[Cast] Clean-up: Merge RtpReceiver+AudioReceiver+VideoReceiver-->FrameReceiver.

Creates a new media/cast/receiver directory, and consolidates all code
in media/cast/audio_receiver and media/cast/video_receiver, and some
modules from media/cast/rtp_receiver into one place.

Removed FrameReceiver proxy interface in cast_receiver.h, since there
are no special threading/lifecycle concerns.  Clients will directly call
methods in the CastReceiver interface to request frames on the MAIN
thread.

BUG=378568

Review URL: https://codereview.chromium.org/308043006

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@274464 0039d316-1c4b-4281-b951-d872f2087c98

31 files changed, 1051 insertions, 1780 deletions

diff --git a/media/cast/audio_receiver/audio_receiver.h b/media/cast/audio_receiver/audio_receiver.h
deleted file mode 100644
index 87c5147..0000000
--- a/media/cast/audio_receiver/audio_receiver.h
+++ /dev/null
@@ -1,210 +0,0 @@
-// Copyright 2013 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.
-
-#ifndef MEDIA_CAST_AUDIO_RECEIVER_AUDIO_RECEIVER_H_
-#define MEDIA_CAST_AUDIO_RECEIVER_AUDIO_RECEIVER_H_
-
-#include "base/basictypes.h"
-#include "base/callback.h"
-#include "base/macros.h"
-#include "base/memory/ref_counted.h"
-#include "base/memory/scoped_ptr.h"
-#include "base/memory/weak_ptr.h"
-#include "base/threading/non_thread_safe.h"
-#include "base/time/tick_clock.h"
-#include "base/time/time.h"
-#include "media/cast/base/clock_drift_smoother.h"
-#include "media/cast/cast_config.h"
-#include "media/cast/cast_environment.h"
-#include "media/cast/cast_receiver.h"
-#include "media/cast/framer/framer.h"
-#include "media/cast/rtcp/receiver_rtcp_event_subscriber.h"
-#include "media/cast/rtcp/rtcp.h"
-#include "media/cast/rtp_receiver/rtp_receiver.h"
-#include "media/cast/rtp_receiver/rtp_receiver_defines.h"
-#include "media/cast/transport/utility/transport_encryption_handler.h"
-
-namespace media {
-namespace cast {
-
-class AudioDecoder;
-
-// AudioReceiver receives packets out-of-order while clients make requests for
-// complete frames in-order.  (A frame consists of one or more packets.)
-//
-// AudioReceiver also includes logic for computing the playout time for each
-// frame, accounting for a constant targeted playout delay.  The purpose of the
-// playout delay is to provide a fixed window of time between the capture event
-// on the sender and the playout on the receiver.  This is important because
-// each step of the pipeline (i.e., encode frame, then transmit/retransmit from
-// the sender, then receive and re-order packets on the receiver, then decode
-// frame) can vary in duration and is typically very hard to predict.
-//
-// Two types of frames can be requested: 1) A frame of decoded audio data; or 2)
-// a frame of still-encoded audio data, to be passed into an external audio
-// decoder.  Each request for a frame includes a callback which AudioReceiver
-// guarantees will be called at some point in the future unless the
-// AudioReceiver is destroyed.  Clients should generally limit the number of
-// outstanding requests (perhaps to just one or two).
-//
-// This class is not thread safe.  Should only be called from the Main cast
-// thread.
-class AudioReceiver : public RtpReceiver,
-                      public RtpPayloadFeedback,
-                      public base::NonThreadSafe,
-                      public base::SupportsWeakPtr<AudioReceiver> {
- public:
-  AudioReceiver(scoped_refptr<CastEnvironment> cast_environment,
-                const FrameReceiverConfig& audio_config,
-                transport::PacedPacketSender* const packet_sender);
-
-  virtual ~AudioReceiver();
-
-  // Request a decoded audio frame.  The audio signal data returned in the
-  // callback will have the sampling rate and number of channels as requested in
-  // the configuration that was passed to the ctor.
-  //
-  // The given |callback| is guaranteed to be run at some point in the future,
-  // even if to respond with NULL at shutdown time.
-  void GetRawAudioFrame(const AudioFrameDecodedCallback& callback);
-
-  // Request an encoded audio frame.
-  //
-  // The given |callback| is guaranteed to be run at some point in the future,
-  // even if to respond with NULL at shutdown time.
-  void GetEncodedAudioFrame(const FrameEncodedCallback& callback);
-
-  // Deliver another packet, possibly a duplicate, and possibly out-of-order.
-  void IncomingPacket(scoped_ptr<Packet> packet);
-
- protected:
-  friend class AudioReceiverTest;  // Invokes OnReceivedPayloadData().
-
-  virtual void OnReceivedPayloadData(const uint8* payload_data,
-                                     size_t payload_size,
-                                     const RtpCastHeader& rtp_header) OVERRIDE;
-
-  // RtpPayloadFeedback implementation.
-  virtual void CastFeedback(const RtcpCastMessage& cast_message) OVERRIDE;
-
- private:
-  // Processes ready-to-consume packets from |framer_|, decrypting each packet's
-  // payload data, and then running the enqueued callbacks in order (one for
-  // each packet).  This method may post a delayed task to re-invoke itself in
-  // the future to wait for missing/incomplete frames.
-  void EmitAvailableEncodedFrames();
-
-  // Clears the |is_waiting_for_consecutive_frame_| flag and invokes
-  // EmitAvailableEncodedFrames().
-  void EmitAvailableEncodedFramesAfterWaiting();
-
-  // Feeds an EncodedFrame into |audio_decoder_|.  GetRawAudioFrame() uses this
-  // as a callback for GetEncodedAudioFrame().
-  void DecodeEncodedAudioFrame(
-      const AudioFrameDecodedCallback& callback,
-      scoped_ptr<transport::EncodedFrame> encoded_frame);
-
-  // Computes the playout time for a frame with the given |rtp_timestamp|.
-  // Because lip-sync info is refreshed regularly, calling this method with the
-  // same argument may return different results.
-  base::TimeTicks GetPlayoutTime(uint32 rtp_timestamp) const;
-
-  // Schedule the next RTCP report.
-  void ScheduleNextRtcpReport();
-
-  // Actually send the next RTCP report.
-  void SendNextRtcpReport();
-
-  // Schedule timing for the next cast message.
-  void ScheduleNextCastMessage();
-
-  // Actually send the next cast message.
-  void SendNextCastMessage();
-
-  // Receives an AudioBus from |audio_decoder_|, logs the event, and passes the
-  // data on by running the given |callback|.  This method is static to ensure
-  // it can be called after an AudioReceiver instance is destroyed.
-  // DecodeEncodedAudioFrame() uses this as a callback for
-  // AudioDecoder::DecodeFrame().
-  static void EmitRawAudioFrame(
-      const scoped_refptr<CastEnvironment>& cast_environment,
-      const AudioFrameDecodedCallback& callback,
-      uint32 frame_id,
-      uint32 rtp_timestamp,
-      const base::TimeTicks& playout_time,
-      scoped_ptr<AudioBus> audio_bus,
-      bool is_continuous);
-
-  const scoped_refptr<CastEnvironment> cast_environment_;
-
-  // Subscribes to raw events.
-  // Processes raw audio events to be sent over to the cast sender via RTCP.
-  ReceiverRtcpEventSubscriber event_subscriber_;
-
-  // Configured audio codec.
-  const transport::AudioCodec codec_;
-
-  // RTP timebase: The number of RTP units advanced per one second.  For audio,
-  // this is the sampling rate.
-  const int frequency_;
-
-  // The total amount of time between a frame's capture/recording on the sender
-  // and its playback on the receiver (i.e., shown to a user).  This is fixed as
-  // a value large enough to give the system sufficient time to encode,
-  // transmit/retransmit, receive, decode, and render; given its run-time
-  // environment (sender/receiver hardware performance, network conditions,
-  // etc.).
-  const base::TimeDelta target_playout_delay_;
-
-  // Hack: This is used in logic that determines whether to skip frames.
-  const base::TimeDelta expected_frame_duration_;
-
-  // Set to false initially, then set to true after scheduling the periodic
-  // sending of reports back to the sender.  Reports are first scheduled just
-  // after receiving a first packet (since the first packet identifies the
-  // sender for the remainder of the session).
-  bool reports_are_scheduled_;
-
-  // Assembles packets into frames, providing this receiver with complete,
-  // decodable EncodedFrames.
-  Framer framer_;
-
-  // Decodes frames into raw audio for playback.
-  scoped_ptr<AudioDecoder> audio_decoder_;
-
-  // Manages sending/receiving of RTCP packets, including sender/receiver
-  // reports.
-  Rtcp rtcp_;
-
-  // Decrypts encrypted frames.
-  transport::TransportEncryptionHandler decryptor_;
-
-  // Outstanding callbacks to run to deliver on client requests for frames.
-  std::list<FrameEncodedCallback> frame_request_queue_;
-
-  // True while there's an outstanding task to re-invoke
-  // EmitAvailableEncodedFrames().
-  bool is_waiting_for_consecutive_frame_;
-
-  // This mapping allows us to log AUDIO_ACK_SENT as a frame event. In addition
-  // it allows the event to be transmitted via RTCP.
-  RtpTimestamp frame_id_to_rtp_timestamp_[256];
-
-  // Lip-sync values used to compute the playout time of each frame from its RTP
-  // timestamp.  These are updated each time the first packet of a frame is
-  // received.
-  RtpTimestamp lip_sync_rtp_timestamp_;
-  base::TimeTicks lip_sync_reference_time_;
-  ClockDriftSmoother lip_sync_drift_;
-
-  // NOTE: Weak pointers must be invalidated before all other member variables.
-  base::WeakPtrFactory<AudioReceiver> weak_factory_;
-
-  DISALLOW_COPY_AND_ASSIGN(AudioReceiver);
-};
-
-}  // namespace cast
-}  // namespace media
-
-#endif  // MEDIA_CAST_AUDIO_RECEIVER_AUDIO_RECEIVER_H_
diff --git a/media/cast/audio_receiver/audio_receiver_unittest.cc b/media/cast/audio_receiver/audio_receiver_unittest.cc
deleted file mode 100644
index e53c1b9..0000000
--- a/media/cast/audio_receiver/audio_receiver_unittest.cc
+++ /dev/null
@@ -1,261 +0,0 @@
-// Copyright 2013 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.
-
-#include <deque>
-#include <utility>
-
-#include "base/bind.h"
-#include "base/memory/ref_counted.h"
-#include "base/memory/scoped_ptr.h"
-#include "base/test/simple_test_tick_clock.h"
-#include "media/cast/audio_receiver/audio_receiver.h"
-#include "media/cast/cast_defines.h"
-#include "media/cast/cast_environment.h"
-#include "media/cast/logging/simple_event_subscriber.h"
-#include "media/cast/rtcp/test_rtcp_packet_builder.h"
-#include "media/cast/test/fake_single_thread_task_runner.h"
-#include "media/cast/test/utility/default_config.h"
-#include "media/cast/transport/pacing/mock_paced_packet_sender.h"
-#include "testing/gmock/include/gmock/gmock.h"
-
-using ::testing::_;
-
-namespace media {
-namespace cast {
-
-namespace {
-
-const uint32 kFirstFrameId = 1234;
-const int kPlayoutDelayMillis = 300;
-
-class FakeAudioClient {
- public:
-  FakeAudioClient() : num_called_(0) {}
-  virtual ~FakeAudioClient() {}
-
-  void AddExpectedResult(uint32 expected_frame_id,
-                         const base::TimeTicks& expected_playout_time) {
-    expected_results_.push_back(
-        std::make_pair(expected_frame_id, expected_playout_time));
-  }
-
-  void DeliverEncodedAudioFrame(
-      scoped_ptr<transport::EncodedFrame> audio_frame) {
-    SCOPED_TRACE(::testing::Message() << "num_called_ is " << num_called_);
-    ASSERT_FALSE(!audio_frame)
-        << "If at shutdown: There were unsatisfied requests enqueued.";
-    ASSERT_FALSE(expected_results_.empty());
-    EXPECT_EQ(expected_results_.front().first, audio_frame->frame_id);
-    EXPECT_EQ(expected_results_.front().second, audio_frame->reference_time);
-    expected_results_.pop_front();
-    num_called_++;
-  }
-
-  int number_times_called() const { return num_called_; }
-
- private:
-  std::deque<std::pair<uint32, base::TimeTicks> > expected_results_;
-  int num_called_;
-
-  DISALLOW_COPY_AND_ASSIGN(FakeAudioClient);
-};
-
-}  // namespace
-
-class AudioReceiverTest : public ::testing::Test {
- protected:
-  AudioReceiverTest() {
-    // Configure the audio receiver to use PCM16.
-    audio_config_ = GetDefaultAudioReceiverConfig();
-    audio_config_.rtp_max_delay_ms = kPlayoutDelayMillis;
-    audio_config_.frequency = 16000;
-    audio_config_.channels = 1;
-    audio_config_.codec.audio = transport::kPcm16;
-    testing_clock_ = new base::SimpleTestTickClock();
-    testing_clock_->Advance(base::TimeTicks::Now() - base::TimeTicks());
-    start_time_ = testing_clock_->NowTicks();
-    task_runner_ = new test::FakeSingleThreadTaskRunner(testing_clock_);
-
-    cast_environment_ = new CastEnvironment(
-        scoped_ptr<base::TickClock>(testing_clock_).Pass(),
-        task_runner_,
-        task_runner_,
-        task_runner_);
-
-    receiver_.reset(new AudioReceiver(cast_environment_, audio_config_,
-                                      &mock_transport_));
-  }
-
-  virtual ~AudioReceiverTest() {}
-
-  virtual void SetUp() {
-    payload_.assign(kMaxIpPacketSize, 0);
-    rtp_header_.is_key_frame = true;
-    rtp_header_.frame_id = kFirstFrameId;
-    rtp_header_.packet_id = 0;
-    rtp_header_.max_packet_id = 0;
-    rtp_header_.reference_frame_id = rtp_header_.frame_id;
-    rtp_header_.rtp_timestamp = 0;
-  }
-
-  void FeedOneFrameIntoReceiver() {
-    receiver_->OnReceivedPayloadData(
-        payload_.data(), payload_.size(), rtp_header_);
-  }
-
-  void FeedLipSyncInfoIntoReceiver() {
-    const base::TimeTicks now = testing_clock_->NowTicks();
-    const int64 rtp_timestamp = (now - start_time_) *
-        audio_config_.frequency / base::TimeDelta::FromSeconds(1);
-    CHECK_LE(0, rtp_timestamp);
-    uint32 ntp_seconds;
-    uint32 ntp_fraction;
-    ConvertTimeTicksToNtp(now, &ntp_seconds, &ntp_fraction);
-    TestRtcpPacketBuilder rtcp_packet;
-    rtcp_packet.AddSrWithNtp(audio_config_.incoming_ssrc,
-                             ntp_seconds, ntp_fraction,
-                             static_cast<uint32>(rtp_timestamp));
-    receiver_->IncomingPacket(rtcp_packet.GetPacket().Pass());
-  }
-
-  FrameReceiverConfig audio_config_;
-  std::vector<uint8> payload_;
-  RtpCastHeader rtp_header_;
-  base::SimpleTestTickClock* testing_clock_;  // Owned by CastEnvironment.
-  base::TimeTicks start_time_;
-  transport::MockPacedPacketSender mock_transport_;
-  scoped_refptr<test::FakeSingleThreadTaskRunner> task_runner_;
-  scoped_refptr<CastEnvironment> cast_environment_;
-  FakeAudioClient fake_audio_client_;
-
-  // Important for the AudioReceiver to be declared last, since its dependencies
-  // must remain alive until after its destruction.
-  scoped_ptr<AudioReceiver> receiver_;
-};
-
-TEST_F(AudioReceiverTest, ReceivesOneFrame) {
-  SimpleEventSubscriber event_subscriber;
-  cast_environment_->Logging()->AddRawEventSubscriber(&event_subscriber);
-
-  EXPECT_CALL(mock_transport_, SendRtcpPacket(_, _))
-      .WillRepeatedly(testing::Return(true));
-
-  FeedLipSyncInfoIntoReceiver();
-  task_runner_->RunTasks();
-
-  // Enqueue a request for an audio frame.
-  receiver_->GetEncodedAudioFrame(
-      base::Bind(&FakeAudioClient::DeliverEncodedAudioFrame,
-                 base::Unretained(&fake_audio_client_)));
-
-  // The request should not be satisfied since no packets have been received.
-  task_runner_->RunTasks();
-  EXPECT_EQ(0, fake_audio_client_.number_times_called());
-
-  // Deliver one audio frame to the receiver and expect to get one frame back.
-  const base::TimeDelta target_playout_delay =
-      base::TimeDelta::FromMilliseconds(kPlayoutDelayMillis);
-  fake_audio_client_.AddExpectedResult(
-      kFirstFrameId, testing_clock_->NowTicks() + target_playout_delay);
-  FeedOneFrameIntoReceiver();
-  task_runner_->RunTasks();
-  EXPECT_EQ(1, fake_audio_client_.number_times_called());
-
-  std::vector<FrameEvent> frame_events;
-  event_subscriber.GetFrameEventsAndReset(&frame_events);
-
-  ASSERT_TRUE(!frame_events.empty());
-  EXPECT_EQ(FRAME_ACK_SENT, frame_events.begin()->type);
-  EXPECT_EQ(AUDIO_EVENT, frame_events.begin()->media_type);
-  EXPECT_EQ(rtp_header_.frame_id, frame_events.begin()->frame_id);
-  EXPECT_EQ(rtp_header_.rtp_timestamp, frame_events.begin()->rtp_timestamp);
-
-  cast_environment_->Logging()->RemoveRawEventSubscriber(&event_subscriber);
-}
-
-TEST_F(AudioReceiverTest, ReceivesFramesSkippingWhenAppropriate) {
-  EXPECT_CALL(mock_transport_, SendRtcpPacket(_, _))
-      .WillRepeatedly(testing::Return(true));
-
-  const uint32 rtp_advance_per_frame =
-      audio_config_.frequency / audio_config_.max_frame_rate;
-  const base::TimeDelta time_advance_per_frame =
-      base::TimeDelta::FromSeconds(1) / audio_config_.max_frame_rate;
-
-  FeedLipSyncInfoIntoReceiver();
-  task_runner_->RunTasks();
-  const base::TimeTicks first_frame_capture_time = testing_clock_->NowTicks();
-
-  // Enqueue a request for an audio frame.
-  const FrameEncodedCallback frame_encoded_callback =
-      base::Bind(&FakeAudioClient::DeliverEncodedAudioFrame,
-                 base::Unretained(&fake_audio_client_));
-  receiver_->GetEncodedAudioFrame(frame_encoded_callback);
-  task_runner_->RunTasks();
-  EXPECT_EQ(0, fake_audio_client_.number_times_called());
-
-  // Receive one audio frame and expect to see the first request satisfied.
-  const base::TimeDelta target_playout_delay =
-      base::TimeDelta::FromMilliseconds(kPlayoutDelayMillis);
-  fake_audio_client_.AddExpectedResult(
-      kFirstFrameId, first_frame_capture_time + target_playout_delay);
-  rtp_header_.rtp_timestamp = 0;
-  FeedOneFrameIntoReceiver();
-  task_runner_->RunTasks();
-  EXPECT_EQ(1, fake_audio_client_.number_times_called());
-
-  // Enqueue a second request for an audio frame, but it should not be
-  // fulfilled yet.
-  receiver_->GetEncodedAudioFrame(frame_encoded_callback);
-  task_runner_->RunTasks();
-  EXPECT_EQ(1, fake_audio_client_.number_times_called());
-
-  // Receive one audio frame out-of-order: Make sure that we are not continuous
-  // and that the RTP timestamp represents a time in the future.
-  rtp_header_.is_key_frame = false;
-  rtp_header_.frame_id = kFirstFrameId + 2;
-  rtp_header_.reference_frame_id = 0;
-  rtp_header_.rtp_timestamp += 2 * rtp_advance_per_frame;
-  fake_audio_client_.AddExpectedResult(
-      kFirstFrameId + 2,
-      first_frame_capture_time + 2 * time_advance_per_frame +
-          target_playout_delay);
-  FeedOneFrameIntoReceiver();
-
-  // Frame 2 should not come out at this point in time.
-  task_runner_->RunTasks();
-  EXPECT_EQ(1, fake_audio_client_.number_times_called());
-
-  // Enqueue a third request for an audio frame.
-  receiver_->GetEncodedAudioFrame(frame_encoded_callback);
-  task_runner_->RunTasks();
-  EXPECT_EQ(1, fake_audio_client_.number_times_called());
-
-  // Now, advance time forward such that the receiver is convinced it should
-  // skip Frame 2.  Frame 3 is emitted (to satisfy the second request) because a
-  // decision was made to skip over the no-show Frame 2.
-  testing_clock_->Advance(2 * time_advance_per_frame + target_playout_delay);
-  task_runner_->RunTasks();
-  EXPECT_EQ(2, fake_audio_client_.number_times_called());
-
-  // Receive Frame 4 and expect it to fulfill the third request immediately.
-  rtp_header_.frame_id = kFirstFrameId + 3;
-  rtp_header_.reference_frame_id = rtp_header_.frame_id - 1;
-  rtp_header_.rtp_timestamp += rtp_advance_per_frame;
-  fake_audio_client_.AddExpectedResult(
-      kFirstFrameId + 3, first_frame_capture_time + 3 * time_advance_per_frame +
-          target_playout_delay);
-  FeedOneFrameIntoReceiver();
-  task_runner_->RunTasks();
-  EXPECT_EQ(3, fake_audio_client_.number_times_called());
-
-  // Move forward to the playout time of an unreceived Frame 5.  Expect no
-  // additional frames were emitted.
-  testing_clock_->Advance(3 * time_advance_per_frame);
-  task_runner_->RunTasks();
-  EXPECT_EQ(3, fake_audio_client_.number_times_called());
-}
-
-}  // namespace cast
-}  // namespace media
diff --git a/media/cast/audio_sender/audio_sender.cc b/media/cast/audio_sender/audio_sender.cc
index 33623e9..a6e8cfe 100644
--- a/media/cast/audio_sender/audio_sender.cc
+++ b/media/cast/audio_sender/audio_sender.cc
@@ -33,7 +33,7 @@ AudioSender::AudioSender(scoped_refptr<CastEnvironment> cast_environment,
             audio_config.rtp_config.ssrc,
             audio_config.incoming_feedback_ssrc,
             audio_config.rtcp_c_name,
-            true),
+            AUDIO_EVENT),
       num_aggressive_rtcp_reports_sent_(0),
       cast_initialization_cb_(STATUS_AUDIO_UNINITIALIZED),
       weak_factory_(this) {
diff --git a/media/cast/cast.gyp b/media/cast/cast.gyp
index 4f0c2dd..45ff016 100644
--- a/media/cast/cast.gyp
+++ b/media/cast/cast.gyp
@@ -99,13 +99,7 @@
         '<(DEPTH)/ui/gfx/gfx.gyp:gfx_geometry',
       ],
       'sources': [
-        'audio_receiver/audio_decoder.h',
-        'audio_receiver/audio_decoder.cc',
-        'audio_receiver/audio_receiver.h',
-        'audio_receiver/audio_receiver.cc',
         'cast_receiver.h',
-        'cast_receiver_impl.cc',
-        'cast_receiver_impl.h',
         'framer/cast_message_builder.cc',
         'framer/cast_message_builder.h',
         'framer/frame_buffer.cc',
@@ -114,18 +108,20 @@
         'framer/frame_id_map.h',
         'framer/framer.cc',
         'framer/framer.h',
+        'receiver/audio_decoder.cc',
+        'receiver/audio_decoder.h',
+        'receiver/cast_receiver_impl.cc',
+        'receiver/cast_receiver_impl.h',
+        'receiver/frame_receiver.cc',
+        'receiver/frame_receiver.h',
+        'receiver/video_decoder.cc',
+        'receiver/video_decoder.h',
         'rtp_receiver/receiver_stats.cc',
         'rtp_receiver/receiver_stats.h',
-        'rtp_receiver/rtp_receiver.cc',
-        'rtp_receiver/rtp_receiver.h',
         'rtp_receiver/rtp_receiver_defines.cc',
         'rtp_receiver/rtp_receiver_defines.h',
         'rtp_receiver/rtp_parser/rtp_parser.cc',
         'rtp_receiver/rtp_parser/rtp_parser.h',
-        'video_receiver/video_decoder.h',
-        'video_receiver/video_decoder.cc',
-        'video_receiver/video_receiver.h',
-        'video_receiver/video_receiver.cc',
       ], # source
     },
     {
diff --git a/media/cast/cast_receiver.h b/media/cast/cast_receiver.h
index 2d83dd2..a9d3ede 100644
--- a/media/cast/cast_receiver.h
+++ b/media/cast/cast_receiver.h
@@ -45,27 +45,8 @@ typedef base::Callback<void(const scoped_refptr<media::VideoFrame>& video_frame,
 // dropped (i.e., frame_id should be incrementing by one each time).  Note: A
 // NULL pointer can be returned on error.
 typedef base::Callback<void(scoped_ptr<transport::EncodedFrame>)>
-    FrameEncodedCallback;
+    ReceiveEncodedFrameCallback;
 
