Merge from Chromium at DEPS revision 284076

This commit was generated by merge_to_master.py. Change-Id: I9a279485b02fe7ceddcd32d992a714ff132e99ae
author: Ben Murdoch <benm@google.com> 2014-07-20 18:25:52 -0700
committer: Ben Murdoch <benm@google.com> 2014-07-20 18:25:52 -0700
commit: 116680a4aac90f2aa7413d9095a592090648e557 (patch)
tree: f7c6fed0e63d6a2804243d4a31a752dca39fb076 /media/cast
parent: 1f14a4515e04c9ffc9bac4dd1e2f68611626b800 (diff)
download: chromium_org-116680a4aac90f2aa7413d9095a592090648e557.tar.gz
147 files changed, 3167 insertions, 2162 deletions
diff --git a/media/cast/DEPS b/media/cast/DEPS
index abee2864b3..e44afe05ab 100644
--- a/media/cast/DEPS
+++ b/media/cast/DEPS
@@ -3,6 +3,7 @@ include_rules = [
    "+media",
    "+net",
   "+third_party/libyuv",
+  "+third_party/mt19937ar",
   "+third_party/zlib",
   "+ui/gfx",
 ]
diff --git a/media/cast/README b/media/cast/README
deleted file mode 100644
index 0930c1e012..0000000000
--- a/media/cast/README
+++ /dev/null
@@ -1,61 +0,0 @@
-This directory contains a RTP/RTCP library used for the Cast mirroring
-protocol. This library is specifically built for low latency purposes and
-enables Chrome to send real-time video and audio streams. 
-
-CONTENTS
-
-cast/
-    Build rules and top level source files and headers.
-
-cast/audio_receiver/
-    Module for receiving and decodes audio RTP stream.
-
-cast/audio_sender/
-    Module for encoding and sending audio RTP stream.
-
-cast/congestion_control/
-    Bandwidth estimation and network congestion handling.
-
-cast/transport/pacing/
-    Module for rate limiting data outflow.
-
-cast/rtcp/
-    Module for handling RTCP messages.
-
-cast/rtp_common/
-    Module for common code used for RTP messages.
-
-cast/rtp_receiver/
-    Module for reciving RTP messages.
-
-cast/transport/rtp_sender/
-    Module for sending RTP messages.
-
-cast/test/
-    Module for test applications.
-
-cast/video_receiver/
-    Module for receiving and decodes video RTP stream.
-
-cast/video_sender/
-    Module for encoding and sending video RTP stream.
-
-DEPENDENCIES
-
-Content of this directory should only depend on:
-
-base/
-    Provides base libraries and platform independent layer.
-
-net/
-    Provides network capabilities.
-
-third_party/libvpx
-    Provides video encoder.
-
-third_party/opus
-    Provides audio encoder.
-
-OWNERS
-
-See OWNERS for ownership.
diff --git a/media/cast/cast.gyp b/media/cast/cast.gyp
index 5de8796079..fb1ee64e6c 100644
--- a/media/cast/cast.gyp
+++ b/media/cast/cast.gyp
@@ -22,6 +22,7 @@
       'dependencies': [
         'cast_logging_proto',
         '<(DEPTH)/base/base.gyp:base',
+        '<(DEPTH)/crypto/crypto.gyp:crypto',
         '<(DEPTH)/net/net.gyp:net',
       ],
       'export_dependent_settings': [
@@ -33,8 +34,10 @@
         'cast_defines.h',
         'cast_environment.cc',
         'cast_environment.h',
-        'base/clock_drift_smoother.cc',
-        'base/clock_drift_smoother.h',
+        'common/clock_drift_smoother.cc',
+        'common/clock_drift_smoother.h',
+        'common/transport_encryption_handler.cc',
+        'common/transport_encryption_handler.h',
         'logging/encoding_event_subscriber.cc',
         'logging/encoding_event_subscriber.h',
         'logging/log_deserializer.cc',
@@ -57,15 +60,10 @@
         'logging/simple_event_subscriber.h',
         'logging/stats_event_subscriber.cc',
         'logging/stats_event_subscriber.h',
-        'rtp_timestamp_helper.cc',
-        'rtp_timestamp_helper.h',
-        'transport/cast_transport_config.cc',
-        'transport/cast_transport_config.h',
-        'transport/cast_transport_defines.h',
-        'transport/cast_transport_sender.h',
       ], # source
     },
     {
+      # GN version: //media/cast/logging/proto
       'target_name': 'cast_logging_proto',
       'type': 'static_library',
       'include_dirs': [
@@ -89,8 +87,7 @@
       ],
       'dependencies': [
         'cast_base',
-        'cast_rtcp',
-        'cast_transport',
+        'cast_net',
         '<(DEPTH)/base/base.gyp:base',
         '<(DEPTH)/media/media.gyp:media',
         '<(DEPTH)/media/media.gyp:shared_memory_support',
@@ -100,14 +97,6 @@
       ],
       'sources': [
         'cast_receiver.h',
-        'framer/cast_message_builder.cc',
-        'framer/cast_message_builder.h',
-        'framer/frame_buffer.cc',
-        'framer/frame_buffer.h',
-        'framer/frame_id_map.cc',
-        'framer/frame_id_map.h',
-        'framer/framer.cc',
-        'framer/framer.h',
         'receiver/audio_decoder.cc',
         'receiver/audio_decoder.h',
         'receiver/cast_receiver_impl.cc',
@@ -116,39 +105,20 @@
         '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_defines.cc',
-        'rtp_receiver/rtp_receiver_defines.h',
-        'rtp_receiver/rtp_parser/rtp_parser.cc',
-        'rtp_receiver/rtp_parser/rtp_parser.h',
-      ], # source
-    },
-    {
-      'target_name': 'cast_rtcp',
-      'type': 'static_library',
-      'include_dirs': [
-        '<(DEPTH)/',
-      ],
-      'dependencies': [
-        'cast_base',
-        'cast_transport',
-        '<(DEPTH)/base/base.gyp:base',
-        '<(DEPTH)/net/net.gyp:net',
-      ],
-      'sources': [
-        'rtcp/rtcp_defines.cc',
-        'rtcp/rtcp_defines.h',
-        'rtcp/rtcp.h',
-        'rtcp/rtcp.cc',
-        'rtcp/rtcp_receiver.cc',
-        'rtcp/rtcp_receiver.h',
-        'rtcp/rtcp_sender.cc',
-        'rtcp/rtcp_sender.h',
