cumulative_lost;
-    unsigned int s_extended_max;
-    unsigned int s_jitter;
-    int s_rtt_ms;
-    if (engine_->vie()->rtp()->GetSentRTCPStatistics(vie_channel_,
-            s_fraction_lost, s_cumulative_lost, s_extended_max,
-            s_jitter, s_rtt_ms) == 0) {
-      // Convert Q8 to float.
-      rinfo.packets_lost = s_cumulative_lost;
-      rinfo.fraction_lost = static_cast<float>(s_fraction_lost) / (1 << 8);
-    }
-    info->receivers.push_back(rinfo);
-  }
-
-  // Build BandwidthEstimationInfo.
-  // TODO: Fill in more BWE stats once we have them.
-  unsigned int total_bitrate_sent;
-  unsigned int video_bitrate_sent;
-  unsigned int fec_bitrate_sent;
-  unsigned int nack_bitrate_sent;
-  if (engine_->vie()->rtp()->GetBandwidthUsage(vie_channel_,
-      total_bitrate_sent, video_bitrate_sent,
-      fec_bitrate_sent, nack_bitrate_sent) == 0) {
-    BandwidthEstimationInfo bwe;
-    bwe.actual_enc_bitrate = video_bitrate_sent;
-    bwe.transmit_bitrate = total_bitrate_sent;
-    bwe.retransmit_bitrate = nack_bitrate_sent;
-    info->bw_estimations.push_back(bwe);
-  } else {
-    LOG_RTCERR1(GetBandwidthUsage, vie_channel_);
-  }
-
-  return true;
-}
-
-bool WebRtcVideoMediaChannel::SendIntraFrame() {
-  bool ret = true;
-  if (engine()->vie()->codec()->SendKeyFrame(vie_channel_) != 0) {
-    LOG_RTCERR1(SendKeyFrame, vie_channel_);
-    ret = false;
-  }
-
-  return ret;
-}
-
-bool WebRtcVideoMediaChannel::RequestIntraFrame() {
-  // There is no API exposed to application to request a key frame
-  // ViE does this internally when there are errors from decoder
-  return false;
-}
-
-void WebRtcVideoMediaChannel::OnPacketReceived(talk_base::Buffer* packet) {
-  engine()->vie()->network()->ReceivedRTPPacket(vie_channel_,
-                                                         packet->data(),
-                                                         packet->length());
-}
-
-void WebRtcVideoMediaChannel::OnRtcpReceived(talk_base::Buffer* packet) {
-  engine_->vie()->network()->ReceivedRTCPPacket(vie_channel_,
-                                                         packet->data(),
-                                                         packet->length());
-}
-
-void WebRtcVideoMediaChannel::SetSendSsrc(uint32 id) {
-  if (!sending_) {
-    if (engine()->vie()->rtp()->SetLocalSSRC(vie_channel_, id) != 0) {
-      LOG_RTCERR1(SetLocalSSRC, vie_channel_);
-    }
-  } else {
-    LOG(LS_ERROR) << "Channel already in send state";
-  }
-}
-
-bool WebRtcVideoMediaChannel::SetRtcpCName(const std::string& cname) {
-  if (engine()->vie()->rtp()->SetRTCPCName(vie_channel_,
-                                           cname.c_str()) != 0) {
-    LOG_RTCERR2(SetRTCPCName, vie_channel_, cname.c_str());
-    return false;
-  }
-  return true;
-}
-
-bool WebRtcVideoMediaChannel::Mute(bool on) {
-  muted_ = on;
-  return true;
-}
-
-bool WebRtcVideoMediaChannel::SetSendBandwidth(bool autobw, int bps) {
-  LOG(LS_INFO) << "RtcVideoMediaChanne::SetSendBandwidth";
-
-  if (!send_codec_.get()) {
-    LOG(LS_INFO) << "The send codec has not been set up yet.";
-    return true;
-  }
-
-  int min_bitrate;
-  int start_bitrate;
-  int max_bitrate;
-  if (autobw) {
-    // Use the default values for min bitrate.