-// This Class is thread safe.
-class FrameReceiver : public base::RefCountedThreadSafe<FrameReceiver> {
- public:
-  virtual void GetRawAudioFrame(const AudioFrameDecodedCallback& callback) = 0;
-
-  virtual void GetCodedAudioFrame(const FrameEncodedCallback& callback) = 0;
-
-  virtual void GetRawVideoFrame(const VideoFrameDecodedCallback& callback) = 0;
-
-  virtual void GetEncodedVideoFrame(const FrameEncodedCallback& callback) = 0;
-
- protected:
-  virtual ~FrameReceiver() {}
-
- private:
-  friend class base::RefCountedThreadSafe<FrameReceiver>;
-};
-
-// This Class is thread safe.
 class CastReceiver {
  public:
   static scoped_ptr<CastReceiver> Create(
@@ -75,13 +56,28 @@ class CastReceiver {
       transport::PacketSender* const packet_sender);
 
   // All received RTP and RTCP packets for the call should be sent to this
-  // PacketReceiver. Can be called from any function.
+  // PacketReceiver.  Can be called from any thread.
   // TODO(hubbe): Replace with:
   //   virtual void ReceivePacket(scoped_ptr<Packet> packet) = 0;
   virtual transport::PacketReceiverCallback packet_receiver() = 0;
 
-  // Polling interface to get audio and video frames from the CastReceiver.
-  virtual scoped_refptr<FrameReceiver> frame_receiver() = 0;
+  // Polling interface to get audio and video frames from the CastReceiver.  The
+  // the RequestDecodedXXXXXFrame() methods utilize internal software-based
+  // decoding, while the RequestEncodedXXXXXFrame() methods provides
+  // still-encoded frames for use with external/hardware decoders.
+  //
+  // In all cases, the given |callback| is guaranteed to be run at some point in
+  // the future, except for those requests still enqueued at destruction time.
+  //
+  // These methods should all be called on the CastEnvironment's MAIN thread.
+  virtual void RequestDecodedAudioFrame(
+      const AudioFrameDecodedCallback& callback) = 0;
+  virtual void RequestEncodedAudioFrame(
+      const ReceiveEncodedFrameCallback& callback) = 0;
+  virtual void RequestDecodedVideoFrame(
+      const VideoFrameDecodedCallback& callback) = 0;
+  virtual void RequestEncodedVideoFrame(
+      const ReceiveEncodedFrameCallback& callback) = 0;
 
   virtual ~CastReceiver() {}
 };
diff --git a/media/cast/cast_receiver_impl.cc b/media/cast/cast_receiver_impl.cc
deleted file mode 100644
index 661cbbe..0000000
--- a/media/cast/cast_receiver_impl.cc
+++ /dev/null
@@ -1,150 +0,0 @@
-// Copyright 2013 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.
-
-#include "media/cast/cast_receiver_impl.h"
-
-#include "base/bind.h"
-#include "base/callback.h"
-#include "base/logging.h"
-#include "base/message_loop/message_loop.h"
-
-namespace media {
-namespace cast {
-
-// The video and audio receivers should only be called from the main thread.
-// LocalFrameReciever posts tasks to the main thread, making the cast interface
-// thread safe.
-class LocalFrameReceiver : public FrameReceiver {
- public:
-  LocalFrameReceiver(scoped_refptr<CastEnvironment> cast_environment,
-                     AudioReceiver* audio_receiver,
-                     VideoReceiver* video_receiver)
-      : cast_environment_(cast_environment),
-        audio_receiver_(audio_receiver),
-        video_receiver_(video_receiver) {}
-
-  virtual void GetRawVideoFrame(const VideoFrameDecodedCallback& callback)
-      OVERRIDE {
-    cast_environment_->PostTask(CastEnvironment::MAIN,
-                                FROM_HERE,
-                                base::Bind(&VideoReceiver::GetRawVideoFrame,
-                                           video_receiver_->AsWeakPtr(),
-                                           callback));
-  }
-
-  virtual void GetEncodedVideoFrame(const FrameEncodedCallback& callback)
-      OVERRIDE {
-    cast_environment_->PostTask(CastEnvironment::MAIN,
-                                FROM_HERE,
-                                base::Bind(&VideoReceiver::GetEncodedVideoFrame,
-                                           video_receiver_->AsWeakPtr(),
-                                           callback));
-  }
-
-  virtual void GetRawAudioFrame(const AudioFrameDecodedCallback& callback)
-      OVERRIDE {
-    cast_environment_->PostTask(CastEnvironment::MAIN,
-                                FROM_HERE,
-                                base::Bind(&AudioReceiver::GetRawAudioFrame,
-                                           audio_receiver_->AsWeakPtr(),
-                                           callback));
-  }
-
-  virtual void GetCodedAudioFrame(const FrameEncodedCallback& callback)
-      OVERRIDE {
-    cast_environment_->PostTask(CastEnvironment::MAIN,
-                                FROM_HERE,
-                                base::Bind(&AudioReceiver::GetEncodedAudioFrame,
-                                           audio_receiver_->AsWeakPtr(),
-                                           callback));
-  }
-
- protected:
-  virtual ~LocalFrameReceiver() {}
-
- private:
-  friend class base::RefCountedThreadSafe<LocalFrameReceiver>;
-
-  scoped_refptr<CastEnvironment> cast_environment_;
-  AudioReceiver* audio_receiver_;
-  VideoReceiver* video_receiver_;
-};
-
-scoped_ptr<CastReceiver> CastReceiver::Create(
-    scoped_refptr<CastEnvironment> cast_environment,
-    const FrameReceiverConfig& audio_config,
-    const FrameReceiverConfig& video_config,
-    transport::PacketSender* const packet_sender) {
-  return scoped_ptr<CastReceiver>(new CastReceiverImpl(
-      cast_environment, audio_config, video_config, packet_sender));
-}
-
-CastReceiverImpl::CastReceiverImpl(
-    scoped_refptr<CastEnvironment> cast_environment,
-    const FrameReceiverConfig& audio_config,
-    const FrameReceiverConfig& video_config,
-    transport::PacketSender* const packet_sender)
-    : pacer_(cast_environment->Clock(),
-             cast_environment->Logging(),
-             packet_sender,
-             cast_environment->GetTaskRunner(CastEnvironment::MAIN)),
-      audio_receiver_(cast_environment, audio_config, &pacer_),
-      video_receiver_(cast_environment,
-                      video_config,
-                      &pacer_),
-      frame_receiver_(new LocalFrameReceiver(cast_environment,
-                                             &audio_receiver_,
-                                             &video_receiver_)),
-      cast_environment_(cast_environment),
-      ssrc_of_audio_sender_(audio_config.incoming_ssrc),
-      ssrc_of_video_sender_(video_config.incoming_ssrc) {}
-
-CastReceiverImpl::~CastReceiverImpl() {}
-
-// The video and audio receivers should only be called from the main thread.
-void CastReceiverImpl::ReceivedPacket(scoped_ptr<Packet> packet) {
-  const uint8_t* data = &packet->front();
-  size_t length = packet->size();
-  if (length < kMinLengthOfRtcp) {
-    VLOG(1) << "Received a packet which is too short " << length;
-    return;
-  }
-  uint32 ssrc_of_sender;
-  if (!Rtcp::IsRtcpPacket(data, length)) {
-    if (length < kMinLengthOfRtp) {
-      VLOG(1) << "Received a RTP packet which is too short " << length;
-      return;
-    }
-    ssrc_of_sender = RtpReceiver::GetSsrcOfSender(data, length);
-  } else {
-    ssrc_of_sender = Rtcp::GetSsrcOfSender(data, length);
-  }
-  if (ssrc_of_sender == ssrc_of_audio_sender_) {
-    cast_environment_->PostTask(CastEnvironment::MAIN,
-                                FROM_HERE,
-                                base::Bind(&AudioReceiver::IncomingPacket,
-                                           audio_receiver_.AsWeakPtr(),
-                                           base::Passed(&packet)));
-  } else if (ssrc_of_sender == ssrc_of_video_sender_) {
-    cast_environment_->PostTask(CastEnvironment::MAIN,
-                                FROM_HERE,
-                                base::Bind(&VideoReceiver::IncomingPacket,
-                                           video_receiver_.AsWeakPtr(),
-                                           base::Passed(&packet)));
-  } else {
-    VLOG(1) << "Received a packet with a non matching sender SSRC "
-            << ssrc_of_sender;
-  }
-}
-
-transport::PacketReceiverCallback CastReceiverImpl::packet_receiver() {
-  return base::Bind(&CastReceiverImpl::ReceivedPacket, base::Unretained(this));
-}
-
-scoped_refptr<FrameReceiver> CastReceiverImpl::frame_receiver() {
-  return frame_receiver_;
-}
-
-}  // namespace cast
-}  // namespace media
diff --git a/media/cast/cast_receiver_impl.h b/media/cast/cast_receiver_impl.h
deleted file mode 100644
index ae7d50e..0000000
--- a/media/cast/cast_receiver_impl.h
+++ /dev/null
@@ -1,55 +0,0 @@
-// Copyright 2013 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.
-
-#ifndef MEDIA_CAST_CAST_RECEIVER_IMPL_H_
-#define MEDIA_CAST_CAST_RECEIVER_IMPL_H_
-
-#include "base/memory/ref_counted.h"
-#include "base/memory/scoped_ptr.h"
-#include "media/cast/audio_receiver/audio_receiver.h"
-#include "media/cast/cast_config.h"
-#include "media/cast/cast_environment.h"
-#include "media/cast/cast_receiver.h"
-#include "media/cast/transport/pacing/paced_sender.h"
-#include "media/cast/video_receiver/video_receiver.h"
-
-namespace media {
-namespace cast {
-// This calls is a pure owner class that group all required receive objects
-// together such as pacer, packet receiver, frame receiver, audio and video
-// receivers.
-class CastReceiverImpl : public CastReceiver {
- public:
-  CastReceiverImpl(scoped_refptr<CastEnvironment> cast_environment,
-                   const FrameReceiverConfig& audio_config,
-                   const FrameReceiverConfig& video_config,
-                   transport::PacketSender* const packet_sender);
-
-  virtual ~CastReceiverImpl();
-
-  // All received RTP and RTCP packets for the call should be sent to this
-  // PacketReceiver;
-  virtual transport::PacketReceiverCallback packet_receiver() OVERRIDE;
-
-  // Interface to get audio and video frames from the CastReceiver.
-  virtual scoped_refptr<FrameReceiver> frame_receiver() OVERRIDE;
-
- private:
-  void ReceivedPacket(scoped_ptr<Packet> packet);
-
-  transport::PacedSender pacer_;
-  AudioReceiver audio_receiver_;
-  VideoReceiver video_receiver_;
-  scoped_refptr<FrameReceiver> frame_receiver_;
-  scoped_refptr<CastEnvironment> cast_environment_;
-  const uint32 ssrc_of_audio_sender_;
-  const uint32 ssrc_of_video_sender_;
-
-  DISALLOW_COPY_AND_ASSIGN(CastReceiverImpl);
-};
-
-}  // namespace cast
-}  // namespace media
-
-#endif  // MEDIA_CAST_CAST_RECEIVER_IMPL_
diff --git a/media/cast/cast_testing.gypi b/media/cast/cast_testing.gypi
index 666b582..ff6bb57 100644
--- a/media/cast/cast_testing.gypi
+++ b/media/cast/cast_testing.gypi
@@ -67,8 +67,6 @@
       ],
       'sources': [
         '<(DEPTH)/media/base/run_all_unittests.cc',
-        'audio_receiver/audio_decoder_unittest.cc',
-        'audio_receiver/audio_receiver_unittest.cc',
         'audio_sender/audio_encoder_unittest.cc',
         'audio_sender/audio_sender_unittest.cc',
         'congestion_control/congestion_control_unittest.cc',
@@ -82,6 +80,9 @@
         'logging/receiver_time_offset_estimator_impl_unittest.cc',
         'logging/simple_event_subscriber_unittest.cc',
         'logging/stats_event_subscriber_unittest.cc',
+        'receiver/audio_decoder_unittest.cc',
+        'receiver/frame_receiver_unittest.cc',
+        'receiver/video_decoder_unittest.cc',
         'rtcp/mock_rtcp_receiver_feedback.cc',
         'rtcp/mock_rtcp_receiver_feedback.h',
         'rtcp/mock_rtcp_sender_feedback.cc',
@@ -117,8 +118,6 @@
         'transport/rtp_sender/rtp_packetizer/test/rtp_header_parser.cc',
         'transport/rtp_sender/rtp_packetizer/test/rtp_header_parser.h',
         'transport/transport/udp_transport_unittest.cc',
-        'video_receiver/video_decoder_unittest.cc',
-        'video_receiver/video_receiver_unittest.cc',
         'video_sender/external_video_encoder_unittest.cc',
         'video_sender/video_encoder_impl_unittest.cc',
         'video_sender/video_sender_unittest.cc',
diff --git a/media/cast/audio_receiver/audio_decoder.cc b/media/cast/receiver/audio_decoder.cc
index e4e9a14..a4d1896 100644
--- a/media/cast/audio_receiver/audio_decoder.cc
+++ b/media/cast/receiver/audio_decoder.cc
@@ -1,8 +1,8 @@
-// Copyright 2013 The Chromium Authors. All rights reserved.
+// Copyright 2014 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.
 
-#include "media/cast/audio_receiver/audio_decoder.h"
+#include "media/cast/receiver/audio_decoder.h"
 
 #include "base/bind.h"
 #include "base/bind_helpers.h"
@@ -200,18 +200,16 @@ class AudioDecoder::Pcm16Impl : public AudioDecoder::ImplBase {
 
 AudioDecoder::AudioDecoder(
     const scoped_refptr<CastEnvironment>& cast_environment,
-    const FrameReceiverConfig& audio_config)
+    int channels,
+    int sampling_rate,
+    transport::AudioCodec codec)
     : cast_environment_(cast_environment) {
-  switch (audio_config.codec.audio) {
+  switch (codec) {
     case transport::kOpus:
-      impl_ = new OpusImpl(cast_environment,
-                           audio_config.channels,
-                           audio_config.frequency);
+      impl_ = new OpusImpl(cast_environment, channels, sampling_rate);
       break;
     case transport::kPcm16:
-      impl_ = new Pcm16Impl(cast_environment,
-                            audio_config.channels,
-                            audio_config.frequency);
+      impl_ = new Pcm16Impl(cast_environment, channels, sampling_rate);
       break;
     default:
       NOTREACHED() << "Unknown or unspecified codec.";
diff --git a/media/cast/audio_receiver/audio_decoder.h b/media/cast/receiver/audio_decoder.h
index a8e264d..c66735e 100644
--- a/media/cast/audio_receiver/audio_decoder.h
+++ b/media/cast/receiver/audio_decoder.h
@@ -1,9 +1,9 @@
-// Copyright 2013 The Chromium Authors. All rights reserved.
+// Copyright 2014 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.
 
-#ifndef MEDIA_CAST_AUDIO_RECEIVER_AUDIO_DECODER_H_
-#define MEDIA_CAST_AUDIO_RECEIVER_AUDIO_DECODER_H_
+#ifndef MEDIA_CAST_RECEIVER_AUDIO_DECODER_H_
+#define MEDIA_CAST_RECEIVER_AUDIO_DECODER_H_
 
 #include "base/callback.h"
 #include "base/memory/ref_counted.h"
@@ -27,7 +27,9 @@ class AudioDecoder {
                               bool is_continuous)> DecodeFrameCallback;
 
   AudioDecoder(const scoped_refptr<CastEnvironment>& cast_environment,
-               const FrameReceiverConfig& audio_config);
+               int channels,
+               int sampling_rate,
+               transport::AudioCodec codec);
   virtual ~AudioDecoder();
 
   // Returns STATUS_AUDIO_INITIALIZED if the decoder was successfully
@@ -59,4 +61,4 @@ class AudioDecoder {
 }  // namespace cast
 }  // namespace media
 
-#endif  // MEDIA_CAST_AUDIO_RECEIVER_AUDIO_DECODER_H_
+#endif  // MEDIA_CAST_RECEIVER_AUDIO_DECODER_H_
diff --git a/media/cast/audio_receiver/audio_decoder_unittest.cc b/media/cast/receiver/audio_decoder_unittest.cc
index 11973ab..6985a69 100644
--- a/media/cast/audio_receiver/audio_decoder_unittest.cc
+++ b/media/cast/receiver/audio_decoder_unittest.cc
@@ -1,4 +1,4 @@
-// Copyright 2013 The Chromium Authors. All rights reserved.
+// Copyright 2014 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.
 