-        'rtcp/rtcp_utility.cc',
-        'rtcp/rtcp_utility.h',
-        'rtcp/receiver_rtcp_event_subscriber.cc',
-        'rtcp/receiver_rtcp_event_subscriber.cc',
+        'net/rtp/cast_message_builder.cc',
+        'net/rtp/cast_message_builder.h',
+        'net/rtp/frame_buffer.cc',
+        'net/rtp/frame_buffer.h',
+        'net/rtp/frame_id_map.cc',
+        'net/rtp/frame_id_map.h',
+        'net/rtp/framer.cc',
+        'net/rtp/framer.h',
+        'net/rtp/receiver_stats.cc',
+        'net/rtp/receiver_stats.h',
+        'net/rtp/rtp_parser.cc',
+        'net/rtp/rtp_parser.h',
+        'net/rtp/rtp_receiver_defines.cc',
+        'net/rtp/rtp_receiver_defines.h',
       ], # source
     },
     {
@@ -159,39 +129,40 @@
       ],
       'dependencies': [
         'cast_base',
-        'cast_rtcp',
-        'cast_transport',
+        'cast_net',
         '<(DEPTH)/media/media.gyp:media',
         '<(DEPTH)/media/media.gyp:shared_memory_support',
         '<(DEPTH)/third_party/opus/opus.gyp:opus',
         '<(DEPTH)/third_party/libvpx/libvpx.gyp:libvpx',
       ], # dependencies
       'sources': [
-        'audio_sender/audio_encoder.h',
-        'audio_sender/audio_encoder.cc',
-        'audio_sender/audio_sender.h',
-        'audio_sender/audio_sender.cc',
         'cast_sender.h',
         'cast_sender_impl.cc',
         'cast_sender_impl.h',
-        'congestion_control/congestion_control.h',
-        'congestion_control/congestion_control.cc',
-        'video_sender/codecs/vp8/vp8_encoder.cc',
-        'video_sender/codecs/vp8/vp8_encoder.h',
-        'video_sender/external_video_encoder.h',
-        'video_sender/external_video_encoder.cc',
-        'video_sender/fake_software_video_encoder.h',
-        'video_sender/fake_software_video_encoder.cc',
-        'video_sender/software_video_encoder.h',
-        'video_sender/video_encoder.h',
-        'video_sender/video_encoder_impl.h',
-        'video_sender/video_encoder_impl.cc',
-        'video_sender/video_sender.h',
-        'video_sender/video_sender.cc',
+        'sender/audio_encoder.h',
+        'sender/audio_encoder.cc',
+        'sender/audio_sender.h',
+        'sender/audio_sender.cc',
+        'sender/congestion_control.h',
+        'sender/congestion_control.cc',
+        'sender/external_video_encoder.h',
+        'sender/external_video_encoder.cc',
+        'sender/fake_software_video_encoder.h',
+        'sender/fake_software_video_encoder.cc',
+        'sender/rtp_timestamp_helper.cc',
+        'sender/rtp_timestamp_helper.h',
+        'sender/software_video_encoder.h',
+        'sender/video_encoder.h',
+        'sender/video_encoder_impl.h',
+        'sender/video_encoder_impl.cc',
+        'sender/video_sender.h',
+        'sender/video_sender.cc',
+        'sender/vp8_encoder.cc',
+        'sender/vp8_encoder.h',
       ], # source
     },
     {
-      'target_name': 'cast_transport',
+      'target_name': 'cast_net',
       'type': 'static_library',
       'include_dirs': [
         '<(DEPTH)/',
@@ -199,26 +170,38 @@
       'dependencies': [
         'cast_base',
         '<(DEPTH)/base/base.gyp:base',
-        '<(DEPTH)/crypto/crypto.gyp:crypto',
         '<(DEPTH)/net/net.gyp:net',
       ],
       'sources': [
-        'transport/cast_transport_sender_impl.cc',
-        'transport/cast_transport_sender_impl.h',
-        'transport/pacing/paced_sender.cc',
-        'transport/pacing/paced_sender.h',
-        'transport/rtcp/rtcp_builder.cc',
-        'transport/rtcp/rtcp_builder.h',
-        'transport/rtp_sender/packet_storage/packet_storage.cc',
-        'transport/rtp_sender/packet_storage/packet_storage.h',
-        'transport/rtp_sender/rtp_packetizer/rtp_packetizer.cc',
-        'transport/rtp_sender/rtp_packetizer/rtp_packetizer.h',
-        'transport/rtp_sender/rtp_sender.cc',
-        'transport/rtp_sender/rtp_sender.h',
-        'transport/transport/udp_transport.cc',
-        'transport/transport/udp_transport.h',
-        'transport/utility/transport_encryption_handler.cc',
-        'transport/utility/transport_encryption_handler.h',
+        'net/cast_transport_config.cc',
+        'net/cast_transport_config.h',
+        'net/cast_transport_defines.h',
+        'net/cast_transport_sender.h',
+        'net/cast_transport_sender_impl.cc',
+        'net/cast_transport_sender_impl.h',
+        'net/pacing/paced_sender.cc',
+        'net/pacing/paced_sender.h',
+        'net/rtcp/receiver_rtcp_event_subscriber.cc',
+        'net/rtcp/rtcp_builder.cc',
+        'net/rtcp/rtcp_builder.h',
+        'net/rtcp/rtcp_defines.cc',
+        'net/rtcp/rtcp_defines.h',
+        'net/rtcp/rtcp.h',
+        'net/rtcp/rtcp.cc',
+        'net/rtcp/rtcp_receiver.cc',
+        'net/rtcp/rtcp_receiver.h',
+        'net/rtcp/rtcp_sender.cc',
+        'net/rtcp/rtcp_sender.h',
+        'net/rtcp/rtcp_utility.cc',
+        'net/rtcp/rtcp_utility.h',
+        'net/rtp/packet_storage.cc',
+        'net/rtp/packet_storage.h',
+        'net/rtp/rtp_packetizer.cc',
+        'net/rtp/rtp_packetizer.h',
+        'net/rtp/rtp_sender.cc',
+        'net/rtp/rtp_sender.h',
+        'net/udp_transport.cc',
+        'net/udp_transport.h',
       ], # source
     },
   ],
diff --git a/media/cast/cast_config.cc b/media/cast/cast_config.cc
index 0e7953af01..f74121632c 100644
--- a/media/cast/cast_config.cc
+++ b/media/cast/cast_config.cc
@@ -22,9 +22,13 @@ namespace cast {
 // these classes to centralize the logic?
 