-    min_bitrate = kMinVideoBitrate;
-    // Use the default value or the bps for the max
-    max_bitrate = (bps <= 0) ? kMaxVideoBitrate : (bps / 1000);
-    // Maximum start bitrate can be kStartVideoBitrate.
-    start_bitrate = talk_base::_min(kStartVideoBitrate, max_bitrate);
-  } else {
-    // Use the default start or the bps as the target bitrate.
-    int target_bitrate = (bps <= 0) ? kStartVideoBitrate : (bps / 1000);
-    min_bitrate = target_bitrate;
-    start_bitrate = target_bitrate;
-    max_bitrate = target_bitrate;
-  }
-
-  if (!SetSendCodec(*send_codec_, min_bitrate, start_bitrate, max_bitrate)) {
-    return false;
-  }
-
-  return true;
-}
-
-bool WebRtcVideoMediaChannel::SetOptions(int options) {
-  return true;
-}
-
-void WebRtcVideoMediaChannel::SetInterface(NetworkInterface* iface) {
-  MediaChannel::SetInterface(iface);
-  // Set the RTP recv/send buffer to a bigger size
-  if (network_interface_) {
-    network_interface_->SetOption(NetworkInterface::ST_RTP,
-                                  talk_base::Socket::OPT_RCVBUF,
-                                  kVideoRtpBufferSize);
-    network_interface_->SetOption(NetworkInterface::ST_RTP,
-                                  talk_base::Socket::OPT_SNDBUF,
-                                  kVideoRtpBufferSize);
-  }
-}
-
-// TODO: Add unittests to test this function.
-bool WebRtcVideoMediaChannel::SendFrame(uint32 ssrc, const VideoFrame* frame) {
-  if (ssrc != 0 || !sending() || !external_capture_) {
-    return false;
-  }
-
-  // Update local stream statistics.
-  local_stream_info_->UpdateFrame(frame->GetWidth(), frame->GetHeight());
-
-  // If the captured video format is smaller than what we asked for, reset send
-  // codec on video engine.
-  if (send_codec_.get() != NULL &&
-      frame->GetWidth() < send_codec_->width &&
-      frame->GetHeight() < send_codec_->height) {
-    LOG(LS_INFO) << "Captured video frame size changed to: "
-                 << frame->GetWidth() << "x" << frame->GetHeight();
-    webrtc::VideoCodec new_codec = *send_codec_;
-    new_codec.width = frame->GetWidth();
-    new_codec.height = frame->GetHeight();
-    if (!SetSendCodec(
-        new_codec, send_min_bitrate_, send_start_bitrate_, send_max_bitrate_)) {
-      LOG(LS_WARNING) << "Failed to switch to new frame size: "
-                      << frame->GetWidth() << "x" << frame->GetHeight();
-    }
-  }
-  
-  // Blacken the frame if video is muted.
-  const VideoFrame* frame_out = frame;
-  talk_base::scoped_ptr<VideoFrame> black_frame;
-  if (muted_) {
-    black_frame.reset(frame->Copy());
-    black_frame->SetToBlack();
-    frame_out = black_frame.get();
-  }
-
-  webrtc::ViEVideoFrameI420 frame_i420;
-  // TODO: Update the webrtc::ViEVideoFrameI420
-  // to use const unsigned char*
-  frame_i420.y_plane = const_cast<unsigned char*>(frame_out->GetYPlane());
-  frame_i420.u_plane = const_cast<unsigned char*>(frame_out->GetUPlane());
-  frame_i420.v_plane = const_cast<unsigned char*>(frame_out->GetVPlane());
-  frame_i420.y_pitch = frame_out->GetYPitch();
-  frame_i420.u_pitch = frame_out->GetUPitch();
-  frame_i420.v_pitch = frame_out->GetVPitch();
-  frame_i420.width = frame_out->GetWidth();
-  frame_i420.height = frame_out->GetHeight();
-
-  // Convert from nanoseconds to milliseconds.