@@ -8,10 +8,9 @@
 #include "base/synchronization/lock.h"
 #include "base/sys_byteorder.h"
 #include "base/time/time.h"
-#include "media/cast/audio_receiver/audio_decoder.h"
 #include "media/cast/cast_config.h"
+#include "media/cast/receiver/audio_decoder.h"
 #include "media/cast/test/utility/audio_utility.h"
-#include "media/cast/test/utility/default_config.h"
 #include "media/cast/test/utility/standalone_cast_environment.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "third_party/opus/src/include/opus.h"
@@ -38,11 +37,10 @@ class AudioDecoderTest : public ::testing::TestWithParam<TestScenario> {
 
  protected:
   virtual void SetUp() OVERRIDE {
-    FrameReceiverConfig decoder_config = GetDefaultAudioReceiverConfig();
-    decoder_config.frequency = GetParam().sampling_rate;
-    decoder_config.channels = GetParam().num_channels;
-    decoder_config.codec.audio = GetParam().codec;
-    audio_decoder_.reset(new AudioDecoder(cast_environment_, decoder_config));
+    audio_decoder_.reset(new AudioDecoder(cast_environment_,
+                                          GetParam().num_channels,
+                                          GetParam().sampling_rate,
+                                          GetParam().codec));
     CHECK_EQ(STATUS_AUDIO_INITIALIZED, audio_decoder_->InitializationResult());
 
     audio_bus_factory_.reset(
diff --git a/media/cast/receiver/cast_receiver_impl.cc b/media/cast/receiver/cast_receiver_impl.cc
new file mode 100644
index 0000000..7cff354
--- /dev/null
+++ b/media/cast/receiver/cast_receiver_impl.cc
@@ -0,0 +1,232 @@
+// Copyright 2014 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.
+
+#include "media/cast/receiver/cast_receiver_impl.h"
+
+#include "base/bind.h"
+#include "base/bind_helpers.h"
+#include "base/callback.h"
+#include "base/debug/trace_event.h"
+#include "base/logging.h"
+#include "base/message_loop/message_loop.h"
+#include "media/cast/receiver/audio_decoder.h"
+#include "media/cast/receiver/video_decoder.h"
+
+namespace media {
+namespace cast {
+
+scoped_ptr<CastReceiver> CastReceiver::Create(
+    scoped_refptr<CastEnvironment> cast_environment,
+    const FrameReceiverConfig& audio_config,
+    const FrameReceiverConfig& video_config,
+    transport::PacketSender* const packet_sender) {
+  return scoped_ptr<CastReceiver>(new CastReceiverImpl(
+      cast_environment, audio_config, video_config, packet_sender));
+}
+
+CastReceiverImpl::CastReceiverImpl(
+    scoped_refptr<CastEnvironment> cast_environment,
+    const FrameReceiverConfig& audio_config,
+    const FrameReceiverConfig& video_config,
+    transport::PacketSender* const packet_sender)
+    : cast_environment_(cast_environment),
+      pacer_(cast_environment->Clock(),
+             cast_environment->Logging(),
+             packet_sender,
+             cast_environment->GetTaskRunner(CastEnvironment::MAIN)),
+      audio_receiver_(cast_environment, audio_config, AUDIO_EVENT, &pacer_),
+      video_receiver_(cast_environment, video_config, VIDEO_EVENT, &pacer_),
+      ssrc_of_audio_sender_(audio_config.incoming_ssrc),
+      ssrc_of_video_sender_(video_config.incoming_ssrc),
+      num_audio_channels_(audio_config.channels),
+      audio_sampling_rate_(audio_config.frequency),
+      audio_codec_(audio_config.codec.audio),
+      video_codec_(video_config.codec.video) {}
+
+CastReceiverImpl::~CastReceiverImpl() {}
+
+void CastReceiverImpl::DispatchReceivedPacket(scoped_ptr<Packet> packet) {
+  const uint8_t* const data = &packet->front();
+  const size_t length = packet->size();
+
+  uint32 ssrc_of_sender;
+  if (Rtcp::IsRtcpPacket(data, length)) {
+    ssrc_of_sender = Rtcp::GetSsrcOfSender(data, length);
+  } else if (!FrameReceiver::ParseSenderSsrc(data, length, &ssrc_of_sender)) {
+    VLOG(1) << "Invalid RTP packet.";
+    return;
+  }
+
+  base::WeakPtr<FrameReceiver> target;
+  if (ssrc_of_sender == ssrc_of_video_sender_) {
+    target = video_receiver_.AsWeakPtr();
+  } else if (ssrc_of_sender == ssrc_of_audio_sender_) {
+    target = audio_receiver_.AsWeakPtr();
+  } else {
+    VLOG(1) << "Dropping packet with a non matching sender SSRC: "
+            << ssrc_of_sender;
+    return;
+  }
+  cast_environment_->PostTask(
+      CastEnvironment::MAIN,
+      FROM_HERE,
+      base::Bind(base::IgnoreResult(&FrameReceiver::ProcessPacket),
+                 target,
+                 base::Passed(&packet)));
+}
+
+transport::PacketReceiverCallback CastReceiverImpl::packet_receiver() {
+  return base::Bind(&CastReceiverImpl::DispatchReceivedPacket,
+                    // TODO(miu): This code structure is dangerous, since the
+                    // callback could be stored and then invoked after
+                    // destruction of |this|.
+                    base::Unretained(this));
+}
+
+void CastReceiverImpl::RequestDecodedAudioFrame(
+    const AudioFrameDecodedCallback& callback) {
+  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
+  DCHECK(!callback.is_null());
+  audio_receiver_.RequestEncodedFrame(base::Bind(
+      &CastReceiverImpl::DecodeEncodedAudioFrame,
+      // Note: Use of Unretained is safe since this Closure is guaranteed to be
+      // invoked or discarded by |audio_receiver_| before destruction of |this|.
+      base::Unretained(this),
+      callback));
+}
+
+void CastReceiverImpl::RequestEncodedAudioFrame(
+    const ReceiveEncodedFrameCallback& callback) {
+  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
+  audio_receiver_.RequestEncodedFrame(callback);
+}
+
+void CastReceiverImpl::RequestDecodedVideoFrame(
+    const VideoFrameDecodedCallback& callback) {
+  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
+  DCHECK(!callback.is_null());
+  video_receiver_.RequestEncodedFrame(base::Bind(
+      &CastReceiverImpl::DecodeEncodedVideoFrame,
+      // Note: Use of Unretained is safe since this Closure is guaranteed to be
+      // invoked or discarded by |video_receiver_| before destruction of |this|.
+      base::Unretained(this),
+      callback));
+}
+
+void CastReceiverImpl::RequestEncodedVideoFrame(
+    const ReceiveEncodedFrameCallback& callback) {
+  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
+  video_receiver_.RequestEncodedFrame(callback);
+}
+
+void CastReceiverImpl::DecodeEncodedAudioFrame(
+    const AudioFrameDecodedCallback& callback,
+    scoped_ptr<transport::EncodedFrame> encoded_frame) {
+  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
+  if (!encoded_frame) {
+    callback.Run(make_scoped_ptr<AudioBus>(NULL), base::TimeTicks(), false);
+    return;
+  }
+
+  if (!audio_decoder_) {
+    audio_decoder_.reset(new AudioDecoder(cast_environment_,
+                                          num_audio_channels_,
+                                          audio_sampling_rate_,
+                                          audio_codec_));
+  }
+  const uint32 frame_id = encoded_frame->frame_id;
+  const uint32 rtp_timestamp = encoded_frame->rtp_timestamp;
+  const base::TimeTicks playout_time = encoded_frame->reference_time;
+  audio_decoder_->DecodeFrame(
+      encoded_frame.Pass(),
+      base::Bind(&CastReceiverImpl::EmitDecodedAudioFrame,
+                 cast_environment_,
+                 callback,
+                 frame_id,
+                 rtp_timestamp,
+                 playout_time));
+}
+
+void CastReceiverImpl::DecodeEncodedVideoFrame(
+    const VideoFrameDecodedCallback& callback,
+    scoped_ptr<transport::EncodedFrame> encoded_frame) {
+  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
+  if (!encoded_frame) {
+    callback.Run(
+        make_scoped_refptr<VideoFrame>(NULL), base::TimeTicks(), false);
+    return;
+  }
+
+  // Used by chrome/browser/extension/api/cast_streaming/performance_test.cc
+  TRACE_EVENT_INSTANT2(
+      "cast_perf_test", "PullEncodedVideoFrame",
+      TRACE_EVENT_SCOPE_THREAD,
+      "rtp_timestamp", encoded_frame->rtp_timestamp,
+      "render_time", encoded_frame->reference_time.ToInternalValue());
+
+  if (!video_decoder_)
+    video_decoder_.reset(new VideoDecoder(cast_environment_, video_codec_));
+  const uint32 frame_id = encoded_frame->frame_id;
+  const uint32 rtp_timestamp = encoded_frame->rtp_timestamp;
+  const base::TimeTicks playout_time = encoded_frame->reference_time;
+  video_decoder_->DecodeFrame(
+      encoded_frame.Pass(),
+      base::Bind(&CastReceiverImpl::EmitDecodedVideoFrame,
+                 cast_environment_,
+                 callback,
+                 frame_id,
+                 rtp_timestamp,
+                 playout_time));
+}
+
+// static
+void CastReceiverImpl::EmitDecodedAudioFrame(
+    const scoped_refptr<CastEnvironment>& cast_environment,
+    const AudioFrameDecodedCallback& callback,
+    uint32 frame_id,
+    uint32 rtp_timestamp,
+    const base::TimeTicks& playout_time,
+    scoped_ptr<AudioBus> audio_bus,
+    bool is_continuous) {
+  DCHECK(cast_environment->CurrentlyOn(CastEnvironment::MAIN));
+  if (audio_bus.get()) {
+    const base::TimeTicks now = cast_environment->Clock()->NowTicks();
+    cast_environment->Logging()->InsertFrameEvent(
+        now, FRAME_DECODED, AUDIO_EVENT, rtp_timestamp, frame_id);
+    cast_environment->Logging()->InsertFrameEventWithDelay(
+        now, FRAME_PLAYOUT, AUDIO_EVENT, rtp_timestamp, frame_id,
+        playout_time - now);
+  }
+  callback.Run(audio_bus.Pass(), playout_time, is_continuous);
+}
+
+// static
+void CastReceiverImpl::EmitDecodedVideoFrame(
+    const scoped_refptr<CastEnvironment>& cast_environment,
+    const VideoFrameDecodedCallback& callback,
+    uint32 frame_id,
+    uint32 rtp_timestamp,
+    const base::TimeTicks& playout_time,
+    const scoped_refptr<VideoFrame>& video_frame,
+    bool is_continuous) {
+  DCHECK(cast_environment->CurrentlyOn(CastEnvironment::MAIN));
+  if (video_frame) {
+    const base::TimeTicks now = cast_environment->Clock()->NowTicks();
+    cast_environment->Logging()->InsertFrameEvent(
+        now, FRAME_DECODED, VIDEO_EVENT, rtp_timestamp, frame_id);
+    cast_environment->Logging()->InsertFrameEventWithDelay(
+        now, FRAME_PLAYOUT, VIDEO_EVENT, rtp_timestamp, frame_id,
+        playout_time - now);
+
+    // Used by chrome/browser/extension/api/cast_streaming/performance_test.cc
+    TRACE_EVENT_INSTANT1(
+        "cast_perf_test", "FrameDecoded",
+        TRACE_EVENT_SCOPE_THREAD,
+        "rtp_timestamp", rtp_timestamp);
+  }
+  callback.Run(video_frame, playout_time, is_continuous);
+}
+
+}  // namespace cast
+}  // namespace media
diff --git a/media/cast/receiver/cast_receiver_impl.h b/media/cast/receiver/cast_receiver_impl.h
new file mode 100644
index 0000000..c0dd5f3
--- /dev/null
+++ b/media/cast/receiver/cast_receiver_impl.h
@@ -0,0 +1,122 @@
+// Copyright 2014 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.
+
+#ifndef MEDIA_CAST_RECEIVER_CAST_RECEIVER_IMPL_H_
+#define MEDIA_CAST_RECEIVER_CAST_RECEIVER_IMPL_H_
+
+#include "base/memory/ref_counted.h"
+#include "base/memory/scoped_ptr.h"
+#include "media/cast/cast_config.h"
+#include "media/cast/cast_environment.h"
+#include "media/cast/cast_receiver.h"
+#include "media/cast/receiver/frame_receiver.h"
+#include "media/cast/transport/pacing/paced_sender.h"
+
+namespace media {
+namespace cast {
+
+class AudioDecoder;
+class VideoDecoder;
+
+// This is a pure owner class that groups all required receiver-related objects
+// together, such as the paced packet sender, audio/video RTP frame receivers,
+// and software decoders (created on-demand).
+class CastReceiverImpl : public CastReceiver {
+ public:
+  CastReceiverImpl(scoped_refptr<CastEnvironment> cast_environment,
+                   const FrameReceiverConfig& audio_config,
+                   const FrameReceiverConfig& video_config,
+                   transport::PacketSender* const packet_sender);
+
+  virtual ~CastReceiverImpl();
+
+  // CastReceiver implementation.
+  virtual transport::PacketReceiverCallback packet_receiver() OVERRIDE;
+  virtual void RequestDecodedAudioFrame(
+      const AudioFrameDecodedCallback& callback) OVERRIDE;
+  virtual void RequestEncodedAudioFrame(
+      const ReceiveEncodedFrameCallback& callback) OVERRIDE;
+  virtual void RequestDecodedVideoFrame(
+      const VideoFrameDecodedCallback& callback) OVERRIDE;
+  virtual void RequestEncodedVideoFrame(
+      const ReceiveEncodedFrameCallback& callback) OVERRIDE;
+
+ private:
+  // Forwards |packet| to a specific RTP frame receiver, or drops it if SSRC
+  // does not map to one of the receivers.
+  void DispatchReceivedPacket(scoped_ptr<Packet> packet);
+
+  // Feeds an EncodedFrame into |audio_decoder_|.  RequestDecodedAudioFrame()
+  // uses this as a callback for RequestEncodedAudioFrame().
+  void DecodeEncodedAudioFrame(
+      const AudioFrameDecodedCallback& callback,
+      scoped_ptr<transport::EncodedFrame> encoded_frame);
+
+  // Feeds an EncodedFrame into |video_decoder_|.  RequestDecodedVideoFrame()
+  // uses this as a callback for RequestEncodedVideoFrame().
+  void DecodeEncodedVideoFrame(
+      const VideoFrameDecodedCallback& callback,
+      scoped_ptr<transport::EncodedFrame> encoded_frame);
+
+  // Receives an AudioBus from |audio_decoder_|, logs the event, and passes the
+  // data on by running the given |callback|.  This method is static to ensure
+  // it can be called after a CastReceiverImpl instance is destroyed.
+  // DecodeEncodedAudioFrame() uses this as a callback for
+  // AudioDecoder::DecodeFrame().
+  static void EmitDecodedAudioFrame(
+      const scoped_refptr<CastEnvironment>& cast_environment,
+      const AudioFrameDecodedCallback& callback,
+      uint32 frame_id,
+      uint32 rtp_timestamp,
+      const base::TimeTicks& playout_time,
+      scoped_ptr<AudioBus> audio_bus,
+      bool is_continuous);
+
+  // Receives a VideoFrame from |video_decoder_|, logs the event, and passes the
+  // data on by running the given |callback|.  This method is static to ensure
+  // it can be called after a CastReceiverImpl instance is destroyed.
+  // DecodeEncodedVideoFrame() uses this as a callback for
+  // VideoDecoder::DecodeFrame().
+  static void EmitDecodedVideoFrame(
+      const scoped_refptr<CastEnvironment>& cast_environment,
+      const VideoFrameDecodedCallback& callback,
+      uint32 frame_id,
+      uint32 rtp_timestamp,
+      const base::TimeTicks& playout_time,
+      const scoped_refptr<VideoFrame>& video_frame,
+      bool is_continuous);
+
+  const scoped_refptr<CastEnvironment> cast_environment_;
+  transport::PacedSender pacer_;
+  FrameReceiver audio_receiver_;
+  FrameReceiver video_receiver_;
+
+  // Used by DispatchReceivedPacket() to direct packets to the appropriate frame
+  // receiver.
+  const uint32 ssrc_of_audio_sender_;
+  const uint32 ssrc_of_video_sender_;
+
+  // Parameters for the decoders that are created on-demand.  The values here
+  // might be nonsense if the client of CastReceiverImpl never intends to use
+  // the internal software-based decoders.
+  const int num_audio_channels_;
+  const int audio_sampling_rate_;
+  const transport::AudioCodec audio_codec_;
+  const transport::VideoCodec video_codec_;
+
+  // Created on-demand to decode frames from |audio_receiver_| into AudioBuses
+  // for playback.
+  scoped_ptr<AudioDecoder> audio_decoder_;
+
+  // Created on-demand to decode frames from |video_receiver_| into VideoFrame
+  // images for playback.
+  scoped_ptr<VideoDecoder> video_decoder_;
+
+  DISALLOW_COPY_AND_ASSIGN(CastReceiverImpl);
+};
+
+}  // namespace cast
+}  // namespace media
+
+#endif  // MEDIA_CAST_RECEIVER_CAST_RECEIVER_IMPL_
diff --git a/media/cast/audio_receiver/audio_receiver.cc b/media/cast/receiver/frame_receiver.cc
index 1f47827..e189cc9 100644
--- a/media/cast/audio_receiver/audio_receiver.cc
+++ b/media/cast/receiver/frame_receiver.cc
@@ -1,16 +1,16 @@
-// Copyright 2013 The Chromium Authors. All rights reserved.
+// Copyright 2014 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.
 