 VideoSenderConfig::VideoSenderConfig()
-    : incoming_feedback_ssrc(0),
+    : ssrc(0),
+      incoming_feedback_ssrc(0),
       rtcp_interval(kDefaultRtcpIntervalMs),
       rtcp_mode(kRtcpReducedSize),
+      target_playout_delay(
+          base::TimeDelta::FromMilliseconds(kDefaultRtpMaxDelayMs)),
+      rtp_payload_type(0),
       use_external_encoder(false),
       width(0),
       height(0),
@@ -36,17 +40,26 @@ VideoSenderConfig::VideoSenderConfig()
       min_qp(kDefaultMinQp),
       max_frame_rate(kDefaultMaxFrameRate),
       max_number_of_video_buffers_used(kDefaultNumberOfVideoBuffers),
-      codec(transport::kVp8),
+      codec(CODEC_VIDEO_VP8),
       number_of_encode_threads(1) {}
 
+VideoSenderConfig::~VideoSenderConfig() {}
+
 AudioSenderConfig::AudioSenderConfig()
-    : incoming_feedback_ssrc(0),
+    : ssrc(0),
+      incoming_feedback_ssrc(0),
       rtcp_interval(kDefaultRtcpIntervalMs),
       rtcp_mode(kRtcpReducedSize),
+      target_playout_delay(
+          base::TimeDelta::FromMilliseconds(kDefaultRtpMaxDelayMs)),
+      rtp_payload_type(0),
       use_external_encoder(false),
       frequency(0),
       channels(0),
-      bitrate(0) {}
+      bitrate(0),
+      codec(CODEC_AUDIO_OPUS) {}
+
+AudioSenderConfig::~AudioSenderConfig() {}
 
 FrameReceiverConfig::FrameReceiverConfig()
     : feedback_ssrc(0),
@@ -57,7 +70,8 @@ FrameReceiverConfig::FrameReceiverConfig()
       rtp_payload_type(0),
       frequency(0),
       channels(0),
-      max_frame_rate(0) {}
+      max_frame_rate(0),
+      codec(CODEC_UNKNOWN) {}
 
 FrameReceiverConfig::~FrameReceiverConfig() {}
 
diff --git a/media/cast/cast_config.h b/media/cast/cast_config.h
index ea25d6b6cf..c2e797b63a 100644
--- a/media/cast/cast_config.h
+++ b/media/cast/cast_config.h
@@ -14,51 +14,79 @@
 #include "base/memory/ref_counted.h"
 #include "base/memory/shared_memory.h"
 #include "base/single_thread_task_runner.h"
+#include "base/time/time.h"
 #include "media/cast/cast_defines.h"
-#include "media/cast/transport/cast_transport_config.h"
+#include "media/cast/net/cast_transport_config.h"
 
 namespace media {
 class VideoEncodeAccelerator;
 
 namespace cast {
 
-enum RtcpMode {
-  kRtcpCompound,     // Compound RTCP mode is described by RFC 4585.
-  kRtcpReducedSize,  // Reduced-size RTCP mode is described by RFC 5506.
-};
-
 // TODO(miu): Merge AudioSenderConfig and VideoSenderConfig and make their
 // naming/documentation consistent with FrameReceiverConfig.
 struct AudioSenderConfig {
   AudioSenderConfig();
+  ~AudioSenderConfig();
+
+  // Identifier referring to the sender, used by the receiver.
+  uint32 ssrc;
 
-  // The sender ssrc is in rtp_config.ssrc.
+  // The receiver's SSRC identifier.
   uint32 incoming_feedback_ssrc;
 
   int rtcp_interval;
   std::string rtcp_c_name;
   RtcpMode rtcp_mode;
 
-  transport::RtpConfig rtp_config;
+  // 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.).
+  base::TimeDelta target_playout_delay;
+
+  // RTP payload type enum: Specifies the type/encoding of frame data.
+  int rtp_payload_type;
 
   bool use_external_encoder;
   int frequency;
   int channels;
   int bitrate;  // Set to <= 0 for "auto variable bitrate" (libopus knows best).
-  transport::AudioCodec codec;
+  Codec codec;
+
+  // The AES crypto key and initialization vector.  Each of these strings
+  // contains the data in binary form, of size kAesKeySize.  If they are empty
+  // strings, crypto is not being used.
+  std::string aes_key;
+  std::string aes_iv_mask;
 };
 
 struct VideoSenderConfig {
   VideoSenderConfig();
+  ~VideoSenderConfig();
 
-  // The sender ssrc is in rtp_config.ssrc.
-  uint32 incoming_feedback_ssrc;
+  // Identifier referring to the sender, used by the receiver.
+  uint32 ssrc;
+
+  // The receiver's SSRC identifier.
+  uint32 incoming_feedback_ssrc;  // TODO(miu): Rename to receiver_ssrc.
 
   int rtcp_interval;
   std::string rtcp_c_name;
   RtcpMode rtcp_mode;
 
-  transport::RtpConfig rtp_config;
+  // 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.).
+  base::TimeDelta target_playout_delay;
+
+  // RTP payload type enum: Specifies the type/encoding of frame data.
+  int rtp_payload_type;
 
   bool use_external_encoder;
   int width;  // Incoming frames will be scaled to this size.
@@ -72,8 +100,14 @@ struct VideoSenderConfig {
   int min_qp;
   int max_frame_rate;
   int max_number_of_video_buffers_used;  // Max value depend on codec.
-  transport::VideoCodec codec;
+  Codec codec;
   int number_of_encode_threads;
+
+  // The AES crypto key and initialization vector.  Each of these strings
+  // contains the data in binary form, of size kAesKeySize.  If they are empty
+  // strings, crypto is not being used.
+  std::string aes_key;
+  std::string aes_iv_mask;
 };
 
 // TODO(miu): Naming and minor type changes are badly needed in a later CL.
@@ -127,11 +161,7 @@ struct FrameReceiverConfig {
   // Codec used for the compression of signal data.
   // TODO(miu): Merge the AudioCodec and VideoCodec enums into one so this union
   // is not necessary.
-  union MergedCodecPlaceholder {
-    transport::AudioCodec audio;
-    transport::VideoCodec video;
-    MergedCodecPlaceholder() : audio(transport::kUnknownAudioCodec) {}
-  } codec;
+  Codec codec;
 
   // The AES crypto key and initialization vector.  Each of these strings
   // contains the data in binary form, of size kAesKeySize.  If they are empty
@@ -140,9 +170,10 @@ struct FrameReceiverConfig {
   std::string aes_iv_mask;
 };
 
-// import from media::cast::transport
-typedef transport::Packet Packet;
-typedef transport::PacketList PacketList;
+// Import from media::cast.
+
+typedef Packet Packet;
+typedef PacketList PacketList;
 
 typedef base::Callback<void(CastInitializationStatus)>
     CastInitializationCallback;
diff --git a/media/cast/cast_defines.h b/media/cast/cast_defines.h
index 64b20c96da..07be3b3f9e 100644
--- a/media/cast/cast_defines.h
+++ b/media/cast/cast_defines.h
@@ -14,7 +14,7 @@
 #include "base/compiler_specific.h"
 #include "base/logging.h"
 #include "base/time/time.h"
-#include "media/cast/transport/cast_transport_config.h"
+#include "media/cast/net/cast_transport_config.h"
 
 namespace media {
 namespace cast {
@@ -29,7 +29,6 @@ const uint32 kStartFrameId = UINT32_C(0xffffffff);
 // frames.
 const int kMaxUnackedFrames = 255;
 
-const size_t kMaxIpPacketSize = 1500;
 const int kStartRttMs = 20;
 const int64 kCastMessageUpdateIntervalMs = 33;
 const int64 kNackRepeatIntervalMs = 30;
@@ -193,13 +192,6 @@ inline base::TimeDelta RtpDeltaToTimeDelta(int64 rtp_delta, int rtp_timebase) {
   return rtp_delta * base::TimeDelta::FromSeconds(1) / rtp_timebase;
 }
 