-  WebRtc_Word64 clocks = frame_out->GetTimeStamp() /
-      talk_base::kNumNanosecsPerMillisec;
-
-  return (external_capture_->IncomingFrameI420(frame_i420, clocks) == 0);
-}
-
-bool WebRtcVideoMediaChannel::EnableRtcp() {
-  if (engine()->vie()->rtp()->SetRTCPStatus(
-          vie_channel_, webrtc::kRtcpCompound_RFC4585) != 0) {
-    LOG_RTCERR2(SetRTCPStatus, vie_channel_, webrtc::kRtcpCompound_RFC4585);
-    return false;
-  }
-  return true;
-}
-
-bool WebRtcVideoMediaChannel::EnablePli() {
-  if (engine_->vie()->rtp()->SetKeyFrameRequestMethod(
-          vie_channel_, webrtc::kViEKeyFrameRequestPliRtcp) != 0) {
-    LOG_RTCERR2(SetRTCPStatus,
-                vie_channel_, webrtc::kViEKeyFrameRequestPliRtcp);
-    return false;
-  }
-  return true;
-}
-
-bool WebRtcVideoMediaChannel::EnableTmmbr() {
-  if (engine_->vie()->rtp()->SetTMMBRStatus(vie_channel_, true) != 0) {
-    LOG_RTCERR1(SetTMMBRStatus, vie_channel_);
-    return false;
-  }
-  return true;
-}
-
-bool WebRtcVideoMediaChannel::EnableNack() {
-  if (engine_->vie()->rtp()->SetNACKStatus(vie_channel_, true) != 0) {
-    LOG_RTCERR1(SetNACKStatus, vie_channel_);
-    return false;
-  }
-  return true;
-}
-
-bool WebRtcVideoMediaChannel::SetNackFec(int red_payload_type,
-                                         int fec_payload_type) {
-  bool enable = (red_payload_type != -1 && fec_payload_type != -1);
-  if (engine_->vie()->rtp()->SetHybridNACKFECStatus(
-          vie_channel_, enable, red_payload_type, fec_payload_type) != 0) {
-    LOG_RTCERR4(SetHybridNACKFECStatus,
-                vie_channel_, enable, red_payload_type, fec_payload_type);
-    return false;
-  }
-  return true;
-}
-
-bool WebRtcVideoMediaChannel::SetSendCodec(const webrtc::VideoCodec& codec,
-                                           int min_bitrate,
-                                           int start_bitrate,
-                                           int max_bitrate) {
-  // Make a copy of the codec
-  webrtc::VideoCodec target_codec = codec;
-  target_codec.startBitrate = start_bitrate;
-  target_codec.minBitrate = min_bitrate;
-  target_codec.maxBitrate = max_bitrate;
-
-  if (engine()->vie()->codec()->SetSendCodec(vie_channel_, target_codec) != 0) {
-    LOG_RTCERR2(SetSendCodec, vie_channel_, send_codec_->plName);
-    return false;
-  }
-
-  // Reset the send_codec_ only if SetSendCodec is success.
-  send_codec_.reset(new webrtc::VideoCodec(target_codec));
-  send_min_bitrate_ = min_bitrate;
-  send_start_bitrate_ = start_bitrate;
-  send_max_bitrate_ = max_bitrate;
-
-  return true;
-}
-
-int WebRtcVideoMediaChannel::SendPacket(int channel, const void* data,
-                                        int len) {
-  if (!network_interface_) {
-    return -1;
-  }
-  talk_base::Buffer packet(data, len, kMaxRtpPacketLen);
-  return network_interface_->SendPacket(&packet) ? len : -1;
-}
-
-int WebRtcVideoMediaChannel::SendRTCPPacket(int channel,
-                                            const void* data,
-                                            int len) {
-  if (!network_interface_) {
-    return -1;
-  }
-  talk_base::Buffer packet(data, len, kMaxRtpPacketLen);
-  return network_interface_->SendRtcp(&packet) ? len : -1;
-}
-
-}  // namespace cricket
-
-#endif  // HAVE_WEBRTC_VIDEO