-#include "media/cast/audio_receiver/audio_receiver.h"
+#include "media/cast/receiver/frame_receiver.h"
 
 #include <algorithm>
 
+#include "base/big_endian.h"
 #include "base/bind.h"
 #include "base/logging.h"
 #include "base/message_loop/message_loop.h"
-#include "media/cast/audio_receiver/audio_decoder.h"
-#include "media/cast/transport/cast_transport_defines.h"
+#include "media/cast/cast_environment.h"
 
 namespace {
 const int kMinSchedulingDelayMs = 1;
@@ -19,56 +19,103 @@ const int kMinSchedulingDelayMs = 1;
 namespace media {
 namespace cast {
 
-AudioReceiver::AudioReceiver(scoped_refptr<CastEnvironment> cast_environment,
-                             const FrameReceiverConfig& audio_config,
-                             transport::PacedPacketSender* const packet_sender)
-    : RtpReceiver(cast_environment->Clock(), &audio_config, NULL),
-      cast_environment_(cast_environment),
-      event_subscriber_(kReceiverRtcpEventHistorySize, AUDIO_EVENT),
-      codec_(audio_config.codec.audio),
-      frequency_(audio_config.frequency),
+FrameReceiver::FrameReceiver(
+    const scoped_refptr<CastEnvironment>& cast_environment,
+    const FrameReceiverConfig& config,
+    EventMediaType event_media_type,
+    transport::PacedPacketSender* const packet_sender)
+    : cast_environment_(cast_environment),
+      packet_parser_(config.incoming_ssrc, config.rtp_payload_type),
+      stats_(cast_environment->Clock()),
+      event_media_type_(event_media_type),
+      event_subscriber_(kReceiverRtcpEventHistorySize, event_media_type),
+      rtp_timebase_(config.frequency),
       target_playout_delay_(
-          base::TimeDelta::FromMilliseconds(audio_config.rtp_max_delay_ms)),
+          base::TimeDelta::FromMilliseconds(config.rtp_max_delay_ms)),
       expected_frame_duration_(
-          base::TimeDelta::FromSeconds(1) / audio_config.max_frame_rate),
+          base::TimeDelta::FromSeconds(1) / config.max_frame_rate),
       reports_are_scheduled_(false),
       framer_(cast_environment->Clock(),
               this,
-              audio_config.incoming_ssrc,
+              config.incoming_ssrc,
               true,
-              audio_config.rtp_max_delay_ms * audio_config.max_frame_rate /
-                  1000),
-      rtcp_(cast_environment,
+              config.rtp_max_delay_ms * config.max_frame_rate / 1000),
+      rtcp_(cast_environment_,
             NULL,
             NULL,
             packet_sender,
-            GetStatistics(),
-            audio_config.rtcp_mode,
-            base::TimeDelta::FromMilliseconds(audio_config.rtcp_interval),
-            audio_config.feedback_ssrc,
-            audio_config.incoming_ssrc,
-            audio_config.rtcp_c_name,
-            true),
+            &stats_,
+            config.rtcp_mode,
+            base::TimeDelta::FromMilliseconds(config.rtcp_interval),
+            config.feedback_ssrc,
+            config.incoming_ssrc,
+            config.rtcp_c_name,
+            event_media_type),
       is_waiting_for_consecutive_frame_(false),
       lip_sync_drift_(ClockDriftSmoother::GetDefaultTimeConstant()),
       weak_factory_(this) {
-  DCHECK_GT(audio_config.rtp_max_delay_ms, 0);
-  DCHECK_GT(audio_config.max_frame_rate, 0);
-  audio_decoder_.reset(new AudioDecoder(cast_environment, audio_config));
-  decryptor_.Initialize(audio_config.aes_key, audio_config.aes_iv_mask);
+  DCHECK_GT(config.rtp_max_delay_ms, 0);
+  DCHECK_GT(config.max_frame_rate, 0);
+  decryptor_.Initialize(config.aes_key, config.aes_iv_mask);
   rtcp_.SetTargetDelay(target_playout_delay_);
   cast_environment_->Logging()->AddRawEventSubscriber(&event_subscriber_);
   memset(frame_id_to_rtp_timestamp_, 0, sizeof(frame_id_to_rtp_timestamp_));
 }
 
-AudioReceiver::~AudioReceiver() {
+FrameReceiver::~FrameReceiver() {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
   cast_environment_->Logging()->RemoveRawEventSubscriber(&event_subscriber_);
 }
 
-void AudioReceiver::OnReceivedPayloadData(const uint8* payload_data,
-                                          size_t payload_size,
-                                          const RtpCastHeader& rtp_header) {
+void FrameReceiver::RequestEncodedFrame(
+    const ReceiveEncodedFrameCallback& callback) {
+  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
+  frame_request_queue_.push_back(callback);
+  EmitAvailableEncodedFrames();
+}
+
+bool FrameReceiver::ProcessPacket(scoped_ptr<Packet> packet) {
+  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
+
+  if (Rtcp::IsRtcpPacket(&packet->front(), packet->size())) {
+    rtcp_.IncomingRtcpPacket(&packet->front(), packet->size());
+  } else {
+    RtpCastHeader rtp_header;
+    const uint8* payload_data;
+    size_t payload_size;
+    if (!packet_parser_.ParsePacket(&packet->front(),
+                                    packet->size(),
+                                    &rtp_header,
+                                    &payload_data,
+                                    &payload_size)) {
+      return false;
+    }
+
+    ProcessParsedPacket(rtp_header, payload_data, payload_size);
+    stats_.UpdateStatistics(rtp_header);
+  }
+
+  if (!reports_are_scheduled_) {
+    ScheduleNextRtcpReport();
+    ScheduleNextCastMessage();
+    reports_are_scheduled_ = true;
+  }
+
+  return true;
+}
+
+// static
+bool FrameReceiver::ParseSenderSsrc(const uint8* packet,
+                                    size_t length,
+                                    uint32* ssrc) {
+  base::BigEndianReader big_endian_reader(
+      reinterpret_cast<const char*>(packet), length);
+  return big_endian_reader.Skip(8) && big_endian_reader.ReadU32(ssrc);
+}
+
+void FrameReceiver::ProcessParsedPacket(const RtpCastHeader& rtp_header,
+                                        const uint8* payload_data,
+                                        size_t payload_size) {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
 
   const base::TimeTicks now = cast_environment_->Clock()->NowTicks();
@@ -76,7 +123,7 @@ void AudioReceiver::OnReceivedPayloadData(const uint8* payload_data,
   frame_id_to_rtp_timestamp_[rtp_header.frame_id & 0xff] =
       rtp_header.rtp_timestamp;
   cast_environment_->Logging()->InsertPacketEvent(
-      now, PACKET_RECEIVED, AUDIO_EVENT, rtp_header.rtp_timestamp,
+      now, PACKET_RECEIVED, event_media_type_, rtp_header.rtp_timestamp,
       rtp_header.frame_id, rtp_header.packet_id, rtp_header.max_packet_id,
       payload_size);
 
@@ -111,81 +158,35 @@ void AudioReceiver::OnReceivedPayloadData(const uint8* payload_data,
     } else {
       lip_sync_reference_time_ += RtpDeltaToTimeDelta(
           static_cast<int32>(fresh_sync_rtp - lip_sync_rtp_timestamp_),
-          frequency_);
+          rtp_timebase_);
     }
     lip_sync_rtp_timestamp_ = fresh_sync_rtp;
     lip_sync_drift_.Update(
         now, fresh_sync_reference - lip_sync_reference_time_);
   }
 
-  // Frame not complete; wait for more packets.
-  if (!complete)
-    return;
-
-  EmitAvailableEncodedFrames();
-}
-
-void AudioReceiver::GetRawAudioFrame(
-    const AudioFrameDecodedCallback& callback) {
-  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
-  DCHECK(!callback.is_null());
-  DCHECK(audio_decoder_.get());
-  GetEncodedAudioFrame(base::Bind(
-      &AudioReceiver::DecodeEncodedAudioFrame,
-      // Note: Use of Unretained is safe since this Closure is guaranteed to be
-      // invoked before destruction of |this|.
-      base::Unretained(this),
-      callback));
+  // Another frame is complete from a non-duplicate packet.  Attempt to emit
+  // more frames to satisfy enqueued requests.
+  if (complete)
+    EmitAvailableEncodedFrames();
 }
 
-void AudioReceiver::DecodeEncodedAudioFrame(
-    const AudioFrameDecodedCallback& callback,
-    scoped_ptr<transport::EncodedFrame> encoded_frame) {
+void FrameReceiver::CastFeedback(const RtcpCastMessage& cast_message) {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
-  if (!encoded_frame) {
-    callback.Run(make_scoped_ptr<AudioBus>(NULL), base::TimeTicks(), false);
-    return;
-  }
-  const uint32 frame_id = encoded_frame->frame_id;
-  const uint32 rtp_timestamp = encoded_frame->rtp_timestamp;
-  const base::TimeTicks playout_time = encoded_frame->reference_time;
-  audio_decoder_->DecodeFrame(encoded_frame.Pass(),
-                              base::Bind(&AudioReceiver::EmitRawAudioFrame,
-                                         cast_environment_,
-                                         callback,
-                                         frame_id,
-                                         rtp_timestamp,
-                                         playout_time));
-}
 
-// static
-void AudioReceiver::EmitRawAudioFrame(
-    const scoped_refptr<CastEnvironment>& cast_environment,
-    const AudioFrameDecodedCallback& callback,
-    uint32 frame_id,
-    uint32 rtp_timestamp,
-    const base::TimeTicks& playout_time,
-    scoped_ptr<AudioBus> audio_bus,
-    bool is_continuous) {
-  DCHECK(cast_environment->CurrentlyOn(CastEnvironment::MAIN));
-  if (audio_bus.get()) {
-    const base::TimeTicks now = cast_environment->Clock()->NowTicks();
-    cast_environment->Logging()->InsertFrameEvent(
-        now, FRAME_DECODED, AUDIO_EVENT, rtp_timestamp, frame_id);
-    cast_environment->Logging()->InsertFrameEventWithDelay(
-        now, FRAME_PLAYOUT, AUDIO_EVENT, rtp_timestamp, frame_id,
-        playout_time - now);
-  }
-  callback.Run(audio_bus.Pass(), playout_time, is_continuous);
-}
+  base::TimeTicks now = cast_environment_->Clock()->NowTicks();
+  RtpTimestamp rtp_timestamp =
+      frame_id_to_rtp_timestamp_[cast_message.ack_frame_id_ & 0xff];
+  cast_environment_->Logging()->InsertFrameEvent(
+      now, FRAME_ACK_SENT, event_media_type_,
+      rtp_timestamp, cast_message.ack_frame_id_);
 
-void AudioReceiver::GetEncodedAudioFrame(const FrameEncodedCallback& callback) {
-  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
-  frame_request_queue_.push_back(callback);
-  EmitAvailableEncodedFrames();
+  ReceiverRtcpEventSubscriber::RtcpEventMultiMap rtcp_events;
+  event_subscriber_.GetRtcpEventsAndReset(&rtcp_events);
+  rtcp_.SendRtcpFromRtpReceiver(&cast_message, &rtcp_events);
 }
 
-void AudioReceiver::EmitAvailableEncodedFrames() {
+void FrameReceiver::EmitAvailableEncodedFrames() {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
 
   while (!frame_request_queue_.empty()) {
@@ -199,8 +200,8 @@ void AudioReceiver::EmitAvailableEncodedFrames() {
     if (!framer_.GetEncodedFrame(encoded_frame.get(),
                                  &is_consecutively_next_frame,
                                  &have_multiple_complete_frames)) {
-      VLOG(1) << "Wait for more audio packets to produce a completed frame.";
-      return;  // OnReceivedPayloadData() will invoke this method in the future.
+      VLOG(1) << "Wait for more packets to produce a completed frame.";
+      return;  // ProcessParsedPacket() will invoke this method in the future.
     }
 
     const base::TimeTicks now = cast_environment_->Clock()->NowTicks();
@@ -229,7 +230,7 @@ void AudioReceiver::EmitAvailableEncodedFrames() {
           cast_environment_->PostDelayedTask(
               CastEnvironment::MAIN,
               FROM_HERE,
-              base::Bind(&AudioReceiver::EmitAvailableEncodedFramesAfterWaiting,
+              base::Bind(&FrameReceiver::EmitAvailableEncodedFramesAfterWaiting,
                          weak_factory_.GetWeakPtr()),
               playout_time - now);
         }
@@ -239,16 +240,15 @@ void AudioReceiver::EmitAvailableEncodedFrames() {
 
     // Decrypt the payload data in the frame, if crypto is being used.
     if (decryptor_.initialized()) {
-      std::string decrypted_audio_data;
+      std::string decrypted_data;
       if (!decryptor_.Decrypt(encoded_frame->frame_id,
                               encoded_frame->data,
-                              &decrypted_audio_data)) {
-        // Decryption failed.  Give up on this frame, releasing it from the
-        // jitter buffer.
+                              &decrypted_data)) {
+        // Decryption failed.  Give up on this frame.
         framer_.ReleaseFrame(encoded_frame->frame_id);
         continue;
       }
-      encoded_frame->data.swap(decrypted_audio_data);
+      encoded_frame->data.swap(decrypted_data);
     }
 
     // At this point, we have a decrypted EncodedFrame ready to be emitted.
@@ -262,95 +262,64 @@ void AudioReceiver::EmitAvailableEncodedFrames() {
   }
 }
 
-void AudioReceiver::EmitAvailableEncodedFramesAfterWaiting() {
+void FrameReceiver::EmitAvailableEncodedFramesAfterWaiting() {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
   DCHECK(is_waiting_for_consecutive_frame_);
   is_waiting_for_consecutive_frame_ = false;
   EmitAvailableEncodedFrames();
 }
 
-base::TimeTicks AudioReceiver::GetPlayoutTime(uint32 rtp_timestamp) const {
+base::TimeTicks FrameReceiver::GetPlayoutTime(uint32 rtp_timestamp) const {
   return lip_sync_reference_time_ +
       lip_sync_drift_.Current() +
       RtpDeltaToTimeDelta(
           static_cast<int32>(rtp_timestamp - lip_sync_rtp_timestamp_),
-          frequency_) +
+          rtp_timebase_) +
       target_playout_delay_;
 }
 
-void AudioReceiver::IncomingPacket(scoped_ptr<Packet> packet) {
+void FrameReceiver::ScheduleNextCastMessage() {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
-  if (Rtcp::IsRtcpPacket(&packet->front(), packet->size())) {
-    rtcp_.IncomingRtcpPacket(&packet->front(), packet->size());
-  } else {
-    ReceivedPacket(&packet->front(), packet->size());
-  }
-  if (!reports_are_scheduled_) {
-    ScheduleNextRtcpReport();
-    ScheduleNextCastMessage();
-    reports_are_scheduled_ = true;
-  }
-}
-
-void AudioReceiver::CastFeedback(const RtcpCastMessage& cast_message) {
-  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
-  base::TimeTicks now = cast_environment_->Clock()->NowTicks();
-  RtpTimestamp rtp_timestamp =
-      frame_id_to_rtp_timestamp_[cast_message.ack_frame_id_ & 0xff];
-  cast_environment_->Logging()->InsertFrameEvent(
-      now, FRAME_ACK_SENT, AUDIO_EVENT, rtp_timestamp,
-      cast_message.ack_frame_id_);
-
-  ReceiverRtcpEventSubscriber::RtcpEventMultiMap rtcp_events;
-  event_subscriber_.GetRtcpEventsAndReset(&rtcp_events);
-  rtcp_.SendRtcpFromRtpReceiver(&cast_message, &rtcp_events);
-}
-
-void AudioReceiver::ScheduleNextRtcpReport() {
-  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
-  base::TimeDelta time_to_send = rtcp_.TimeToSendNextRtcpReport() -
-                                 cast_environment_->Clock()->NowTicks();
-
+  base::TimeTicks send_time;
+  framer_.TimeToSendNextCastMessage(&send_time);
+  base::TimeDelta time_to_send =
+      send_time - cast_environment_->Clock()->NowTicks();
   time_to_send = std::max(
       time_to_send, base::TimeDelta::FromMilliseconds(kMinSchedulingDelayMs));
-
   cast_environment_->PostDelayedTask(
       CastEnvironment::MAIN,
       FROM_HERE,
-      base::Bind(&AudioReceiver::SendNextRtcpReport,
+      base::Bind(&FrameReceiver::SendNextCastMessage,
                  weak_factory_.GetWeakPtr()),
       time_to_send);
 }
 
-void AudioReceiver::SendNextRtcpReport() {
+void FrameReceiver::SendNextCastMessage() {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
-  // TODO(pwestin): add logging.
-  rtcp_.SendRtcpFromRtpReceiver(NULL, NULL);
-  ScheduleNextRtcpReport();
+  framer_.SendCastMessage();  // Will only send a message if it is time.
+  ScheduleNextCastMessage();
 }
 
-// Cast messages should be sent within a maximum interval. Schedule a call
-// if not triggered elsewhere, e.g. by the cast message_builder.
-void AudioReceiver::ScheduleNextCastMessage() {
+void FrameReceiver::ScheduleNextRtcpReport() {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
-  base::TimeTicks send_time;
-  framer_.TimeToSendNextCastMessage(&send_time);
-  base::TimeDelta time_to_send =
-      send_time - cast_environment_->Clock()->NowTicks();
-  time_to_send = std::max(
-      time_to_send, base::TimeDelta::FromMilliseconds(kMinSchedulingDelayMs));
+  base::TimeDelta time_to_next = rtcp_.TimeToSendNextRtcpReport() -
+                                 cast_environment_->Clock()->NowTicks();
+
+  time_to_next = std::max(
+      time_to_next, base::TimeDelta::FromMilliseconds(kMinSchedulingDelayMs));
+
   cast_environment_->PostDelayedTask(
       CastEnvironment::MAIN,
       FROM_HERE,
-      base::Bind(&AudioReceiver::SendNextCastMessage,
+      base::Bind(&FrameReceiver::SendNextRtcpReport,
                  weak_factory_.GetWeakPtr()),
-      time_to_send);
+      time_to_next);
 }
 
-void AudioReceiver::SendNextCastMessage() {
+void FrameReceiver::SendNextRtcpReport() {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
-  framer_.SendCastMessage();  // Will only send a message if it is time.
-  ScheduleNextCastMessage();
+  rtcp_.SendRtcpFromRtpReceiver(NULL, NULL);
+  ScheduleNextRtcpReport();
 }
 
 }  // namespace cast
diff --git a/media/cast/video_receiver/video_receiver.h b/media/cast/receiver/frame_receiver.h
index 14a4bfa..ac14ab1 100644
--- a/media/cast/video_receiver/video_receiver.h
+++ b/media/cast/receiver/frame_receiver.h
@@ -1,41 +1,35 @@
-// Copyright 2013 The Chromium Authors. All rights reserved.
+// Copyright 2014 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.
 
-#ifndef MEDIA_CAST_VIDEO_RECEIVER_VIDEO_RECEIVER_H_
-#define MEDIA_CAST_VIDEO_RECEIVER_VIDEO_RECEIVER_H_
+#ifndef MEDIA_CAST_RECEIVER_FRAME_RECEIVER_H_
+#define MEDIA_CAST_RECEIVER_FRAME_RECEIVER_H_
 
-#include "base/basictypes.h"
-#include "base/callback.h"
 #include "base/memory/ref_counted.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/memory/weak_ptr.h"
-#include "base/threading/non_thread_safe.h"
-#include "base/time/tick_clock.h"
 #include "base/time/time.h"
 #include "media/cast/base/clock_drift_smoother.h"
 #include "media/cast/cast_config.h"
-#include "media/cast/cast_environment.h"
 #include "media/cast/cast_receiver.h"
 #include "media/cast/framer/framer.h"
+#include "media/cast/logging/logging_defines.h"
 #include "media/cast/rtcp/receiver_rtcp_event_subscriber.h"
 #include "media/cast/rtcp/rtcp.h"
-#include "media/cast/rtp_receiver/rtp_receiver.h"
+#include "media/cast/rtp_receiver/receiver_stats.h"
+#include "media/cast/rtp_receiver/rtp_parser/rtp_parser.h"
 #include "media/cast/rtp_receiver/rtp_receiver_defines.h"
 #include "media/cast/transport/utility/transport_encryption_handler.h"
 
 namespace media {
-
-class VideoFrame;
-
 namespace cast {
 
-class VideoDecoder;
+class CastEnvironment;
 
-// VideoReceiver receives packets out-of-order while clients make requests for
+// FrameReceiver receives packets out-of-order while clients make requests for
 // complete frames in-order.  (A frame consists of one or more packets.)
 //
-// VideoReceiver also includes logic for computing the playout time for each
+// FrameReceiver also includes logic for computing the playout time for each
 // frame, accounting for a constant targeted playout delay.  The purpose of the
 // playout delay is to provide a fixed window of time between the capture event
 // on the sender and the playout on the receiver.  This is important because
@@ -43,47 +37,43 @@ class VideoDecoder;
 // the sender, then receive and re-order packets on the receiver, then decode
 // frame) can vary in duration and is typically very hard to predict.
 //
-// Two types of frames can be requested: 1) A frame of decoded video data; or 2)
-// a frame of still-encoded video data, to be passed into an external video
-// decoder.  Each request for a frame includes a callback which VideoReceiver
-// guarantees will be called at some point in the future unless the
-// VideoReceiver is destroyed. Clients should generally limit the number of
-// outstanding requests (perhaps to just one or two).
+// Each request for a frame includes a callback which FrameReceiver guarantees
+// will be called at some point in the future unless the FrameReceiver is
+// destroyed.  Clients should generally limit the number of outstanding requests
+// (perhaps to just one or two).
 //
 // This class is not thread safe.  Should only be called from the Main cast
 // thread.
-class VideoReceiver : public RtpReceiver,
-                      public RtpPayloadFeedback,
-                      public base::NonThreadSafe,
-                      public base::SupportsWeakPtr<VideoReceiver> {
+class FrameReceiver : public RtpPayloadFeedback,
+                      public base::SupportsWeakPtr<FrameReceiver> {
  public:
-  VideoReceiver(scoped_refptr<CastEnvironment> cast_environment,
-                const FrameReceiverConfig& video_config,
+  FrameReceiver(const scoped_refptr<CastEnvironment>& cast_environment,
+                const FrameReceiverConfig& config,
+                EventMediaType event_media_type,
                 transport::PacedPacketSender* const packet_sender);
 
-  virtual ~VideoReceiver();
+  virtual ~FrameReceiver();
 
-  // Request a decoded video frame.
+  // Request an encoded frame.
   //
   // The given |callback| is guaranteed to be run at some point in the future,
-  // even if to respond with NULL at shutdown time.
-  void GetRawVideoFrame(const VideoFrameDecodedCallback& callback);
+  // except for those requests still enqueued at destruction time.
+  void RequestEncodedFrame(const ReceiveEncodedFrameCallback& callback);
 
-  // Request an encoded video frame.
-  //
-  // The given |callback| is guaranteed to be run at some point in the future,
-  // even if to respond with NULL at shutdown time.
-  void GetEncodedVideoFrame(const FrameEncodedCallback& callback);
+  // Called to deliver another packet, possibly a duplicate, and possibly
+  // out-of-order.  Returns true if the parsing of the packet succeeded.
+  bool ProcessPacket(scoped_ptr<Packet> packet);
 
-  // Deliver another packet, possibly a duplicate, and possibly out-of-order.
-  void IncomingPacket(scoped_ptr<Packet> packet);
+  // TODO(miu): This is the wrong place for this, but the (de)serialization
+  // implementation needs to be consolidated first.
+  static bool ParseSenderSsrc(const uint8* packet, size_t length, uint32* ssrc);
 
  protected:
-  friend class VideoReceiverTest;  // Invokes OnReceivedPayloadData().
+  friend class FrameReceiverTest;  // Invokes ProcessParsedPacket().
 