-inline uint32 GetVideoRtpTimestamp(const base::TimeTicks& time_ticks) {
-  base::TimeTicks zero_time;
-  base::TimeDelta recorded_delta = time_ticks - zero_time;
-  // Timestamp is in 90 KHz for video.
-  return static_cast<uint32>(recorded_delta.InMilliseconds() * 90);
-}
-
 }  // namespace cast
 }  // namespace media
 
diff --git a/media/cast/cast_receiver.h b/media/cast/cast_receiver.h
index a9d3edeb78..f57942231d 100644
--- a/media/cast/cast_receiver.h
+++ b/media/cast/cast_receiver.h
@@ -22,9 +22,7 @@ class VideoFrame;
 
 namespace cast {
 
-namespace transport {
 class PacketSender;
-}
 
 // The following callbacks are used to deliver decoded audio/video frame data,
 // the frame's corresponding play-out time, and a continuity flag.
@@ -44,7 +42,7 @@ typedef base::Callback<void(const scoped_refptr<media::VideoFrame>& video_frame,
 // should examine the |frame_id| field to determine whether any frames have been
 // 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>)>
+typedef base::Callback<void(scoped_ptr<EncodedFrame>)>
     ReceiveEncodedFrameCallback;
 
 class CastReceiver {
@@ -53,13 +51,13 @@ class CastReceiver {
       scoped_refptr<CastEnvironment> cast_environment,
       const FrameReceiverConfig& audio_config,
       const FrameReceiverConfig& video_config,
-      transport::PacketSender* const packet_sender);
+      PacketSender* const packet_sender);
 
   // All received RTP and RTCP packets for the call should be sent to this
   // 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;
+  virtual PacketReceiverCallback packet_receiver() = 0;
 
   // Polling interface to get audio and video frames from the CastReceiver.  The
   // the RequestDecodedXXXXXFrame() methods utilize internal software-based
diff --git a/media/cast/cast_sender.h b/media/cast/cast_sender.h
index eb3327ff3d..abe0a01713 100644
--- a/media/cast/cast_sender.h
+++ b/media/cast/cast_sender.h
@@ -19,7 +19,7 @@
 #include "media/base/audio_bus.h"
 #include "media/cast/cast_config.h"
 #include "media/cast/cast_environment.h"
-#include "media/cast/transport/cast_transport_sender.h"
+#include "media/cast/net/cast_transport_sender.h"
 
 namespace media {
 class VideoFrame;
@@ -63,7 +63,7 @@ class CastSender {
  public:
   static scoped_ptr<CastSender> Create(
       scoped_refptr<CastEnvironment> cast_environment,
-      transport::CastTransportSender* const transport_sender);
+      CastTransportSender* const transport_sender);
 
   virtual ~CastSender() {}
 
@@ -75,7 +75,7 @@ class CastSender {
 
   // All RTCP packets for the session should be inserted to this object.
   // This function and the callback must be called on the main thread.
-  virtual transport::PacketReceiverCallback packet_receiver() = 0;
+  virtual PacketReceiverCallback packet_receiver() = 0;
 
   // Initialize the audio stack. Must be called in order to send audio frames.
   // Status of the initialization will be returned on cast_initialization_cb.
diff --git a/media/cast/cast_sender_impl.cc b/media/cast/cast_sender_impl.cc
index 361e4d8dc1..a120d3a930 100644
--- a/media/cast/cast_sender_impl.cc
+++ b/media/cast/cast_sender_impl.cc
@@ -1,6 +1,7 @@
 // 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_sender_impl.h"
 
 #include "base/bind.h"
@@ -75,7 +76,7 @@ class LocalAudioFrameInput : public AudioFrameInput {
 
 scoped_ptr<CastSender> CastSender::Create(
     scoped_refptr<CastEnvironment> cast_environment,
-    transport::CastTransportSender* const transport_sender) {
+    CastTransportSender* const transport_sender) {
   CHECK(cast_environment);
   return scoped_ptr<CastSender>(
       new CastSenderImpl(cast_environment, transport_sender));
@@ -83,7 +84,7 @@ scoped_ptr<CastSender> CastSender::Create(
 
 CastSenderImpl::CastSenderImpl(
     scoped_refptr<CastEnvironment> cast_environment,
-    transport::CastTransportSender* const transport_sender)
+    CastTransportSender* const transport_sender)
     : cast_environment_(cast_environment),
       transport_sender_(transport_sender),
       weak_factory_(this) {
@@ -211,7 +212,7 @@ scoped_refptr<VideoFrameInput> CastSenderImpl::video_frame_input() {
   return video_frame_input_;
 }
 
-transport::PacketReceiverCallback CastSenderImpl::packet_receiver() {
+PacketReceiverCallback CastSenderImpl::packet_receiver() {
   return base::Bind(&CastSenderImpl::ReceivedPacket,
                     weak_factory_.GetWeakPtr());
 }
diff --git a/media/cast/cast_sender_impl.h b/media/cast/cast_sender_impl.h
index d09a869712..bc320f7b54 100644
--- a/media/cast/cast_sender_impl.h
+++ b/media/cast/cast_sender_impl.h
@@ -6,12 +6,12 @@
 
 #include "base/memory/ref_counted.h"
 #include "base/memory/scoped_ptr.h"
-#include "media/cast/audio_sender/audio_sender.h"
 #include "media/cast/cast_config.h"
 #include "media/cast/cast_defines.h"
 #include "media/cast/cast_environment.h"
 #include "media/cast/cast_sender.h"
-#include "media/cast/video_sender/video_sender.h"
+#include "media/cast/sender/audio_sender.h"
+#include "media/cast/sender/video_sender.h"
 
 namespace media {
 class VideoFrame;
@@ -25,7 +25,7 @@ class VideoSender;
 class CastSenderImpl : public CastSender {
  public:
   CastSenderImpl(scoped_refptr<CastEnvironment> cast_environment,
-                 transport::CastTransportSender* const transport_sender);
+                 CastTransportSender* const transport_sender);
 
   virtual void InitializeAudio(
       const AudioSenderConfig& audio_config,
@@ -42,7 +42,7 @@ class CastSenderImpl : public CastSender {
   virtual scoped_refptr<AudioFrameInput> audio_frame_input() OVERRIDE;
   virtual scoped_refptr<VideoFrameInput> video_frame_input() OVERRIDE;
 