-  virtual void OnReceivedPayloadData(const uint8* payload_data,
-                                     size_t payload_size,
-                                     const RtpCastHeader& rtp_header) OVERRIDE;
+  void ProcessParsedPacket(const RtpCastHeader& rtp_header,
+                           const uint8* payload_data,
+                           size_t payload_size);
 
   // RtpPayloadFeedback implementation.
   virtual void CastFeedback(const RtcpCastMessage& cast_message) OVERRIDE;
@@ -99,12 +89,6 @@ class VideoReceiver : public RtpReceiver,
   // EmitAvailableEncodedFrames().
   void EmitAvailableEncodedFramesAfterWaiting();
 
-  // Feeds an EncodedFrame into |video_decoder_|.  GetRawVideoFrame() uses this
-  // as a callback for GetEncodedVideoFrame().
-  void DecodeEncodedVideoFrame(
-      const VideoFrameDecodedCallback& callback,
-      scoped_ptr<transport::EncodedFrame> encoded_frame);
-
   // Computes the playout time for a frame with the given |rtp_timestamp|.
   // Because lip-sync info is refreshed regularly, calling this method with the
   // same argument may return different results.
@@ -122,28 +106,23 @@ class VideoReceiver : public RtpReceiver,
   // Actually send the next RTCP report.
   void SendNextRtcpReport();
 
-  // Receives a VideoFrame from |video_decoder_|, logs the event, and passes the
-  // data on by running the given |callback|.  This method is static to ensure
-  // it can be called after a VideoReceiver instance is destroyed.
-  // DecodeEncodedVideoFrame() uses this as a callback for
-  // VideoDecoder::DecodeFrame().
-  static void EmitRawVideoFrame(
-      const scoped_refptr<CastEnvironment>& cast_environment,
-      const VideoFrameDecodedCallback& callback,
-      uint32 frame_id,
-      uint32 rtp_timestamp,
-      const base::TimeTicks& playout_time,
-      const scoped_refptr<VideoFrame>& video_frame,
-      bool is_continuous);
-
   const scoped_refptr<CastEnvironment> cast_environment_;
 
+  // Deserializes a packet into a RtpHeader + payload bytes.
+  RtpParser packet_parser_;
+
+  // Accumulates packet statistics, including packet loss, counts, and jitter.
+  ReceiverStats stats_;
+
+  // Partitions logged events by the type of media passing through.
+  EventMediaType event_media_type_;
+
   // Subscribes to raw events.
-  // Processes raw audio events to be sent over to the cast sender via RTCP.
+  // Processes raw events to be sent over to the cast sender via RTCP.
   ReceiverRtcpEventSubscriber event_subscriber_;
 
-  // Configured video codec.
-  const transport::VideoCodec codec_;
+  // RTP timebase: The number of RTP units advanced per one second.
+  const int rtp_timebase_;
 
   // The total amount of time between a frame's capture/recording on the sender
   // and its playback on the receiver (i.e., shown to a user).  This is fixed as
@@ -154,6 +133,8 @@ class VideoReceiver : public RtpReceiver,
   const base::TimeDelta target_playout_delay_;
 
   // Hack: This is used in logic that determines whether to skip frames.
+  // TODO(miu): Revisit this.  Logic needs to also account for expected decode
+  // time.
   const base::TimeDelta expected_frame_duration_;
 
   // Set to false initially, then set to true after scheduling the periodic
@@ -166,9 +147,6 @@ class VideoReceiver : public RtpReceiver,
   // decodable EncodedFrames.
   Framer framer_;
 
-  // Decodes frames into media::VideoFrame images for playback.
-  scoped_ptr<VideoDecoder> video_decoder_;
-
   // Manages sending/receiving of RTCP packets, including sender/receiver
   // reports.
   Rtcp rtcp_;
@@ -177,7 +155,7 @@ class VideoReceiver : public RtpReceiver,
   transport::TransportEncryptionHandler decryptor_;
 
   // Outstanding callbacks to run to deliver on client requests for frames.
-  std::list<FrameEncodedCallback> frame_request_queue_;
+  std::list<ReceiveEncodedFrameCallback> frame_request_queue_;
 
   // True while there's an outstanding task to re-invoke
   // EmitAvailableEncodedFrames().
@@ -195,12 +173,12 @@ class VideoReceiver : public RtpReceiver,
   ClockDriftSmoother lip_sync_drift_;
 
   // NOTE: Weak pointers must be invalidated before all other member variables.
-  base::WeakPtrFactory<VideoReceiver> weak_factory_;
+  base::WeakPtrFactory<FrameReceiver> weak_factory_;
 
-  DISALLOW_COPY_AND_ASSIGN(VideoReceiver);
+  DISALLOW_COPY_AND_ASSIGN(FrameReceiver);
 };
 
 }  // namespace cast
 }  // namespace media
 
-#endif  // MEDIA_CAST_VIDEO_RECEIVER_VIDEO_RECEIVER_H_
+#endif  // MEDIA_CAST_RECEIVER_FRAME_RECEIVER_H_
diff --git a/media/cast/receiver/frame_receiver_unittest.cc b/media/cast/receiver/frame_receiver_unittest.cc
new file mode 100644
index 0000000..4d8273e1
--- /dev/null
+++ b/media/cast/receiver/frame_receiver_unittest.cc
@@ -0,0 +1,419 @@
+// Copyright 2014 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.
+
+#include <deque>
+#include <utility>
+
+#include "base/bind.h"
+#include "base/memory/ref_counted.h"
+#include "base/memory/scoped_ptr.h"
+#include "base/test/simple_test_tick_clock.h"
+#include "media/cast/cast_defines.h"
+#include "media/cast/cast_environment.h"
+#include "media/cast/logging/simple_event_subscriber.h"
+#include "media/cast/receiver/frame_receiver.h"
+#include "media/cast/rtcp/test_rtcp_packet_builder.h"
+#include "media/cast/test/fake_single_thread_task_runner.h"
+#include "media/cast/test/utility/default_config.h"
+#include "media/cast/transport/pacing/mock_paced_packet_sender.h"
+#include "testing/gmock/include/gmock/gmock.h"
+
+using ::testing::_;
+
+namespace media {
+namespace cast {
+
+namespace {
+
+const int kPacketSize = 1500;
+const uint32 kFirstFrameId = 1234;
+const int kPlayoutDelayMillis = 100;
+
+class FakeFrameClient {
+ public:
+  FakeFrameClient() : num_called_(0) {}
+  virtual ~FakeFrameClient() {}
+
+  void AddExpectedResult(uint32 expected_frame_id,
+                         const base::TimeTicks& expected_playout_time) {
+    expected_results_.push_back(
+        std::make_pair(expected_frame_id, expected_playout_time));
+  }
+
+  void DeliverEncodedFrame(scoped_ptr<transport::EncodedFrame> frame) {
+    SCOPED_TRACE(::testing::Message() << "num_called_ is " << num_called_);
+    ASSERT_FALSE(!frame)
+        << "If at shutdown: There were unsatisfied requests enqueued.";
+    ASSERT_FALSE(expected_results_.empty());
+    EXPECT_EQ(expected_results_.front().first, frame->frame_id);
+    EXPECT_EQ(expected_results_.front().second, frame->reference_time);
+    expected_results_.pop_front();
+    ++num_called_;
+  }
+
+  int number_times_called() const { return num_called_; }
+
+ private:
+  std::deque<std::pair<uint32, base::TimeTicks> > expected_results_;
+  int num_called_;
+
+  DISALLOW_COPY_AND_ASSIGN(FakeFrameClient);
+};
+}  // namespace
+
+class FrameReceiverTest : public ::testing::Test {
+ protected:
+  FrameReceiverTest() {
+    testing_clock_ = new base::SimpleTestTickClock();
+    testing_clock_->Advance(base::TimeTicks::Now() - base::TimeTicks());
+    start_time_ = testing_clock_->NowTicks();
+    task_runner_ = new test::FakeSingleThreadTaskRunner(testing_clock_);
+
+    cast_environment_ =
+        new CastEnvironment(scoped_ptr<base::TickClock>(testing_clock_).Pass(),
+                            task_runner_,
+                            task_runner_,
+                            task_runner_);
+  }
+
+  virtual ~FrameReceiverTest() {}
+
+  virtual void SetUp() {
+    payload_.assign(kPacketSize, 0);
+
+    // Always start with a key frame.
+    rtp_header_.is_key_frame = true;
+    rtp_header_.frame_id = kFirstFrameId;
+    rtp_header_.packet_id = 0;
+    rtp_header_.max_packet_id = 0;
+    rtp_header_.reference_frame_id = rtp_header_.frame_id;
+    rtp_header_.rtp_timestamp = 0;
+  }
+
+  void CreateFrameReceiverOfAudio() {
+    config_ = GetDefaultAudioReceiverConfig();
+    config_.rtp_max_delay_ms = kPlayoutDelayMillis;
+
+    receiver_.reset(new FrameReceiver(
+        cast_environment_, config_, AUDIO_EVENT, &mock_transport_));
+  }
+
+  void CreateFrameReceiverOfVideo() {
+    config_ = GetDefaultVideoReceiverConfig();
+    config_.rtp_max_delay_ms = kPlayoutDelayMillis;
+    // Note: Frame rate must divide 1000 without remainder so the test code
+    // doesn't have to account for rounding errors.
+    config_.max_frame_rate = 25;
+
+    receiver_.reset(new FrameReceiver(
+        cast_environment_, config_, VIDEO_EVENT, &mock_transport_));
+  }
+
+  void FeedOneFrameIntoReceiver() {
+    // Note: For testing purposes, a frame consists of only a single packet.
+    receiver_->ProcessParsedPacket(
+        rtp_header_, payload_.data(), payload_.size());
+  }
+
+  void FeedLipSyncInfoIntoReceiver() {
+    const base::TimeTicks now = testing_clock_->NowTicks();
+    const int64 rtp_timestamp = (now - start_time_) *
+        config_.frequency / base::TimeDelta::FromSeconds(1);
+    CHECK_LE(0, rtp_timestamp);
+    uint32 ntp_seconds;
+    uint32 ntp_fraction;
+    ConvertTimeTicksToNtp(now, &ntp_seconds, &ntp_fraction);
+    TestRtcpPacketBuilder rtcp_packet;
+    rtcp_packet.AddSrWithNtp(config_.incoming_ssrc,
+                             ntp_seconds, ntp_fraction,
+                             static_cast<uint32>(rtp_timestamp));
+    ASSERT_TRUE(receiver_->ProcessPacket(rtcp_packet.GetPacket().Pass()));
+  }
+
+  FrameReceiverConfig config_;
+  std::vector<uint8> payload_;
+  RtpCastHeader rtp_header_;
+  base::SimpleTestTickClock* testing_clock_;  // Owned by CastEnvironment.
+  base::TimeTicks start_time_;
+  transport::MockPacedPacketSender mock_transport_;
+  scoped_refptr<test::FakeSingleThreadTaskRunner> task_runner_;
+  scoped_refptr<CastEnvironment> cast_environment_;
+  FakeFrameClient frame_client_;
+
+  // Important for the FrameReceiver to be declared last, since its dependencies
+  // must remain alive until after its destruction.
+  scoped_ptr<FrameReceiver> receiver_;
+
+  DISALLOW_COPY_AND_ASSIGN(FrameReceiverTest);
+};
+
+TEST_F(FrameReceiverTest, RejectsUnparsablePackets) {
+  CreateFrameReceiverOfVideo();
+
+  SimpleEventSubscriber event_subscriber;
+  cast_environment_->Logging()->AddRawEventSubscriber(&event_subscriber);
+
+  const bool success = receiver_->ProcessPacket(
+      scoped_ptr<Packet>(new Packet(kPacketSize, 0xff)).Pass());
+  EXPECT_FALSE(success);
+
+  // Confirm no log events.
+  std::vector<FrameEvent> frame_events;
+  event_subscriber.GetFrameEventsAndReset(&frame_events);
+  EXPECT_TRUE(frame_events.empty());
+  cast_environment_->Logging()->RemoveRawEventSubscriber(&event_subscriber);
+}
+
+TEST_F(FrameReceiverTest, ReceivesOneFrame) {
+  CreateFrameReceiverOfAudio();
+
+  SimpleEventSubscriber event_subscriber;
+  cast_environment_->Logging()->AddRawEventSubscriber(&event_subscriber);
+
+  EXPECT_CALL(mock_transport_, SendRtcpPacket(_, _))
+      .WillRepeatedly(testing::Return(true));
+
+  FeedLipSyncInfoIntoReceiver();
+  task_runner_->RunTasks();
+
+  // Enqueue a request for a frame.
+  receiver_->RequestEncodedFrame(
+      base::Bind(&FakeFrameClient::DeliverEncodedFrame,
+                 base::Unretained(&frame_client_)));
+
+  // The request should not be satisfied since no packets have been received.
+  task_runner_->RunTasks();
+  EXPECT_EQ(0, frame_client_.number_times_called());
+
+  // Deliver one frame to the receiver and expect to get one frame back.
+  const base::TimeDelta target_playout_delay =
+      base::TimeDelta::FromMilliseconds(kPlayoutDelayMillis);
+  frame_client_.AddExpectedResult(
+      kFirstFrameId, testing_clock_->NowTicks() + target_playout_delay);
+  FeedOneFrameIntoReceiver();
+  task_runner_->RunTasks();
+  EXPECT_EQ(1, frame_client_.number_times_called());
+
+  // Was the frame logged?
+  std::vector<FrameEvent> frame_events;
+  event_subscriber.GetFrameEventsAndReset(&frame_events);
+  ASSERT_TRUE(!frame_events.empty());
+  EXPECT_EQ(FRAME_ACK_SENT, frame_events.begin()->type);
+  EXPECT_EQ(AUDIO_EVENT, frame_events.begin()->media_type);
+  EXPECT_EQ(rtp_header_.frame_id, frame_events.begin()->frame_id);
+  EXPECT_EQ(rtp_header_.rtp_timestamp, frame_events.begin()->rtp_timestamp);
+  cast_environment_->Logging()->RemoveRawEventSubscriber(&event_subscriber);
+}
+
+TEST_F(FrameReceiverTest, ReceivesFramesSkippingWhenAppropriate) {
+  CreateFrameReceiverOfAudio();
+
+  SimpleEventSubscriber event_subscriber;
+  cast_environment_->Logging()->AddRawEventSubscriber(&event_subscriber);
+
+  EXPECT_CALL(mock_transport_, SendRtcpPacket(_, _))
+      .WillRepeatedly(testing::Return(true));
+
+  const uint32 rtp_advance_per_frame =
+      config_.frequency / config_.max_frame_rate;
+  const base::TimeDelta time_advance_per_frame =
+      base::TimeDelta::FromSeconds(1) / config_.max_frame_rate;
+
+  // Feed and process lip sync in receiver.
+  FeedLipSyncInfoIntoReceiver();
+  task_runner_->RunTasks();
+  const base::TimeTicks first_frame_capture_time = testing_clock_->NowTicks();
+
+  // Enqueue a request for a frame.
+  const ReceiveEncodedFrameCallback frame_encoded_callback =
+      base::Bind(&FakeFrameClient::DeliverEncodedFrame,
+                 base::Unretained(&frame_client_));
+  receiver_->RequestEncodedFrame(frame_encoded_callback);
+  task_runner_->RunTasks();
+  EXPECT_EQ(0, frame_client_.number_times_called());
+
+  // Receive one frame and expect to see the first request satisfied.
+  const base::TimeDelta target_playout_delay =
+      base::TimeDelta::FromMilliseconds(kPlayoutDelayMillis);
+  frame_client_.AddExpectedResult(
+      kFirstFrameId, first_frame_capture_time + target_playout_delay);
+  rtp_header_.rtp_timestamp = 0;
+  FeedOneFrameIntoReceiver();  // Frame 1
+  task_runner_->RunTasks();
+  EXPECT_EQ(1, frame_client_.number_times_called());
+
+  // Enqueue a second request for a frame, but it should not be fulfilled yet.
+  receiver_->RequestEncodedFrame(frame_encoded_callback);
+  task_runner_->RunTasks();
+  EXPECT_EQ(1, frame_client_.number_times_called());
+
+  // Receive one frame out-of-order: Make sure that we are not continuous and
+  // that the RTP timestamp represents a time in the future.
+  rtp_header_.frame_id = kFirstFrameId + 2;  // "Frame 3"
+  rtp_header_.reference_frame_id = rtp_header_.frame_id;
+  rtp_header_.rtp_timestamp += 2 * rtp_advance_per_frame;
+  frame_client_.AddExpectedResult(
+      kFirstFrameId + 2,
+      first_frame_capture_time + 2 * time_advance_per_frame +
+          target_playout_delay);
+  FeedOneFrameIntoReceiver();  // Frame 3
+
+  // Frame 2 should not come out at this point in time.
+  task_runner_->RunTasks();
+  EXPECT_EQ(1, frame_client_.number_times_called());
+
+  // Enqueue a third request for a frame.
+  receiver_->RequestEncodedFrame(frame_encoded_callback);
+  task_runner_->RunTasks();
+  EXPECT_EQ(1, frame_client_.number_times_called());
+
+  // Now, advance time forward such that the receiver is convinced it should
+  // skip Frame 2.  Frame 3 is emitted (to satisfy the second request) because a
+  // decision was made to skip over the no-show Frame 2.
+  testing_clock_->Advance(2 * time_advance_per_frame + target_playout_delay);
+  task_runner_->RunTasks();
+  EXPECT_EQ(2, frame_client_.number_times_called());
+
+  // Receive Frame 4 and expect it to fulfill the third request immediately.
+  rtp_header_.frame_id = kFirstFrameId + 3;  // "Frame 4"
+  rtp_header_.reference_frame_id = rtp_header_.frame_id;
+  rtp_header_.rtp_timestamp += rtp_advance_per_frame;
+  frame_client_.AddExpectedResult(
+      kFirstFrameId + 3, first_frame_capture_time + 3 * time_advance_per_frame +
+          target_playout_delay);
+  FeedOneFrameIntoReceiver();    // Frame 4
+  task_runner_->RunTasks();
+  EXPECT_EQ(3, frame_client_.number_times_called());
+
+  // Move forward to the playout time of an unreceived Frame 5.  Expect no
+  // additional frames were emitted.
+  testing_clock_->Advance(3 * time_advance_per_frame);
+  task_runner_->RunTasks();
+  EXPECT_EQ(3, frame_client_.number_times_called());
+
+  // Were only non-skipped frames logged?
+  std::vector<FrameEvent> frame_events;
+  event_subscriber.GetFrameEventsAndReset(&frame_events);
+  ASSERT_TRUE(!frame_events.empty());
+  for (size_t i = 0; i < frame_events.size(); ++i) {
+    EXPECT_EQ(FRAME_ACK_SENT, frame_events[i].type);
+    EXPECT_EQ(AUDIO_EVENT, frame_events[i].media_type);
+    EXPECT_LE(kFirstFrameId, frame_events[i].frame_id);
+    EXPECT_GE(kFirstFrameId + 4, frame_events[i].frame_id);
+    const int frame_offset = frame_events[i].frame_id - kFirstFrameId;
+    EXPECT_NE(frame_offset, 1);  // Frame 2 never received.
+    EXPECT_EQ(frame_offset * rtp_advance_per_frame,
+              frame_events[i].rtp_timestamp);
+  }
+  cast_environment_->Logging()->RemoveRawEventSubscriber(&event_subscriber);
+}
+
+TEST_F(FrameReceiverTest, ReceivesFramesRefusingToSkipAny) {
+  CreateFrameReceiverOfVideo();
+
+  SimpleEventSubscriber event_subscriber;
+  cast_environment_->Logging()->AddRawEventSubscriber(&event_subscriber);
+
+  EXPECT_CALL(mock_transport_, SendRtcpPacket(_, _))
+      .WillRepeatedly(testing::Return(true));
+
+  const uint32 rtp_advance_per_frame =
+      config_.frequency / config_.max_frame_rate;
+  const base::TimeDelta time_advance_per_frame =
+      base::TimeDelta::FromSeconds(1) / config_.max_frame_rate;
+
+  // Feed and process lip sync in receiver.
+  FeedLipSyncInfoIntoReceiver();
+  task_runner_->RunTasks();
+  const base::TimeTicks first_frame_capture_time = testing_clock_->NowTicks();
+
+  // Enqueue a request for a frame.
+  const ReceiveEncodedFrameCallback frame_encoded_callback =
+      base::Bind(&FakeFrameClient::DeliverEncodedFrame,
+                 base::Unretained(&frame_client_));
+  receiver_->RequestEncodedFrame(frame_encoded_callback);
+  task_runner_->RunTasks();
+  EXPECT_EQ(0, frame_client_.number_times_called());
+
+  // Receive one frame and expect to see the first request satisfied.
+  const base::TimeDelta target_playout_delay =
+      base::TimeDelta::FromMilliseconds(kPlayoutDelayMillis);
+  frame_client_.AddExpectedResult(
+      kFirstFrameId, first_frame_capture_time + target_playout_delay);
+  rtp_header_.rtp_timestamp = 0;
+  FeedOneFrameIntoReceiver();  // Frame 1
+  task_runner_->RunTasks();
+  EXPECT_EQ(1, frame_client_.number_times_called());
+
+  // Enqueue a second request for a frame, but it should not be fulfilled yet.
+  receiver_->RequestEncodedFrame(frame_encoded_callback);
+  task_runner_->RunTasks();
+  EXPECT_EQ(1, frame_client_.number_times_called());
+
+  // Receive one frame out-of-order: Make sure that we are not continuous and
+  // that the RTP timestamp represents a time in the future.
+  rtp_header_.is_key_frame = false;
+  rtp_header_.frame_id = kFirstFrameId + 2;  // "Frame 3"
+  rtp_header_.reference_frame_id = kFirstFrameId + 1;  // "Frame 2"
+  rtp_header_.rtp_timestamp += 2 * rtp_advance_per_frame;
+  FeedOneFrameIntoReceiver();  // Frame 3
+
+  // Frame 2 should not come out at this point in time.
+  task_runner_->RunTasks();
+  EXPECT_EQ(1, frame_client_.number_times_called());
+
+  // Enqueue a third request for a frame.
+  receiver_->RequestEncodedFrame(frame_encoded_callback);
+  task_runner_->RunTasks();
+  EXPECT_EQ(1, frame_client_.number_times_called());
+
+  // Now, advance time forward such that Frame 2 is now too late for playback.
+  // Regardless, the receiver must NOT emit Frame 3 yet because it is not
+  // allowed to skip frames when dependencies are not satisfied.  In other
+  // words, Frame 3 is not decodable without Frame 2.
+  testing_clock_->Advance(2 * time_advance_per_frame + target_playout_delay);
+  task_runner_->RunTasks();
+  EXPECT_EQ(1, frame_client_.number_times_called());
+
+  // Now receive Frame 2 and expect both the second and third requests to be
+  // fulfilled immediately.
+  frame_client_.AddExpectedResult(
+      kFirstFrameId + 1,  // "Frame 2"
+      first_frame_capture_time + 1 * time_advance_per_frame +
+          target_playout_delay);
+  frame_client_.AddExpectedResult(
+      kFirstFrameId + 2,  // "Frame 3"
+      first_frame_capture_time + 2 * time_advance_per_frame +
+          target_playout_delay);
+  --rtp_header_.frame_id;  // "Frame 2"
+  --rtp_header_.reference_frame_id;  // "Frame 1"
+  rtp_header_.rtp_timestamp -= rtp_advance_per_frame;
+  FeedOneFrameIntoReceiver();  // Frame 2
+  task_runner_->RunTasks();
+  EXPECT_EQ(3, frame_client_.number_times_called());
+
+  // Move forward to the playout time of an unreceived Frame 5.  Expect no
+  // additional frames were emitted.
+  testing_clock_->Advance(3 * time_advance_per_frame);
+  task_runner_->RunTasks();
+  EXPECT_EQ(3, frame_client_.number_times_called());
+
+  // Sanity-check logging results.
+  std::vector<FrameEvent> frame_events;
+  event_subscriber.GetFrameEventsAndReset(&frame_events);
+  ASSERT_TRUE(!frame_events.empty());
+  for (size_t i = 0; i < frame_events.size(); ++i) {
+    EXPECT_EQ(FRAME_ACK_SENT, frame_events[i].type);
+    EXPECT_EQ(VIDEO_EVENT, frame_events[i].media_type);
+    EXPECT_LE(kFirstFrameId, frame_events[i].frame_id);
+    EXPECT_GE(kFirstFrameId + 3, frame_events[i].frame_id);
+    const int frame_offset = frame_events[i].frame_id - kFirstFrameId;
+    EXPECT_EQ(frame_offset * rtp_advance_per_frame,
+              frame_events[i].rtp_timestamp);
+  }
+  cast_environment_->Logging()->RemoveRawEventSubscriber(&event_subscriber);
+}
+
+}  // namespace cast
+}  // namespace media
diff --git a/media/cast/video_receiver/video_decoder.cc b/media/cast/receiver/video_decoder.cc
index c2d2562..21f49d8 100644
--- a/media/cast/video_receiver/video_decoder.cc
+++ b/media/cast/receiver/video_decoder.cc
@@ -1,8 +1,8 @@
-// Copyright 2013 The Chromium Authors. All rights reserved.
+// Copyright 2014 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.
 