-  virtual transport::PacketReceiverCallback packet_receiver() OVERRIDE;
+  virtual PacketReceiverCallback packet_receiver() OVERRIDE;
 
  private:
   void ReceivedPacket(scoped_ptr<Packet> packet);
@@ -55,7 +55,7 @@ class CastSenderImpl : public CastSender {
   scoped_refptr<CastEnvironment> cast_environment_;
   // The transport sender is owned by the owner of the CastSender, and should be
   // valid throughout the lifetime of the CastSender.
-  transport::CastTransportSender* const transport_sender_;
+  CastTransportSender* const transport_sender_;
   uint32 ssrc_of_audio_sender_;
   uint32 ssrc_of_video_sender_;
 
diff --git a/media/cast/cast_testing.gypi b/media/cast/cast_testing.gypi
index aef0fbd8c3..97e831e42b 100644
--- a/media/cast/cast_testing.gypi
+++ b/media/cast/cast_testing.gypi
@@ -11,19 +11,25 @@
          '<(DEPTH)/',
       ],
       'dependencies': [
+        'cast_net',
         'cast_receiver',
-        'cast_transport',
         '<(DEPTH)/testing/gtest.gyp:gtest',
+        '<(DEPTH)/third_party/ffmpeg/ffmpeg.gyp:ffmpeg',
         '<(DEPTH)/third_party/libyuv/libyuv.gyp:libyuv',
+        '<(DEPTH)/third_party/mt19937ar/mt19937ar.gyp:mt19937ar',
         '<(DEPTH)/ui/gfx/gfx.gyp:gfx_geometry',
       ],
       'sources': [
+        'test/fake_media_source.cc',
+        'test/fake_media_source.h',
         'test/fake_single_thread_task_runner.cc',
         'test/fake_single_thread_task_runner.h',
         'test/skewed_single_thread_task_runner.cc',
         'test/skewed_single_thread_task_runner.h',
         'test/skewed_tick_clock.cc',
         'test/skewed_tick_clock.h',
+	'test/loopback_transport.cc',
+	'test/loopback_transport.h',
         'test/utility/audio_utility.cc',
         'test/utility/audio_utility.h',
         'test/utility/barcode.cc',
@@ -52,14 +58,13 @@
       ],
       'dependencies': [
         'cast_base',
+        'cast_net',
         'cast_receiver',
-        'cast_rtcp',
         'cast_sender',
         'cast_test_utility',
         # Not a true dependency. This is here to make sure the CQ can verify
         # the tools compile correctly.
         'cast_tools',
-        'cast_transport',
         '<(DEPTH)/base/base.gyp:test_support_base',
         '<(DEPTH)/net/net.gyp:net',
         '<(DEPTH)/testing/gmock.gyp:gmock',
@@ -67,12 +72,6 @@
       ],
       'sources': [
         '<(DEPTH)/media/base/run_all_unittests.cc',
-        'audio_sender/audio_encoder_unittest.cc',
-        'audio_sender/audio_sender_unittest.cc',
-        'congestion_control/congestion_control_unittest.cc',
-        'framer/cast_message_builder_unittest.cc',
-        'framer/frame_buffer_unittest.cc',
-        'framer/framer_unittest.cc',
         'logging/encoding_event_subscriber_unittest.cc',
         'logging/serialize_deserialize_test.cc',
         'logging/logging_impl_unittest.cc',
@@ -80,26 +79,44 @@
         'logging/receiver_time_offset_estimator_impl_unittest.cc',
         'logging/simple_event_subscriber_unittest.cc',
         'logging/stats_event_subscriber_unittest.cc',
+        'net/cast_transport_sender_impl_unittest.cc',
+        'net/pacing/mock_paced_packet_sender.cc',
+        'net/pacing/mock_paced_packet_sender.h',
+        'net/pacing/paced_sender_unittest.cc',
+        'net/rtcp/mock_rtcp_receiver_feedback.cc',
+        'net/rtcp/mock_rtcp_receiver_feedback.h',
+        'net/rtcp/mock_rtcp_sender_feedback.cc',
+        'net/rtcp/mock_rtcp_sender_feedback.h',
+        'net/rtcp/rtcp_receiver_unittest.cc',
+        'net/rtcp/rtcp_sender_unittest.cc',
+        'net/rtcp/rtcp_unittest.cc',
+        'net/rtcp/receiver_rtcp_event_subscriber_unittest.cc',
+# TODO(miu): The following two are test utility modules.  Rename/move the files.
+        'net/rtcp/test_rtcp_packet_builder.cc',
+        'net/rtcp/test_rtcp_packet_builder.h',
+        'net/rtp/cast_message_builder_unittest.cc',
+        'net/rtp/frame_buffer_unittest.cc',
+        'net/rtp/framer_unittest.cc',
+        'net/rtp/mock_rtp_payload_feedback.cc',
+        'net/rtp/mock_rtp_payload_feedback.h',
+        'net/rtp/packet_storage_unittest.cc',
+        'net/rtp/receiver_stats_unittest.cc',
+        'net/rtp/rtp_header_parser.cc',
+        'net/rtp/rtp_header_parser.h',
+        'net/rtp/rtp_packet_builder.cc',
+        'net/rtp/rtp_parser_unittest.cc',
+        'net/rtp/rtp_packetizer_unittest.cc',
+        'net/rtp/rtp_receiver_defines.h',
+        'net/udp_transport_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',
-        'rtcp/mock_rtcp_sender_feedback.h',
-        'rtcp/rtcp_receiver_unittest.cc',
-        'rtcp/rtcp_sender_unittest.cc',
-        'rtcp/rtcp_unittest.cc',
-        'rtcp/receiver_rtcp_event_subscriber_unittest.cc',
-# TODO(miu): The following two are test utility modules.  Rename/move the files.
-        'rtcp/test_rtcp_packet_builder.cc',
-        'rtcp/test_rtcp_packet_builder.h',
-        'rtp_receiver/rtp_receiver_defines.h',
-        'rtp_receiver/mock_rtp_payload_feedback.cc',
-        'rtp_receiver/mock_rtp_payload_feedback.h',
-        'rtp_receiver/receiver_stats_unittest.cc',
-        'rtp_receiver/rtp_parser/test/rtp_packet_builder.cc',
-        'rtp_receiver/rtp_parser/rtp_parser_unittest.cc',
+        'sender/audio_encoder_unittest.cc',
+        'sender/audio_sender_unittest.cc',
+        'sender/congestion_control_unittest.cc',
+        'sender/external_video_encoder_unittest.cc',
+        'sender/video_encoder_impl_unittest.cc',
+        'sender/video_sender_unittest.cc',
         'test/end2end_unittest.cc',
         'test/fake_receiver_time_offset_estimator.cc',
         'test/fake_receiver_time_offset_estimator.h',
@@ -109,18 +126,6 @@
         'test/fake_video_encode_accelerator.h',
         'test/utility/audio_utility_unittest.cc',
         'test/utility/barcode_unittest.cc',
-        'transport/cast_transport_sender_impl_unittest.cc',
-        'transport/pacing/mock_paced_packet_sender.cc',
-        'transport/pacing/mock_paced_packet_sender.h',
-        'transport/pacing/paced_sender_unittest.cc',
-        'transport/rtp_sender/packet_storage/packet_storage_unittest.cc',
-        'transport/rtp_sender/rtp_packetizer/rtp_packetizer_unittest.cc',
-        '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_sender/external_video_encoder_unittest.cc',
-        'video_sender/video_encoder_impl_unittest.cc',
-        'video_sender/video_sender_unittest.cc',
       ], # source
     },
     {
@@ -131,11 +136,10 @@
       ],
       'dependencies': [
         'cast_base',
+        'cast_net',
         'cast_receiver',
-        'cast_rtcp',
         'cast_sender',
         'cast_test_utility',
-        'cast_transport',
         '<(DEPTH)/base/base.gyp:test_support_base',
         '<(DEPTH)/net/net.gyp:net',
         '<(DEPTH)/testing/gtest.gyp:gtest',
@@ -167,6 +171,7 @@
       'dependencies': [
         'cast_receiver_app',
         'cast_sender_app',
+        'cast_simulator',
         'udp_proxy',
       ],
     },
@@ -178,9 +183,9 @@
       ],
       'dependencies': [
         'cast_base',
+        'cast_net',
         'cast_receiver',
         'cast_test_utility',
-        'cast_transport',
         '<(DEPTH)/net/net.gyp:net_test_support',
         '<(DEPTH)/media/media.gyp:media',
         '<(DEPTH)/testing/gtest.gyp:gtest',
@@ -211,9 +216,9 @@
       ],
       'dependencies': [
         'cast_base',
+        'cast_net',
         'cast_sender',
         'cast_test_utility',
-        'cast_transport',
         '<(DEPTH)/net/net.gyp:net_test_support',
         '<(DEPTH)/media/media.gyp:media',
         '<(DEPTH)/testing/gtest.gyp:gtest',
@@ -226,6 +231,45 @@
       ],
     },
     {
+      'target_name': 'cast_simulator',
+      'type': 'executable',
+      'include_dirs': [
+        '<(DEPTH)/',
+      ],
+      'dependencies': [
+        'cast_base',
+        'cast_net',
+        'cast_network_model_proto',
+        'cast_sender',
+        'cast_test_utility',
+        '<(DEPTH)/net/net.gyp:net_test_support',
+        '<(DEPTH)/media/media.gyp:media',
+        '<(DEPTH)/testing/gtest.gyp:gtest',
+        '<(DEPTH)/third_party/ffmpeg/ffmpeg.gyp:ffmpeg',
+        '<(DEPTH)/third_party/opus/opus.gyp:opus',
+        '<(DEPTH)/ui/gfx/gfx.gyp:gfx_geometry',
+      ],
+      'sources': [
+        '<(DEPTH)/media/cast/test/simulator.cc',
+      ],
+    },
+    {
+      # GN version: //media/cast/test/proto
+      'target_name': 'cast_network_model_proto',
+      'type': 'static_library',
+      'include_dirs': [
+        '<(DEPTH)/',
+      ],
+      'sources': [
+        'test/proto/network_simulation_model.proto',
+      ],
+      'variables': {
+        'proto_in_dir': 'test/proto',
+        'proto_out_dir': 'media/cast/test/proto',
+      },
+      'includes': ['../../build/protoc.gypi'],
+    },
+    {
       'target_name': 'generate_barcode_video',
       'type': 'executable',
       'include_dirs': [
@@ -248,8 +292,8 @@
       ],
       'dependencies': [
         'cast_base',
+        'cast_net',
         'cast_test_utility',
-        'cast_transport',
         '<(DEPTH)/base/base.gyp:base',
         '<(DEPTH)/media/media.gyp:media',
       ],
diff --git a/media/cast/base/clock_drift_smoother.cc b/media/cast/common/clock_drift_smoother.cc
index ca0380533e..aff9a396de 100644
--- a/media/cast/base/clock_drift_smoother.cc
+++ b/media/cast/common/clock_drift_smoother.cc
@@ -2,7 +2,7 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-#include "media/cast/base/clock_drift_smoother.h"
+#include "media/cast/common/clock_drift_smoother.h"
 