-#include "media/cast/video_receiver/video_decoder.h"
+#include "media/cast/receiver/video_decoder.h"
 
 #include "base/bind.h"
 #include "base/bind_helpers.h"
@@ -209,9 +209,9 @@ class VideoDecoder::FakeImpl : public VideoDecoder::ImplBase {
 
 VideoDecoder::VideoDecoder(
     const scoped_refptr<CastEnvironment>& cast_environment,
-    const FrameReceiverConfig& video_config)
+    transport::VideoCodec codec)
     : cast_environment_(cast_environment) {
-  switch (video_config.codec.video) {
+  switch (codec) {
 #ifndef OFFICIAL_BUILD
     case transport::kFakeSoftwareVideo:
       impl_ = new FakeImpl(cast_environment);
diff --git a/media/cast/video_receiver/video_decoder.h b/media/cast/receiver/video_decoder.h
index ea40173..66dc36b 100644
--- a/media/cast/video_receiver/video_decoder.h
+++ b/media/cast/receiver/video_decoder.h
@@ -1,9 +1,9 @@
-// Copyright 2013 The Chromium Authors. All rights reserved.
+// Copyright 2014 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.
 
-#ifndef MEDIA_CAST_VIDEO_RECEIVER_VIDEO_DECODER_H_
-#define MEDIA_CAST_VIDEO_RECEIVER_VIDEO_DECODER_H_
+#ifndef MEDIA_CAST_RECEIVER_VIDEO_DECODER_H_
+#define MEDIA_CAST_RECEIVER_VIDEO_DECODER_H_
 
 #include "base/callback.h"
 #include "base/memory/ref_counted.h"
@@ -28,7 +28,7 @@ class VideoDecoder {
                               bool is_continuous)> DecodeFrameCallback;
 
   VideoDecoder(const scoped_refptr<CastEnvironment>& cast_environment,
-               const FrameReceiverConfig& video_config);
+               transport::VideoCodec codec);
   virtual ~VideoDecoder();
 
   // Returns STATUS_VIDEO_INITIALIZED if the decoder was successfully
@@ -60,4 +60,4 @@ class VideoDecoder {
 }  // namespace cast
 }  // namespace media
 
-#endif  // MEDIA_CAST_VIDEO_RECEIVER_VIDEO_DECODER_H_
+#endif  // MEDIA_CAST_RECEIVER_VIDEO_DECODER_H_
diff --git a/media/cast/video_receiver/video_decoder_unittest.cc b/media/cast/receiver/video_decoder_unittest.cc
index 2ca28b0..1d16534 100644
--- a/media/cast/video_receiver/video_decoder_unittest.cc
+++ b/media/cast/receiver/video_decoder_unittest.cc
@@ -1,4 +1,4 @@
-// Copyright 2013 The Chromium Authors. All rights reserved.
+// Copyright 2014 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.
 
@@ -10,10 +10,9 @@
 #include "base/synchronization/lock.h"
 #include "base/time/time.h"
 #include "media/cast/cast_config.h"
-#include "media/cast/test/utility/default_config.h"
+#include "media/cast/receiver/video_decoder.h"
 #include "media/cast/test/utility/standalone_cast_environment.h"
 #include "media/cast/test/utility/video_utility.h"
-#include "media/cast/video_receiver/video_decoder.h"
 #include "media/cast/video_sender/codecs/vp8/vp8_encoder.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
@@ -48,9 +47,7 @@ class VideoDecoderTest
 
  protected:
   virtual void SetUp() OVERRIDE {
-    FrameReceiverConfig decoder_config = GetDefaultVideoReceiverConfig();
-    decoder_config.codec.video = GetParam();
-    video_decoder_.reset(new VideoDecoder(cast_environment_, decoder_config));
+    video_decoder_.reset(new VideoDecoder(cast_environment_, GetParam()));
     CHECK_EQ(STATUS_VIDEO_INITIALIZED, video_decoder_->InitializationResult());
 
     next_frame_timestamp_ = base::TimeDelta();
diff --git a/media/cast/rtcp/rtcp.cc b/media/cast/rtcp/rtcp.cc
index 3aa936b..480b2ac 100644
--- a/media/cast/rtcp/rtcp.cc
+++ b/media/cast/rtcp/rtcp.cc
@@ -79,7 +79,8 @@ Rtcp::Rtcp(scoped_refptr<CastEnvironment> cast_environment,
            transport::PacedPacketSender* paced_packet_sender,
            RtpReceiverStatistics* rtp_receiver_statistics, RtcpMode rtcp_mode,
            const base::TimeDelta& rtcp_interval, uint32 local_ssrc,
-           uint32 remote_ssrc, const std::string& c_name, bool is_audio)
+           uint32 remote_ssrc, const std::string& c_name,
+           EventMediaType event_media_type)
     : cast_environment_(cast_environment),
       transport_sender_(transport_sender),
       rtcp_interval_(rtcp_interval),
@@ -87,6 +88,7 @@ Rtcp::Rtcp(scoped_refptr<CastEnvironment> cast_environment,
       local_ssrc_(local_ssrc),
       remote_ssrc_(remote_ssrc),
       c_name_(c_name),
+      event_media_type_(event_media_type),
       rtp_receiver_statistics_(rtp_receiver_statistics),
       rtt_feedback_(new LocalRtcpRttFeedback(this)),
       receiver_feedback_(new LocalRtcpReceiverFeedback(this, cast_environment)),
@@ -97,8 +99,7 @@ Rtcp::Rtcp(scoped_refptr<CastEnvironment> cast_environment,
       lip_sync_rtp_timestamp_(0),
       lip_sync_ntp_timestamp_(0),
       min_rtt_(base::TimeDelta::FromMilliseconds(kMaxRttMs)),
-      number_of_rtt_in_avg_(0),
-      is_audio_(is_audio) {
+      number_of_rtt_in_avg_(0) {
   rtcp_receiver_.reset(new RtcpReceiver(cast_environment, sender_feedback,
                                         receiver_feedback_.get(),
                                         rtt_feedback_.get(), local_ssrc));
@@ -394,7 +395,6 @@ void Rtcp::UpdateNextTimeToSendRtcp() {
 void Rtcp::OnReceivedReceiverLog(const RtcpReceiverLogMessage& receiver_log) {
   // Add received log messages into our log system.
   RtcpReceiverLogMessage::const_iterator it = receiver_log.begin();
-  EventMediaType media_type = is_audio_ ? AUDIO_EVENT : VIDEO_EVENT;
   for (; it != receiver_log.end(); ++it) {
     uint32 rtp_timestamp = it->rtp_timestamp_;
 
@@ -405,18 +405,18 @@ void Rtcp::OnReceivedReceiverLog(const RtcpReceiverLogMessage& receiver_log) {
         case PACKET_RECEIVED:
           cast_environment_->Logging()->InsertPacketEvent(
               event_it->event_timestamp, event_it->type,
-              media_type, rtp_timestamp,
+              event_media_type_, rtp_timestamp,
               kFrameIdUnknown, event_it->packet_id, 0, 0);
           break;
         case FRAME_ACK_SENT:
         case FRAME_DECODED:
           cast_environment_->Logging()->InsertFrameEvent(
-              event_it->event_timestamp, event_it->type, media_type,
+              event_it->event_timestamp, event_it->type, event_media_type_,
               rtp_timestamp, kFrameIdUnknown);
           break;
         case FRAME_PLAYOUT:
           cast_environment_->Logging()->InsertFrameEventWithDelay(
-              event_it->event_timestamp, event_it->type, media_type,
+              event_it->event_timestamp, event_it->type, event_media_type_,
               rtp_timestamp, kFrameIdUnknown, event_it->delay_delta);
           break;
         default:
diff --git a/media/cast/rtcp/rtcp.h b/media/cast/rtcp/rtcp.h
index ff81bb9..9d0184f 100644
--- a/media/cast/rtcp/rtcp.h
+++ b/media/cast/rtcp/rtcp.h
@@ -68,7 +68,7 @@ class Rtcp {
        uint32 local_ssrc,
        uint32 remote_ssrc,
        const std::string& c_name,
-       bool is_audio);
+       EventMediaType event_media_type);
 
   virtual ~Rtcp();
 
@@ -156,6 +156,7 @@ class Rtcp {
   const uint32 local_ssrc_;
   const uint32 remote_ssrc_;
   const std::string c_name_;
+  const EventMediaType event_media_type_;
 
   // Not owned by this class.
   RtpReceiverStatistics* const rtp_receiver_statistics_;
@@ -195,7 +196,6 @@ class Rtcp {
   int number_of_rtt_in_avg_;
   double avg_rtt_ms_;
   uint16 target_delay_ms_;
-  bool is_audio_;
 
   DISALLOW_COPY_AND_ASSIGN(Rtcp);
 };
diff --git a/media/cast/rtcp/rtcp_unittest.cc b/media/cast/rtcp/rtcp_unittest.cc
index 3cae9b9..d5bf312 100644
--- a/media/cast/rtcp/rtcp_unittest.cc
+++ b/media/cast/rtcp/rtcp_unittest.cc
@@ -144,7 +144,7 @@ class RtcpPeer : public Rtcp {
              local_ssrc,
              remote_ssrc,
              c_name,
-             true) {}
+             AUDIO_EVENT) {}
 
   using Rtcp::OnReceivedNtp;
   using Rtcp::OnReceivedLipSyncInfo;
@@ -219,7 +219,7 @@ TEST_F(RtcpTest, TimeToSend) {
             kSenderSsrc,
             kReceiverSsrc,
             kCName,
-            true);
+            AUDIO_EVENT);
   receiver_to_sender_.set_rtcp_receiver(&rtcp);
   EXPECT_LE(start_time, rtcp.TimeToSendNextRtcpReport());
   EXPECT_GE(
@@ -241,7 +241,7 @@ TEST_F(RtcpTest, BasicSenderReport) {
             kSenderSsrc,
             kReceiverSsrc,
             kCName,
-            true);
+            AUDIO_EVENT);
   sender_to_receiver_.set_rtcp_receiver(&rtcp);
   rtcp.SendRtcpFromRtpSender(base::TimeTicks(), 0);
 }
@@ -257,7 +257,7 @@ TEST_F(RtcpTest, BasicReceiverReport) {
             kSenderSsrc,
             kReceiverSsrc,
             kCName,
-            true);
+            AUDIO_EVENT);
   receiver_to_sender_.set_rtcp_receiver(&rtcp);
   rtcp.SendRtcpFromRtpReceiver(NULL, NULL);
 }
@@ -276,7 +276,7 @@ TEST_F(RtcpTest, BasicCast) {
             kSenderSsrc,
             kSenderSsrc,
             kCName,
-            true);
+            AUDIO_EVENT);
   receiver_to_sender_.set_rtcp_receiver(&rtcp);
   RtcpCastMessage cast_message(kSenderSsrc);
   cast_message.ack_frame_id_ = kAckFrameId;
@@ -303,7 +303,7 @@ TEST_F(RtcpTest, RttReducedSizeRtcp) {
                      kReceiverSsrc,
                      kSenderSsrc,
                      kCName,
-                     true);
+                     AUDIO_EVENT);
 
   // Media sender.
   Rtcp rtcp_sender(cast_environment_,
@@ -316,7 +316,7 @@ TEST_F(RtcpTest, RttReducedSizeRtcp) {
                    kSenderSsrc,
                    kReceiverSsrc,
                    kCName,
-                   true);
+                   AUDIO_EVENT);
 
   sender_to_receiver_.set_rtcp_receiver(&rtcp_receiver);
   receiver_to_sender_.set_rtcp_receiver(&rtcp_sender);
@@ -359,7 +359,7 @@ TEST_F(RtcpTest, Rtt) {
                      kReceiverSsrc,
                      kSenderSsrc,
                      kCName,
-                     true);
+                     AUDIO_EVENT);
 
   // Media sender.
   Rtcp rtcp_sender(cast_environment_,
@@ -372,7 +372,7 @@ TEST_F(RtcpTest, Rtt) {
                    kSenderSsrc,
                    kReceiverSsrc,
                    kCName,
-                   true);
+                   AUDIO_EVENT);
 
   receiver_to_sender_.set_rtcp_receiver(&rtcp_sender);
   sender_to_receiver_.set_rtcp_receiver(&rtcp_receiver);
@@ -452,7 +452,7 @@ TEST_F(RtcpTest, RttWithPacketLoss) {
                      kReceiverSsrc,
                      kSenderSsrc,
                      kCName,
-                     true);
+                     AUDIO_EVENT);
 