 #include "base/logging.h"
 
diff --git a/media/cast/base/clock_drift_smoother.h b/media/cast/common/clock_drift_smoother.h
index 67de4cb51a..0511da9f3d 100644
--- a/media/cast/base/clock_drift_smoother.h
+++ b/media/cast/common/clock_drift_smoother.h
@@ -2,8 +2,8 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-#ifndef MEDIA_CAST_BASE_CLOCK_DRIFT_SMOOTHER_H_
-#define MEDIA_CAST_BASE_CLOCK_DRIFT_SMOOTHER_H_
+#ifndef MEDIA_CAST_COMMON_CLOCK_DRIFT_SMOOTHER_H_
+#define MEDIA_CAST_COMMON_CLOCK_DRIFT_SMOOTHER_H_
 
 #include "base/time/time.h"
 
@@ -49,4 +49,4 @@ class ClockDriftSmoother {
 }  // namespace cast
 }  // namespace media
 
-#endif  // MEDIA_CAST_BASE_CLOCK_DRIFT_SMOOTHER_H_
+#endif  // MEDIA_CAST_COMMON_CLOCK_DRIFT_SMOOTHER_H_
diff --git a/media/cast/transport/utility/transport_encryption_handler.cc b/media/cast/common/transport_encryption_handler.cc
index 89db2cf95b..54a43e8b52 100644
--- a/media/cast/transport/utility/transport_encryption_handler.cc
+++ b/media/cast/common/transport_encryption_handler.cc
@@ -2,32 +2,55 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-#include "media/cast/transport/utility/transport_encryption_handler.h"
+#include "media/cast/common/transport_encryption_handler.h"
 
 #include "base/logging.h"
 #include "crypto/encryptor.h"
 #include "crypto/symmetric_key.h"
-#include "media/cast/transport/cast_transport_defines.h"
+#include "media/cast/net/cast_transport_defines.h"
+
+namespace {
+
+// Crypto.
+const size_t kAesBlockSize = 16;
+const size_t kAesKeySize = 16;
+
+std::string GetAesNonce(uint32 frame_id, const std::string& iv_mask) {
+  std::string aes_nonce(kAesBlockSize, 0);
+
+  // Serializing frame_id in big-endian order (aes_nonce[8] is the most
+  // significant byte of frame_id).
+  aes_nonce[11] = frame_id & 0xff;
+  aes_nonce[10] = (frame_id >> 8) & 0xff;
+  aes_nonce[9] = (frame_id >> 16) & 0xff;
+  aes_nonce[8] = (frame_id >> 24) & 0xff;
+
+  for (size_t i = 0; i < kAesBlockSize; ++i) {
+    aes_nonce[i] ^= iv_mask[i];
+  }
+  return aes_nonce;
+}
+
+}  // namespace
 
 namespace media {
 namespace cast {
-namespace transport {
 
 TransportEncryptionHandler::TransportEncryptionHandler()
-    : key_(), encryptor_(), iv_mask_(), initialized_(false) {}
+    : key_(), encryptor_(), iv_mask_(), is_activated_(false) {}
 