   // Media sender.
   Rtcp rtcp_sender(cast_environment_,
@@ -465,7 +465,7 @@ TEST_F(RtcpTest, RttWithPacketLoss) {
                    kSenderSsrc,
                    kReceiverSsrc,
                    kCName,
-                   true);
+                   AUDIO_EVENT);
 
   receiver_to_sender_.set_rtcp_receiver(&rtcp_sender);
   sender_to_receiver_.set_rtcp_receiver(&rtcp_receiver);
diff --git a/media/cast/rtp_receiver/rtp_parser/rtp_parser.cc b/media/cast/rtp_receiver/rtp_parser/rtp_parser.cc
index d75696a..f44e82d 100644
--- a/media/cast/rtp_receiver/rtp_parser/rtp_parser.cc
+++ b/media/cast/rtp_receiver/rtp_parser/rtp_parser.cc
@@ -7,7 +7,6 @@
 #include "base/big_endian.h"
 #include "base/logging.h"
 #include "media/cast/cast_defines.h"
-#include "media/cast/rtp_receiver/rtp_receiver.h"
 
 namespace media {
 namespace cast {
diff --git a/media/cast/rtp_receiver/rtp_parser/rtp_parser_unittest.cc b/media/cast/rtp_receiver/rtp_parser/rtp_parser_unittest.cc
index 4e3560a9..47c7913 100644
--- a/media/cast/rtp_receiver/rtp_parser/rtp_parser_unittest.cc
+++ b/media/cast/rtp_receiver/rtp_parser/rtp_parser_unittest.cc
@@ -6,7 +6,6 @@
 #include "base/rand_util.h"
 #include "media/cast/rtp_receiver/rtp_parser/rtp_parser.h"
 #include "media/cast/rtp_receiver/rtp_parser/test/rtp_packet_builder.h"
-#include "media/cast/rtp_receiver/rtp_receiver.h"
 #include "media/cast/rtp_receiver/rtp_receiver_defines.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
diff --git a/media/cast/rtp_receiver/rtp_receiver.cc b/media/cast/rtp_receiver/rtp_receiver.cc
deleted file mode 100644
index f7ff50b..0000000
--- a/media/cast/rtp_receiver/rtp_receiver.cc
+++ /dev/null
@@ -1,55 +0,0 @@
-// Copyright 2013 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.
-
-#include "media/cast/rtp_receiver/rtp_receiver.h"
-
-#include "base/big_endian.h"
-#include "base/logging.h"
-#include "media/cast/rtp_receiver/receiver_stats.h"
-#include "media/cast/rtp_receiver/rtp_parser/rtp_parser.h"
-#include "media/cast/rtp_receiver/rtp_receiver_defines.h"
-
-namespace media {
-namespace cast {
-
-RtpReceiver::RtpReceiver(base::TickClock* clock,
-                         const FrameReceiverConfig* audio_config,
-                         const FrameReceiverConfig* video_config) :
-    packet_parser_(audio_config ? audio_config->incoming_ssrc :
-                   (video_config ? video_config->incoming_ssrc : 0),
-                   audio_config ? audio_config->rtp_payload_type :
-                   (video_config ? video_config->rtp_payload_type : 0)),
-    stats_(clock) {
-  DCHECK(audio_config || video_config);
-}
-
-RtpReceiver::~RtpReceiver() {}
-
-// static
-uint32 RtpReceiver::GetSsrcOfSender(const uint8* rtcp_buffer, size_t length) {
-  DCHECK_GE(length, kMinLengthOfRtp) << "Invalid RTP packet";
-  uint32 ssrc_of_sender;
-  base::BigEndianReader big_endian_reader(
-      reinterpret_cast<const char*>(rtcp_buffer), length);
-  big_endian_reader.Skip(8);  // Skip header
-  big_endian_reader.ReadU32(&ssrc_of_sender);
-  return ssrc_of_sender;
-}
-
-bool RtpReceiver::ReceivedPacket(const uint8* packet, size_t length) {
-  RtpCastHeader rtp_header;
-  const uint8* payload_data;
-  size_t payload_size;
-  if (!packet_parser_.ParsePacket(
-          packet, length, &rtp_header, &payload_data, &payload_size)) {
-    return false;
-  }
-
-  OnReceivedPayloadData(payload_data, payload_size, rtp_header);
-  stats_.UpdateStatistics(rtp_header);
-  return true;
-}
-
-}  // namespace cast
-}  // namespace media
diff --git a/media/cast/rtp_receiver/rtp_receiver.h b/media/cast/rtp_receiver/rtp_receiver.h
deleted file mode 100644
index 35c7c93..0000000
--- a/media/cast/rtp_receiver/rtp_receiver.h
+++ /dev/null
@@ -1,53 +0,0 @@
-// Copyright 2013 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.
-
-// Interface to the rtp receiver.
-
-#ifndef MEDIA_CAST_RTP_RECEIVER_RTP_RECEIVER_H_
-#define MEDIA_CAST_RTP_RECEIVER_RTP_RECEIVER_H_
-
-#include "base/memory/scoped_ptr.h"
-#include "media/cast/cast_config.h"
-#include "media/cast/rtcp/rtcp.h"
-#include "media/cast/rtp_receiver/receiver_stats.h"
-#include "media/cast/rtp_receiver/rtp_parser/rtp_parser.h"
-#include "media/cast/rtp_receiver/rtp_receiver_defines.h"
-
-namespace media {
-namespace cast {
-
-// TODO(miu): This is a good candidate to contain common functionality that's
-// identical in both AudioReceiver and VideoReceiver.
-class RtpReceiver {
- public:
-  RtpReceiver(base::TickClock* clock,
-              const FrameReceiverConfig* audio_config,
-              const FrameReceiverConfig* video_config);
-  virtual ~RtpReceiver();
-
-  static uint32 GetSsrcOfSender(const uint8* rtcp_buffer, size_t length);
-
-  bool ReceivedPacket(const uint8* packet, size_t length);
-
-  RtpReceiverStatistics* GetStatistics() {
-    return &stats_;
-  }
-
- protected:
-  // Subclasses implement this to consume and process deserialized packets.
-  virtual void OnReceivedPayloadData(const uint8* payload_data,
-                                     size_t payload_size,
-                                     const RtpCastHeader& rtp_header) = 0;
-
- private:
-  RtpParser packet_parser_;
-  ReceiverStats stats_;
-
-  DISALLOW_COPY_AND_ASSIGN(RtpReceiver);
-};
-
-}  // namespace cast
-}  // namespace media
-
-#endif  // MEDIA_CAST_RTP_RECEIVER_RTP_RECEIVER_H_
diff --git a/media/cast/test/end2end_unittest.cc b/media/cast/test/end2end_unittest.cc
index cfba81d..4a0d820 100644
--- a/media/cast/test/end2end_unittest.cc
+++ b/media/cast/test/end2end_unittest.cc
@@ -566,7 +566,7 @@ class End2EndTest : public ::testing::Test {
 
   void RequestAudioFrames(int count, bool with_check) {
     for (int i = 0; i < count; ++i) {
-      frame_receiver_->GetRawAudioFrame(
+      cast_receiver_->RequestDecodedAudioFrame(
           base::Bind(with_check ? &TestReceiverAudioCallback::CheckAudioFrame :
                                   &TestReceiverAudioCallback::IgnoreAudioFrame,
                      test_receiver_audio_callback_));
@@ -611,8 +611,6 @@ class End2EndTest : public ::testing::Test {
     audio_frame_input_ = cast_sender_->audio_frame_input();
     video_frame_input_ = cast_sender_->video_frame_input();
 
-    frame_receiver_ = cast_receiver_->frame_receiver();
-
     audio_bus_factory_.reset(
         new TestAudioBusFactory(audio_sender_config_.channels,
                                 audio_sender_config_.frequency,
@@ -684,7 +682,7 @@ class End2EndTest : public ::testing::Test {
     video_ticks_.push_back(std::make_pair(
         testing_clock_receiver_->NowTicks(),
         playout_time));
-    frame_receiver_->GetRawVideoFrame(
+    cast_receiver_->RequestDecodedVideoFrame(
         base::Bind(&End2EndTest::BasicPlayerGotVideoFrame,
                    base::Unretained(this)));
   }
@@ -701,16 +699,16 @@ class End2EndTest : public ::testing::Test {
     audio_ticks_.push_back(std::make_pair(
         testing_clock_receiver_->NowTicks(),
         playout_time));
-    frame_receiver_->GetRawAudioFrame(
+    cast_receiver_->RequestDecodedAudioFrame(
         base::Bind(&End2EndTest::BasicPlayerGotAudioFrame,
                    base::Unretained(this)));
   }
 
   void StartBasicPlayer() {
-    frame_receiver_->GetRawVideoFrame(
+    cast_receiver_->RequestDecodedVideoFrame(
         base::Bind(&End2EndTest::BasicPlayerGotVideoFrame,
                    base::Unretained(this)));
-    frame_receiver_->GetRawAudioFrame(
+    cast_receiver_->RequestDecodedAudioFrame(
         base::Bind(&End2EndTest::BasicPlayerGotAudioFrame,
                    base::Unretained(this)));
   }
@@ -768,7 +766,6 @@ class End2EndTest : public ::testing::Test {
   scoped_ptr<CastSender> cast_sender_;
   scoped_refptr<AudioFrameInput> audio_frame_input_;
   scoped_refptr<VideoFrameInput> video_frame_input_;
-  scoped_refptr<FrameReceiver> frame_receiver_;
 
   scoped_refptr<TestReceiverAudioCallback> test_receiver_audio_callback_;
   scoped_refptr<TestReceiverVideoCallback> test_receiver_video_callback_;
@@ -820,7 +817,7 @@ TEST_F(End2EndTest, LoopNoLossPcm16) {
     RequestAudioFrames(num_audio_frames, true);
     num_audio_frames_requested += num_audio_frames;
 
-    frame_receiver_->GetRawVideoFrame(
+    cast_receiver_->RequestDecodedVideoFrame(
         base::Bind(&TestReceiverVideoCallback::CheckVideoFrame,
                    test_receiver_video_callback_));
 
@@ -846,7 +843,7 @@ TEST_F(End2EndTest, LoopNoLossPcm16ExternalDecoder) {
   for (int i = 0; i < kNumIterations; ++i) {
     FeedAudioFrames(1, true);
     RunTasks(kAudioFrameDurationMs);
-    frame_receiver_->GetCodedAudioFrame(
+    cast_receiver_->RequestEncodedAudioFrame(
         base::Bind(&TestReceiverAudioCallback::CheckCodedAudioFrame,
                    test_receiver_audio_callback_));
   }
@@ -954,7 +951,7 @@ TEST_F(End2EndTest, DISABLED_StartSenderBeforeReceiver) {
     RequestAudioFrames(num_audio_frames, true);
     num_audio_frames_requested += num_audio_frames;
 
-    frame_receiver_->GetRawVideoFrame(
+    cast_receiver_->RequestDecodedVideoFrame(
         base::Bind(&TestReceiverVideoCallback::CheckVideoFrame,
                    test_receiver_video_callback_));
 
@@ -990,7 +987,7 @@ TEST_F(End2EndTest, DISABLED_GlitchWith3Buffers) {
         video_sender_config_.height,
         capture_time + base::TimeDelta::FromMilliseconds(kTargetPlayoutDelayMs),
         true);
-    frame_receiver_->GetRawVideoFrame(
+    cast_receiver_->RequestDecodedVideoFrame(
         base::Bind(&TestReceiverVideoCallback::CheckVideoFrame,
                    test_receiver_video_callback_));
     RunTasks(kFrameTimerMs);
@@ -1023,7 +1020,7 @@ TEST_F(End2EndTest, DISABLED_GlitchWith3Buffers) {
       capture_time + base::TimeDelta::FromMilliseconds(kTargetPlayoutDelayMs),
       true);
 
-  frame_receiver_->GetRawVideoFrame(
+  cast_receiver_->RequestDecodedVideoFrame(
       base::Bind(&TestReceiverVideoCallback::CheckVideoFrame,
                  test_receiver_video_callback_));
 
@@ -1059,7 +1056,7 @@ TEST_F(End2EndTest, DISABLED_DropEveryOtherFrame3Buffers) {
 
       // GetRawVideoFrame will not return the frame until we are close in
       // time before we should render the frame.
-      frame_receiver_->GetRawVideoFrame(
+      cast_receiver_->RequestDecodedVideoFrame(
           base::Bind(&TestReceiverVideoCallback::CheckVideoFrame,
                      test_receiver_video_callback_));
     }
@@ -1100,7 +1097,7 @@ TEST_F(End2EndTest, CryptoVideo) {
 
     RunTasks(kFrameTimerMs);
 
-    frame_receiver_->GetRawVideoFrame(
+    cast_receiver_->RequestDecodedVideoFrame(
         base::Bind(&TestReceiverVideoCallback::CheckVideoFrame,
                    test_receiver_video_callback_));
   }
@@ -1156,7 +1153,7 @@ TEST_F(End2EndTest, VideoLogging) {
     SendVideoFrame(video_start, capture_time);
     RunTasks(kFrameTimerMs);
 
-    frame_receiver_->GetRawVideoFrame(
+    cast_receiver_->RequestDecodedVideoFrame(
         base::Bind(&TestReceiverVideoCallback::CheckVideoFrame,
                    test_receiver_video_callback_));
 
diff --git a/media/cast/test/utility/in_process_receiver.cc b/media/cast/test/utility/in_process_receiver.cc
index ba190ce..cfcc1fc 100644
--- a/media/cast/test/utility/in_process_receiver.cc
+++ b/media/cast/test/utility/in_process_receiver.cc
@@ -114,14 +114,14 @@ void InProcessReceiver::GotVideoFrame(
 
 void InProcessReceiver::PullNextAudioFrame() {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
-  cast_receiver_->frame_receiver()->GetRawAudioFrame(
+  cast_receiver_->RequestDecodedAudioFrame(
       base::Bind(&InProcessReceiver::GotAudioFrame,
                  weak_factory_.GetWeakPtr()));
 }
 
 void InProcessReceiver::PullNextVideoFrame() {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
-  cast_receiver_->frame_receiver()->GetRawVideoFrame(base::Bind(
+  cast_receiver_->RequestDecodedVideoFrame(base::Bind(
       &InProcessReceiver::GotVideoFrame, weak_factory_.GetWeakPtr()));
 }
 
diff --git a/media/cast/video_receiver/video_receiver.cc b/media/cast/video_receiver/video_receiver.cc
deleted file mode 100644
index d9000de..0000000
--- a/media/cast/video_receiver/video_receiver.cc
+++ /dev/null
@@ -1,380 +0,0 @@
-// Copyright 2013 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.
-
-#include "media/cast/video_receiver/video_receiver.h"
-
-#include <algorithm>
-
-#include "base/bind.h"
-#include "base/debug/trace_event.h"
-#include "base/logging.h"
-#include "base/message_loop/message_loop.h"
-#include "media/base/video_frame.h"
-#include "media/cast/logging/logging_defines.h"
-#include "media/cast/transport/cast_transport_defines.h"
-#include "media/cast/video_receiver/video_decoder.h"
-
-namespace {
-const int kMinSchedulingDelayMs = 1;
-}  // namespace
-
-namespace media {
-namespace cast {
-
-VideoReceiver::VideoReceiver(scoped_refptr<CastEnvironment> cast_environment,
-                             const FrameReceiverConfig& video_config,
-                             transport::PacedPacketSender* const packet_sender)
-    : RtpReceiver(cast_environment->Clock(), NULL, &video_config),
-      cast_environment_(cast_environment),
-      event_subscriber_(kReceiverRtcpEventHistorySize, VIDEO_EVENT),
-      codec_(video_config.codec.video),
-      target_playout_delay_(
-          base::TimeDelta::FromMilliseconds(video_config.rtp_max_delay_ms)),
-      expected_frame_duration_(
-          base::TimeDelta::FromSeconds(1) / video_config.max_frame_rate),
-      reports_are_scheduled_(false),
-      framer_(cast_environment->Clock(),
-              this,
-              video_config.incoming_ssrc,
-              true,
-              video_config.rtp_max_delay_ms * video_config.max_frame_rate /
-                  1000),
-      rtcp_(cast_environment_,
-            NULL,
-            NULL,
-            packet_sender,
-            GetStatistics(),
-            video_config.rtcp_mode,
-            base::TimeDelta::FromMilliseconds(video_config.rtcp_interval),
-            video_config.feedback_ssrc,
-            video_config.incoming_ssrc,
-            video_config.rtcp_c_name,
-            false),
-      is_waiting_for_consecutive_frame_(false),
-      lip_sync_drift_(ClockDriftSmoother::GetDefaultTimeConstant()),
-      weak_factory_(this) {
-  DCHECK_GT(video_config.rtp_max_delay_ms, 0);
-  DCHECK_GT(video_config.max_frame_rate, 0);
-  video_decoder_.reset(new VideoDecoder(cast_environment, video_config));
-  decryptor_.Initialize(video_config.aes_key, video_config.aes_iv_mask);
-  rtcp_.SetTargetDelay(target_playout_delay_);
-  cast_environment_->Logging()->AddRawEventSubscriber(&event_subscriber_);
-  memset(frame_id_to_rtp_timestamp_, 0, sizeof(frame_id_to_rtp_timestamp_));
-}
-
-VideoReceiver::~VideoReceiver() {
-  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
-  cast_environment_->Logging()->RemoveRawEventSubscriber(&event_subscriber_);
-}
-
-void VideoReceiver::GetRawVideoFrame(
-    const VideoFrameDecodedCallback& callback) {
-  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
-  DCHECK(!callback.is_null());
-  DCHECK(video_decoder_.get());
-  GetEncodedVideoFrame(base::Bind(
-      &VideoReceiver::DecodeEncodedVideoFrame,
-      // Note: Use of Unretained is safe since this Closure is guaranteed to be
-      // invoked before destruction of |this|.
-      base::Unretained(this),
-      callback));
-}
-
-void VideoReceiver::DecodeEncodedVideoFrame(
-    const VideoFrameDecodedCallback& callback,
-    scoped_ptr<transport::EncodedFrame> encoded_frame) {
-  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
-  if (!encoded_frame) {
-    callback.Run(
-        make_scoped_refptr<VideoFrame>(NULL), base::TimeTicks(), false);
-    return;
-  }
-  const uint32 frame_id = encoded_frame->frame_id;
-  const uint32 rtp_timestamp = encoded_frame->rtp_timestamp;
-  const base::TimeTicks playout_time = encoded_frame->reference_time;
-  video_decoder_->DecodeFrame(encoded_frame.Pass(),
-                              base::Bind(&VideoReceiver::EmitRawVideoFrame,
-                                         cast_environment_,
-                                         callback,
-                                         frame_id,
-                                         rtp_timestamp,
-                                         playout_time));
-}
-
-// static
-void VideoReceiver::EmitRawVideoFrame(
-    const scoped_refptr<CastEnvironment>& cast_environment,
-    const VideoFrameDecodedCallback& callback,
-    uint32 frame_id,
-    uint32 rtp_timestamp,
-    const base::TimeTicks& playout_time,
-    const scoped_refptr<VideoFrame>& video_frame,
-    bool is_continuous) {
-  DCHECK(cast_environment->CurrentlyOn(CastEnvironment::MAIN));
-  if (video_frame) {
-    const base::TimeTicks now = cast_environment->Clock()->NowTicks();
-    cast_environment->Logging()->InsertFrameEvent(
-        now, FRAME_DECODED, VIDEO_EVENT, rtp_timestamp, frame_id);
-    cast_environment->Logging()->InsertFrameEventWithDelay(
-        now, FRAME_PLAYOUT, VIDEO_EVENT, rtp_timestamp, frame_id,
-        playout_time - now);
-    // Used by chrome/browser/extension/api/cast_streaming/performance_test.cc
-    TRACE_EVENT_INSTANT1(
-        "cast_perf_test", "FrameDecoded",
-        TRACE_EVENT_SCOPE_THREAD,
-        "rtp_timestamp", rtp_timestamp);
-  }
-  callback.Run(video_frame, playout_time, is_continuous);
-}
-
-void VideoReceiver::GetEncodedVideoFrame(const FrameEncodedCallback& callback) {
-  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
-  frame_request_queue_.push_back(callback);
-  EmitAvailableEncodedFrames();
-}
-
-void VideoReceiver::EmitAvailableEncodedFrames() {
-  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
-
-  while (!frame_request_queue_.empty()) {
-    // Attempt to peek at the next completed frame from the |framer_|.
-    // TODO(miu): We should only be peeking at the metadata, and not copying the
-    // payload yet!  Or, at least, peek using a StringPiece instead of a copy.
-    scoped_ptr<transport::EncodedFrame> encoded_frame(
-        new transport::EncodedFrame());
-    bool is_consecutively_next_frame = false;
-    bool have_multiple_complete_frames = false;
-
-    if (!framer_.GetEncodedFrame(encoded_frame.get(),
-                                 &is_consecutively_next_frame,
-                                 &have_multiple_complete_frames)) {
-      VLOG(1) << "Wait for more video packets to produce a completed frame.";
-      return;  // OnReceivedPayloadData() will invoke this method in the future.
-    }
-
-    const base::TimeTicks now = cast_environment_->Clock()->NowTicks();
-    const base::TimeTicks playout_time =
-        GetPlayoutTime(encoded_frame->rtp_timestamp);
-
-    // If we have multiple decodable frames, and the current frame is
-    // too old, then skip it and decode the next frame instead.
-    if (have_multiple_complete_frames && now > playout_time) {
-      framer_.ReleaseFrame(encoded_frame->frame_id);
-      continue;
-    }
-
-    // If |framer_| has a frame ready that is out of sequence, examine the
-    // playout time to determine whether it's acceptable to continue, thereby
-    // skipping one or more frames.  Skip if the missing frame wouldn't complete
-    // playing before the start of playback of the available frame.
-    if (!is_consecutively_next_frame) {
-      // TODO(miu): Also account for expected decode time here?
-      const base::TimeTicks earliest_possible_end_time_of_missing_frame =
-          now + expected_frame_duration_;
-      if (earliest_possible_end_time_of_missing_frame < playout_time) {
-        VLOG(1) << "Wait for next consecutive frame instead of skipping.";
-        if (!is_waiting_for_consecutive_frame_) {
-          is_waiting_for_consecutive_frame_ = true;
-          cast_environment_->PostDelayedTask(
-              CastEnvironment::MAIN,
-              FROM_HERE,
-              base::Bind(&VideoReceiver::EmitAvailableEncodedFramesAfterWaiting,
-                         weak_factory_.GetWeakPtr()),
-              playout_time - now);
-        }
-        return;
-      }
-    }
-
-    // Decrypt the payload data in the frame, if crypto is being used.
-    if (decryptor_.initialized()) {
-      std::string decrypted_video_data;
-      if (!decryptor_.Decrypt(encoded_frame->frame_id,
-                              encoded_frame->data,
-                              &decrypted_video_data)) {
-        // Decryption failed.  Give up on this frame, releasing it from the
-        // jitter buffer.
-        framer_.ReleaseFrame(encoded_frame->frame_id);
-        continue;
-      }
-      encoded_frame->data.swap(decrypted_video_data);
-    }
-
-    // At this point, we have a decrypted EncodedFrame ready to be emitted.
-    encoded_frame->reference_time = playout_time;
-    framer_.ReleaseFrame(encoded_frame->frame_id);
-    // Used by chrome/browser/extension/api/cast_streaming/performance_test.cc
-    TRACE_EVENT_INSTANT2(
-        "cast_perf_test", "PullEncodedVideoFrame",
-        TRACE_EVENT_SCOPE_THREAD,
-        "rtp_timestamp", encoded_frame->rtp_timestamp,
-        // TODO(miu): Need to find an alternative to using ToInternalValue():
-        "render_time", playout_time.ToInternalValue());
-    cast_environment_->PostTask(CastEnvironment::MAIN,
-                                FROM_HERE,
-                                base::Bind(frame_request_queue_.front(),
-                                           base::Passed(&encoded_frame)));
-    frame_request_queue_.pop_front();
-  }
-}
-
-void VideoReceiver::EmitAvailableEncodedFramesAfterWaiting() {
-  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
-  DCHECK(is_waiting_for_consecutive_frame_);
-  is_waiting_for_consecutive_frame_ = false;
-  EmitAvailableEncodedFrames();
-}
-
-base::TimeTicks VideoReceiver::GetPlayoutTime(uint32 rtp_timestamp) const {
-  return lip_sync_reference_time_ +
-      lip_sync_drift_.Current() +
-      RtpDeltaToTimeDelta(
-          static_cast<int32>(rtp_timestamp - lip_sync_rtp_timestamp_),
-          kVideoFrequency) +
-      target_playout_delay_;
-}
-
-void VideoReceiver::IncomingPacket(scoped_ptr<Packet> packet) {
-  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
-  if (Rtcp::IsRtcpPacket(&packet->front(), packet->size())) {
-    rtcp_.IncomingRtcpPacket(&packet->front(), packet->size());
-  } else {
-    ReceivedPacket(&packet->front(), packet->size());
-  }
-  if (!reports_are_scheduled_) {
-    ScheduleNextRtcpReport();
-    ScheduleNextCastMessage();
-    reports_are_scheduled_ = true;
-  }
-}
-
-void VideoReceiver::OnReceivedPayloadData(const uint8* payload_data,
-                                          size_t payload_size,
-                                          const RtpCastHeader& rtp_header) {
-  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
-
-  const base::TimeTicks now = cast_environment_->Clock()->NowTicks();
-
-  frame_id_to_rtp_timestamp_[rtp_header.frame_id & 0xff] =
-      rtp_header.rtp_timestamp;
-  cast_environment_->Logging()->InsertPacketEvent(
-      now,
-      PACKET_RECEIVED,
-      VIDEO_EVENT,
-      rtp_header.rtp_timestamp,
-      rtp_header.frame_id,
-      rtp_header.packet_id,
-      rtp_header.max_packet_id,
-      payload_size);
-
-  bool duplicate = false;
-  const bool complete =
-      framer_.InsertPacket(payload_data, payload_size, rtp_header, &duplicate);
-
-  // Duplicate packets are ignored.
-  if (duplicate)
-    return;
-
-  // Update lip-sync values upon receiving the first packet of each frame, or if
-  // they have never been set yet.
-  if (rtp_header.packet_id == 0 || lip_sync_reference_time_.is_null()) {
-    RtpTimestamp fresh_sync_rtp;
-    base::TimeTicks fresh_sync_reference;
-    if (!rtcp_.GetLatestLipSyncTimes(&fresh_sync_rtp, &fresh_sync_reference)) {
-      // HACK: The sender should have provided Sender Reports before the first
-      // frame was sent.  However, the spec does not currently require this.
-      // Therefore, when the data is missing, the local clock is used to
-      // generate reference timestamps.
-      VLOG(2) << "Lip sync info missing.  Falling-back to local clock.";
-      fresh_sync_rtp = rtp_header.rtp_timestamp;
-      fresh_sync_reference = now;
-    }
-    // |lip_sync_reference_time_| is always incremented according to the time
-    // delta computed from the difference in RTP timestamps.  Then,
-    // |lip_sync_drift_| accounts for clock drift and also smoothes-out any
-    // sudden/discontinuous shifts in the series of reference time values.
-    if (lip_sync_reference_time_.is_null()) {
-      lip_sync_reference_time_ = fresh_sync_reference;
-    } else {
-      lip_sync_reference_time_ += RtpDeltaToTimeDelta(
-          static_cast<int32>(fresh_sync_rtp - lip_sync_rtp_timestamp_),
-          kVideoFrequency);
-    }
-    lip_sync_rtp_timestamp_ = fresh_sync_rtp;
-    lip_sync_drift_.Update(
-        now, fresh_sync_reference - lip_sync_reference_time_);
-  }
-
-  // Video frame not complete; wait for more packets.
-  if (!complete)
-    return;
-
-  EmitAvailableEncodedFrames();
-}
-
-// Send a cast feedback message. Actual message created in the framer (cast
-// message builder).
-void VideoReceiver::CastFeedback(const RtcpCastMessage& cast_message) {
-  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
-
-  base::TimeTicks now = cast_environment_->Clock()->NowTicks();
-  RtpTimestamp rtp_timestamp =
-      frame_id_to_rtp_timestamp_[cast_message.ack_frame_id_ & 0xff];
-  cast_environment_->Logging()->InsertFrameEvent(
-      now, FRAME_ACK_SENT, VIDEO_EVENT,
-      rtp_timestamp, cast_message.ack_frame_id_);
-
-  ReceiverRtcpEventSubscriber::RtcpEventMultiMap rtcp_events;
-  event_subscriber_.GetRtcpEventsAndReset(&rtcp_events);
-  rtcp_.SendRtcpFromRtpReceiver(&cast_message, &rtcp_events);
-}
-
-// Cast messages should be sent within a maximum interval. Schedule a call
-// if not triggered elsewhere, e.g. by the cast message_builder.
-void VideoReceiver::ScheduleNextCastMessage() {
-  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
-  base::TimeTicks send_time;
-  framer_.TimeToSendNextCastMessage(&send_time);
-  base::TimeDelta time_to_send =
-      send_time - cast_environment_->Clock()->NowTicks();
-  time_to_send = std::max(
-      time_to_send, base::TimeDelta::FromMilliseconds(kMinSchedulingDelayMs));
-  cast_environment_->PostDelayedTask(
-      CastEnvironment::MAIN,
-      FROM_HERE,
-      base::Bind(&VideoReceiver::SendNextCastMessage,
-                 weak_factory_.GetWeakPtr()),
-      time_to_send);
-}
-
-void VideoReceiver::SendNextCastMessage() {
-  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
-  framer_.SendCastMessage();  // Will only send a message if it is time.
-  ScheduleNextCastMessage();
-}
-
-void VideoReceiver::ScheduleNextRtcpReport() {
-  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
-  base::TimeDelta time_to_next = rtcp_.TimeToSendNextRtcpReport() -
-                                 cast_environment_->Clock()->NowTicks();
-
-  time_to_next = std::max(
-      time_to_next, base::TimeDelta::FromMilliseconds(kMinSchedulingDelayMs));
-
-  cast_environment_->PostDelayedTask(
-      CastEnvironment::MAIN,
-      FROM_HERE,
-      base::Bind(&VideoReceiver::SendNextRtcpReport,
-                 weak_factory_.GetWeakPtr()),
-      time_to_next);
-}
-
-void VideoReceiver::SendNextRtcpReport() {
-  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
-  rtcp_.SendRtcpFromRtpReceiver(NULL, NULL);
-  ScheduleNextRtcpReport();
-}
-
-}  // namespace cast
-}  // namespace media
diff --git a/media/cast/video_receiver/video_receiver_unittest.cc b/media/cast/video_receiver/video_receiver_unittest.cc
deleted file mode 100644
index d158d7e..0000000
--- a/media/cast/video_receiver/video_receiver_unittest.cc
+++ /dev/null
@@ -1,266 +0,0 @@
-// Copyright 2013 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.
-
-#include <deque>
-#include <utility>
-
-#include "base/bind.h"
-#include "base/memory/ref_counted.h"
-#include "base/memory/scoped_ptr.h"
-#include "base/test/simple_test_tick_clock.h"
-#include "media/cast/cast_defines.h"
-#include "media/cast/cast_environment.h"
-#include "media/cast/logging/simple_event_subscriber.h"
-#include "media/cast/rtcp/test_rtcp_packet_builder.h"
-#include "media/cast/test/fake_single_thread_task_runner.h"
-#include "media/cast/test/utility/default_config.h"
-#include "media/cast/transport/pacing/mock_paced_packet_sender.h"
-#include "media/cast/video_receiver/video_receiver.h"
-#include "testing/gmock/include/gmock/gmock.h"
-
-using ::testing::_;
-
-namespace media {
-namespace cast {
-
-namespace {
-
-const int kPacketSize = 1500;
-const uint32 kFirstFrameId = 1234;
-const int kPlayoutDelayMillis = 100;
-
-class FakeVideoClient {
- public:
-  FakeVideoClient() : num_called_(0) {}
-  virtual ~FakeVideoClient() {}
-
-  void AddExpectedResult(uint32 expected_frame_id,
-                         const base::TimeTicks& expected_playout_time) {
-    expected_results_.push_back(
-        std::make_pair(expected_frame_id, expected_playout_time));
-  }
-
-  void DeliverEncodedVideoFrame(
-      scoped_ptr<transport::EncodedFrame> video_frame) {
-    SCOPED_TRACE(::testing::Message() << "num_called_ is " << num_called_);
-    ASSERT_FALSE(!video_frame)
-        << "If at shutdown: There were unsatisfied requests enqueued.";
-    ASSERT_FALSE(expected_results_.empty());
-    EXPECT_EQ(expected_results_.front().first, video_frame->frame_id);
-    EXPECT_EQ(expected_results_.front().second, video_frame->reference_time);
-    expected_results_.pop_front();
-    ++num_called_;
-  }
-
-  int number_times_called() const { return num_called_; }
-
- private:
-  std::deque<std::pair<uint32, base::TimeTicks> > expected_results_;
-  int num_called_;
-
-  DISALLOW_COPY_AND_ASSIGN(FakeVideoClient);
-};
-}  // namespace
-
-class VideoReceiverTest : public ::testing::Test {
- protected:
-  VideoReceiverTest() {
-    config_ = GetDefaultVideoReceiverConfig();
-    config_.rtp_max_delay_ms = kPlayoutDelayMillis;
-    // Note: Frame rate must divide 1000 without remainder so the test code
-    // doesn't have to account for rounding errors.
-    config_.max_frame_rate = 25;
-    config_.codec.video = transport::kVp8;  // Frame skipping not allowed.
-    testing_clock_ = new base::SimpleTestTickClock();
-    testing_clock_->Advance(base::TimeTicks::Now() - base::TimeTicks());
-    start_time_ = testing_clock_->NowTicks();
-    task_runner_ = new test::FakeSingleThreadTaskRunner(testing_clock_);
-
-    cast_environment_ =
-        new CastEnvironment(scoped_ptr<base::TickClock>(testing_clock_).Pass(),
-                            task_runner_,
-                            task_runner_,
-                            task_runner_);
-
-    receiver_.reset(new VideoReceiver(
-        cast_environment_, config_, &mock_transport_));
-  }
-
-  virtual ~VideoReceiverTest() {}
-
-  virtual void SetUp() {
-    payload_.assign(kPacketSize, 0);
-
-    // Always start with a key frame.
-    rtp_header_.is_key_frame = true;
-    rtp_header_.frame_id = kFirstFrameId;
-    rtp_header_.reference_frame_id = rtp_header_.frame_id;
-    rtp_header_.packet_id = 0;
-    rtp_header_.max_packet_id = 0;
-  }
-
-  void FeedOneFrameIntoReceiver() {
-    receiver_->OnReceivedPayloadData(
-        payload_.data(), payload_.size(), rtp_header_);
-  }
-
-  void FeedLipSyncInfoIntoReceiver() {
-    const base::TimeTicks now = testing_clock_->NowTicks();
-    const int64 rtp_timestamp = (now - start_time_) *
-        kVideoFrequency / base::TimeDelta::FromSeconds(1);
-    CHECK_LE(0, rtp_timestamp);
-    uint32 ntp_seconds;
-    uint32 ntp_fraction;
-    ConvertTimeTicksToNtp(now, &ntp_seconds, &ntp_fraction);
-    TestRtcpPacketBuilder rtcp_packet;
-    rtcp_packet.AddSrWithNtp(config_.incoming_ssrc,
-                             ntp_seconds, ntp_fraction,
-                             static_cast<uint32>(rtp_timestamp));
-    receiver_->IncomingPacket(rtcp_packet.GetPacket().Pass());
-  }
-
-  FrameReceiverConfig config_;
-  std::vector<uint8> payload_;
-  RtpCastHeader rtp_header_;
-  base::SimpleTestTickClock* testing_clock_;  // Owned by CastEnvironment.
-  base::TimeTicks start_time_;
-  transport::MockPacedPacketSender mock_transport_;
-  scoped_refptr<test::FakeSingleThreadTaskRunner> task_runner_;
-  scoped_refptr<CastEnvironment> cast_environment_;
-  FakeVideoClient fake_video_client_;
-
-  // Important for the VideoReceiver to be declared last, since its dependencies
-  // must remain alive until after its destruction.
-  scoped_ptr<VideoReceiver> receiver_;
-
-  DISALLOW_COPY_AND_ASSIGN(VideoReceiverTest);
-};
-
-TEST_F(VideoReceiverTest, ReceivesOneFrame) {
-  SimpleEventSubscriber event_subscriber;
-  cast_environment_->Logging()->AddRawEventSubscriber(&event_subscriber);
-
-  EXPECT_CALL(mock_transport_, SendRtcpPacket(_, _))
-      .WillRepeatedly(testing::Return(true));
-
-  FeedLipSyncInfoIntoReceiver();
-  task_runner_->RunTasks();
-
-  // Enqueue a request for a video frame.
-  receiver_->GetEncodedVideoFrame(
-      base::Bind(&FakeVideoClient::DeliverEncodedVideoFrame,
-                 base::Unretained(&fake_video_client_)));
-
-  // The request should not be satisfied since no packets have been received.
-  task_runner_->RunTasks();
-  EXPECT_EQ(0, fake_video_client_.number_times_called());
-
-  // Deliver one video frame to the receiver and expect to get one frame back.
-  const base::TimeDelta target_playout_delay =
-      base::TimeDelta::FromMilliseconds(kPlayoutDelayMillis);
-  fake_video_client_.AddExpectedResult(
-      kFirstFrameId, testing_clock_->NowTicks() + target_playout_delay);
-  FeedOneFrameIntoReceiver();
-  task_runner_->RunTasks();
-  EXPECT_EQ(1, fake_video_client_.number_times_called());
-
-  std::vector<FrameEvent> frame_events;
-  event_subscriber.GetFrameEventsAndReset(&frame_events);
-
-  ASSERT_TRUE(!frame_events.empty());
-  EXPECT_EQ(FRAME_ACK_SENT, frame_events.begin()->type);
-  EXPECT_EQ(rtp_header_.frame_id, frame_events.begin()->frame_id);
-  EXPECT_EQ(rtp_header_.rtp_timestamp, frame_events.begin()->rtp_timestamp);
-
-  cast_environment_->Logging()->RemoveRawEventSubscriber(&event_subscriber);
-}
-
-TEST_F(VideoReceiverTest, ReceivesFramesRefusingToSkipAny) {
-  EXPECT_CALL(mock_transport_, SendRtcpPacket(_, _))
-      .WillRepeatedly(testing::Return(true));
-
-  const uint32 rtp_advance_per_frame =
-      config_.frequency / config_.max_frame_rate;
-  const base::TimeDelta time_advance_per_frame =
-      base::TimeDelta::FromSeconds(1) / config_.max_frame_rate;
-
-  // Feed and process lip sync in receiver.
-  FeedLipSyncInfoIntoReceiver();
-  task_runner_->RunTasks();
-  const base::TimeTicks first_frame_capture_time = testing_clock_->NowTicks();
-
-  // Enqueue a request for a video frame.
-  const FrameEncodedCallback frame_encoded_callback =
-      base::Bind(&FakeVideoClient::DeliverEncodedVideoFrame,
-                 base::Unretained(&fake_video_client_));
-  receiver_->GetEncodedVideoFrame(frame_encoded_callback);
-  task_runner_->RunTasks();
-  EXPECT_EQ(0, fake_video_client_.number_times_called());
-
-  // Receive one video frame and expect to see the first request satisfied.
-  const base::TimeDelta target_playout_delay =
-      base::TimeDelta::FromMilliseconds(kPlayoutDelayMillis);
-  fake_video_client_.AddExpectedResult(
-      kFirstFrameId, first_frame_capture_time + target_playout_delay);
-  rtp_header_.rtp_timestamp = 0;
-  FeedOneFrameIntoReceiver();
-  task_runner_->RunTasks();
-  EXPECT_EQ(1, fake_video_client_.number_times_called());
-
-  // Enqueue a second request for a video frame, but it should not be
-  // fulfilled yet.
-  receiver_->GetEncodedVideoFrame(frame_encoded_callback);
-  task_runner_->RunTasks();
-  EXPECT_EQ(1, fake_video_client_.number_times_called());
-
-  // Receive one video frame out-of-order: Make sure that we are not continuous
-  // and that the RTP timestamp represents a time in the future.
-  rtp_header_.is_key_frame = false;
-  rtp_header_.frame_id = kFirstFrameId + 2;  // "Frame 3"
-  rtp_header_.reference_frame_id = kFirstFrameId + 1;  // "Frame 2"
-  rtp_header_.rtp_timestamp += 2 * rtp_advance_per_frame;
-  FeedOneFrameIntoReceiver();
-
-  // Frame 2 should not come out at this point in time.
-  task_runner_->RunTasks();
-  EXPECT_EQ(1, fake_video_client_.number_times_called());
-
-  // Enqueue a third request for a video frame.
-  receiver_->GetEncodedVideoFrame(frame_encoded_callback);
-  task_runner_->RunTasks();
-  EXPECT_EQ(1, fake_video_client_.number_times_called());
-
-  // Now, advance time forward such that Frame 2 is now too late for playback.
-  // Regardless, the receiver must NOT emit Frame 3 yet because it is not
-  // allowed to skip frames for VP8.
-  testing_clock_->Advance(2 * time_advance_per_frame + target_playout_delay);
-  task_runner_->RunTasks();
-  EXPECT_EQ(1, fake_video_client_.number_times_called());
-
-  // Now receive Frame 2 and expect both the second and third requests to be
-  // fulfilled immediately.
-  fake_video_client_.AddExpectedResult(
-      kFirstFrameId + 1,  // "Frame 2"
-      first_frame_capture_time + 1 * time_advance_per_frame +
-          target_playout_delay);
-  fake_video_client_.AddExpectedResult(
-      kFirstFrameId + 2,  // "Frame 3"
-      first_frame_capture_time + 2 * time_advance_per_frame +
-          target_playout_delay);
-  --rtp_header_.frame_id;  // "Frame 2"
-  --rtp_header_.reference_frame_id;  // "Frame 1"
-  rtp_header_.rtp_timestamp -= rtp_advance_per_frame;
-  FeedOneFrameIntoReceiver();
-  task_runner_->RunTasks();
-  EXPECT_EQ(3, fake_video_client_.number_times_called());
-
-  // Move forward to the playout time of an unreceived Frame 5.  Expect no
-  // additional frames were emitted.
-  testing_clock_->Advance(3 * time_advance_per_frame);
-  task_runner_->RunTasks();
-  EXPECT_EQ(3, fake_video_client_.number_times_called());
-}
-
-}  // namespace cast
-}  // namespace media
diff --git a/media/cast/video_sender/video_sender.cc b/media/cast/video_sender/video_sender.cc
index 57b8ea7..2642e79 100644
--- a/media/cast/video_sender/video_sender.cc
+++ b/media/cast/video_sender/video_sender.cc
@@ -85,7 +85,7 @@ VideoSender::VideoSender(
                video_config.rtp_config.ssrc,
                video_config.incoming_feedback_ssrc,
                video_config.rtcp_c_name,
-               false));
+               VIDEO_EVENT));
   rtcp_->SetCastReceiverEventHistorySize(kReceiverRtcpEventHistorySize);
 
   // TODO(pwestin): pass cast_initialization_cb to |video_encoder_|