 TransportEncryptionHandler::~TransportEncryptionHandler() {}
 
 bool TransportEncryptionHandler::Initialize(std::string aes_key,
                                             std::string aes_iv_mask) {
-  initialized_ = false;
+  is_activated_ = false;
   if (aes_iv_mask.size() == kAesKeySize && aes_key.size() == kAesKeySize) {
     iv_mask_ = aes_iv_mask;
     key_.reset(
         crypto::SymmetricKey::Import(crypto::SymmetricKey::AES, aes_key));
     encryptor_.reset(new crypto::Encryptor());
     encryptor_->Init(key_.get(), crypto::Encryptor::CTR, std::string());
-    initialized_ = true;
+    is_activated_ = true;
   } else if (aes_iv_mask.size() != 0 || aes_key.size() != 0) {
     DCHECK_EQ(aes_iv_mask.size(), 0u)
         << "Invalid Crypto configuration: aes_iv_mask.size";
@@ -41,7 +64,7 @@ bool TransportEncryptionHandler::Initialize(std::string aes_key,
 bool TransportEncryptionHandler::Encrypt(uint32 frame_id,
                                          const base::StringPiece& data,
                                          std::string* encrypted_data) {
-  if (!initialized_)
+  if (!is_activated_)
     return false;
   if (!encryptor_->SetCounter(GetAesNonce(frame_id, iv_mask_))) {
     NOTREACHED() << "Failed to set counter";
@@ -57,10 +80,10 @@ bool TransportEncryptionHandler::Encrypt(uint32 frame_id,
 bool TransportEncryptionHandler::Decrypt(uint32 frame_id,
                                          const base::StringPiece& ciphertext,
                                          std::string* plaintext) {
-  if (!initialized_) {
+  if (!is_activated_) {
     return false;
   }
-  if (!encryptor_->SetCounter(transport::GetAesNonce(frame_id, iv_mask_))) {
+  if (!encryptor_->SetCounter(GetAesNonce(frame_id, iv_mask_))) {
     NOTREACHED() << "Failed to set counter";
     return false;
   }
@@ -71,6 +94,5 @@ bool TransportEncryptionHandler::Decrypt(uint32 frame_id,
   return true;
 }
 
-}  // namespace transport
 }  // namespace cast
 }  // namespace media
diff --git a/media/cast/transport/utility/transport_encryption_handler.h b/media/cast/common/transport_encryption_handler.h
index 06d0e3f34d..d4798dc78b 100644
--- a/media/cast/transport/utility/transport_encryption_handler.h
+++ b/media/cast/common/transport_encryption_handler.h
@@ -2,8 +2,8 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-#ifndef MEDIA_CAST_TRANSPORT_TRANSPORT_UTILITY_ENCRYPTION_HANDLER_H_
-#define MEDIA_CAST_TRANSPORT_TRANSPORT_UTILITY_ENCRYPTION_HANDLER_H_
+#ifndef MEDIA_CAST_COMMON_TRANSPORT_ENCRYPTION_HANDLER_H_
+#define MEDIA_CAST_COMMON_TRANSPORT_ENCRYPTION_HANDLER_H_
 
 // Helper class to handle encryption for the Cast Transport library.
 #include <string>
@@ -20,7 +20,6 @@ class SymmetricKey;
 
 namespace media {
 namespace cast {
-namespace transport {
 
 class TransportEncryptionHandler : public base::NonThreadSafe {
  public:
@@ -37,22 +36,18 @@ class TransportEncryptionHandler : public base::NonThreadSafe {
                const base::StringPiece& ciphertext,
                std::string* plaintext);
 
-  // TODO(miu): This naming is very misleading.  It should be called
-  // is_activated() since Initialize() without keys (i.e., cypto is disabled)
-  // may have succeeded.
-  bool initialized() const { return initialized_; }
+  bool is_activated() const { return is_activated_; }
 
  private:
   scoped_ptr<crypto::SymmetricKey> key_;
   scoped_ptr<crypto::Encryptor> encryptor_;
   std::string iv_mask_;
-  bool initialized_;
+  bool is_activated_;
 
   DISALLOW_COPY_AND_ASSIGN(TransportEncryptionHandler);
 };
 
-}  // namespace transport
 }  // namespace cast
 }  // namespace media
 
-#endif  // MEDIA_CAST_TRANSPORT_TRANSPORT_UTILITY_ENCRYPTION_HANDLER_H_
+#endif  // MEDIA_CAST_COMMON_TRANSPORT_ENCRYPTION_HANDLER_H_
diff --git a/media/cast/logging/proto/BUILD.gn b/media/cast/logging/proto/BUILD.gn
new file mode 100644
index 0000000000..c9fb89bbae
--- /dev/null
+++ b/media/cast/logging/proto/BUILD.gn
@@ -0,0 +1,22 @@
+# 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.
+
+import("//third_party/protobuf/proto_library.gni")
+
+# GYP version: media/cast/cast.gyp:cast_logging_proto
+source_set("proto") {
+  sources = [
+    "proto_utils.cc",
+  ]
+  deps = [
+    ":cast_logging_proto",
+  ]
+}
+
+proto_library("cast_logging_proto") {
+  visibility = ":proto"
+  sources = [
+    "raw_events.proto",
+  ]
+}
diff --git a/media/cast/net/DEPS b/media/cast/net/DEPS
new file mode 100644
index 0000000000..3fa74fa027
--- /dev/null
+++ b/media/cast/net/DEPS
@@ -0,0 +1,16 @@
+include_rules = [
+  "-media/cast",
+  "+media/cast/cast_config.h",
+  "+media/cast/cast_defines.h",
+  "+media/cast/cast_environment.h",
+  "+media/cast/common",
+  "+media/cast/logging",
+  "+media/cast/net",
+  "+net",
+]
+
+specific_include_rules = {
+  ".*unittest.cc": [
+    "+media/cast/test",
+  ],
+}
diff --git a/media/cast/transport/cast_transport_config.cc b/media/cast/net/cast_transport_config.cc
index 16e9034713..ae8ec60551 100644
--- a/media/cast/transport/cast_transport_config.cc
+++ b/media/cast/net/cast_transport_config.cc
@@ -2,37 +2,16 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-#include "media/cast/transport/cast_transport_config.h"
+#include "media/cast/net/cast_transport_config.h"
 
 namespace media {
 namespace cast {
-namespace transport {
-
-namespace {
-const int kDefaultRtpMaxDelayMs = 100;
-}  // namespace
-
-RtpConfig::RtpConfig()
-    : ssrc(0),
-      max_delay_ms(kDefaultRtpMaxDelayMs),
-      payload_type(0) {}
-
-RtpConfig::~RtpConfig() {}
 
 CastTransportRtpConfig::CastTransportRtpConfig()
-    : max_outstanding_frames(-1) {}
+    : ssrc(0), rtp_payload_type(0), stored_frames(0) {}
 
 CastTransportRtpConfig::~CastTransportRtpConfig() {}
 
-CastTransportAudioConfig::CastTransportAudioConfig()
-    : codec(kOpus), frequency(0), channels(0) {}
-
-CastTransportAudioConfig::~CastTransportAudioConfig() {}
-
-CastTransportVideoConfig::CastTransportVideoConfig() : codec(kVp8) {}
-
-CastTransportVideoConfig::~CastTransportVideoConfig() {}
-
 EncodedFrame::EncodedFrame()
     : dependency(UNKNOWN_DEPENDENCY),
       frame_id(0),
@@ -77,6 +56,5 @@ SendRtcpFromRtpSenderData::SendRtcpFromRtpSenderData()
     : packet_type_flags(0), sending_ssrc(0) {}
 SendRtcpFromRtpSenderData::~SendRtcpFromRtpSenderData() {}
 
-}  // namespace transport
 }  // namespace cast
 }  // namespace media
diff --git a/media/cast/transport/cast_transport_config.h b/media/cast/net/cast_transport_config.h
index 96b771acb9..db87b8c9fc 100644
--- a/media/cast/transport/cast_transport_config.h
+++ b/media/cast/net/cast_transport_config.h
@@ -2,8 +2,8 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-#ifndef MEDIA_CAST_TRANSPORT_CAST_TRANSPORT_CONFIG_H_
-#define MEDIA_CAST_TRANSPORT_CAST_TRANSPORT_CONFIG_H_
+#ifndef MEDIA_CAST_NET_CAST_TRANSPORT_CONFIG_H_
+#define MEDIA_CAST_NET_CAST_TRANSPORT_CONFIG_H_
 
 #include <string>
 #include <vector>
@@ -13,66 +13,45 @@
 #include "base/memory/linked_ptr.h"
 #include "base/memory/ref_counted.h"
 #include "base/stl_util.h"
-#include "media/cast/transport/cast_transport_defines.h"
-#include "net/base/ip_endpoint.h"
+#include "media/cast/net/cast_transport_defines.h"
 
 namespace media {
 namespace cast {
-namespace transport {
 
 enum RtcpMode {
   kRtcpCompound,     // Compound RTCP mode is described by RFC 4585.
   kRtcpReducedSize,  // Reduced-size RTCP mode is described by RFC 5506.
 };
 
-enum VideoCodec {
-  kUnknownVideoCodec,
-  kFakeSoftwareVideo,
-  kVp8,
-  kH264,
-  kVideoCodecLast = kH264
-};
-
-enum AudioCodec {
-  kUnknownAudioCodec,
-  kOpus,
-  kPcm16,
-  kAudioCodecLast = kPcm16
-};
-
-struct RtpConfig {
-  RtpConfig();
-  ~RtpConfig();
-  uint32 ssrc;
-  int max_delay_ms;
-  int payload_type;
-  std::string aes_key;      // Binary string of size kAesKeySize.
-  std::string aes_iv_mask;  // Binary string of size kAesBlockSize.
+enum Codec {
+  CODEC_UNKNOWN,
+  CODEC_AUDIO_OPUS,
+  CODEC_AUDIO_PCM16,
+  CODEC_VIDEO_FAKE,
+  CODEC_VIDEO_VP8,
+  CODEC_VIDEO_H264,
+  CODEC_LAST = CODEC_VIDEO_H264
 };
 
 struct CastTransportRtpConfig {
   CastTransportRtpConfig();
   ~CastTransportRtpConfig();
-  RtpConfig config;
-  int max_outstanding_frames;
-};
 
-struct CastTransportAudioConfig {
-  CastTransportAudioConfig();
-  ~CastTransportAudioConfig();
+  // Identifier refering to this sender.
+  uint32 ssrc;
 
-  CastTransportRtpConfig rtp;
-  AudioCodec codec;
-  int frequency;
-  int channels;
-};
+  // RTP payload type enum: Specifies the type/encoding of frame data.
+  int rtp_payload_type;
 
-struct CastTransportVideoConfig {
-  CastTransportVideoConfig();
-  ~CastTransportVideoConfig();
+  // The number of most-recent frames that must be stored in the transport
+  // layer, to facilitate re-transmissions.
+  int stored_frames;
 
-  CastTransportRtpConfig rtp;
-  VideoCodec codec;
+  // The AES crypto key and initialization vector.  Each of these strings
+  // contains the data in binary form, of size kAesKeySize.  If they are empty
+  // strings, crypto is not being used.
+  std::string aes_key;
+  std::string aes_iv_mask;
 };
 
 // A combination of metadata and data for one encoded frame.  This can contain
@@ -214,8 +193,7 @@ inline bool operator==(RtcpSenderInfo lhs, RtcpSenderInfo rhs) {
          lhs.send_octet_count == rhs.send_octet_count;
 }
 
-}  // namespace transport
 }  // namespace cast
 }  // namespace media
 
-#endif  // MEDIA_CAST_TRANSPORT_CAST_TRANSPORT_CONFIG_H_
+#endif  // MEDIA_CAST_NET_CAST_TRANSPORT_CONFIG_H_
diff --git a/media/cast/transport/cast_transport_defines.h b/media/cast/net/cast_transport_defines.h
index a34f7c539a..b2020f33af 100644
--- a/media/cast/transport/cast_transport_defines.h
+++ b/media/cast/net/cast_transport_defines.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_TRANSPORT_CAST_TRANSPORT_DEFINES_H_
-#define MEDIA_CAST_TRANSPORT_CAST_TRANSPORT_DEFINES_H_
+#ifndef MEDIA_CAST_NET_CAST_TRANSPORT_DEFINES_H_
+#define MEDIA_CAST_NET_CAST_TRANSPORT_DEFINES_H_
 
 #include <stdint.h>
 
@@ -16,7 +16,6 @@
 
 namespace media {
 namespace cast {
-namespace transport {
 
 // TODO(mikhal): Implement and add more types.
 enum CastTransportStatus {
@@ -35,26 +34,6 @@ typedef std::set<uint16> PacketIdSet;
 // Each uint8 represents one cast frame.
 typedef std::map<uint8, PacketIdSet> MissingFramesAndPacketsMap;
 
-// Crypto.
-const size_t kAesBlockSize = 16;
-const size_t kAesKeySize = 16;
-
-inline std::string GetAesNonce(uint32 frame_id, const std::string& iv_mask) {
-  std::string aes_nonce(kAesBlockSize, 0);
-
-  // Serializing frame_id in big-endian order (aes_nonce[8] is the most
-  // significant byte of frame_id).
-  aes_nonce[11] = frame_id & 0xff;
-  aes_nonce[10] = (frame_id >> 8) & 0xff;
-  aes_nonce[9] = (frame_id >> 16) & 0xff;
-  aes_nonce[8] = (frame_id >> 24) & 0xff;
-
-  for (size_t i = 0; i < kAesBlockSize; ++i) {
-    aes_nonce[i] ^= iv_mask[i];
-  }
-  return aes_nonce;
-}
-
 // Rtcp defines.
 
 enum RtcpPacketFields {
@@ -162,8 +141,7 @@ inline uint32 GetVideoRtpTimestamp(const base::TimeTicks& time_ticks) {
   return static_cast<uint32>(recorded_delta.InMilliseconds() * 90);
 }
 
-}  // namespace transport
 }  // namespace cast
 }  // namespace media
 
-#endif  // MEDIA_CAST_TRANSPORT_CAST_TRANSPORT_DEFINES_H_
+#endif  // MEDIA_CAST_NET_CAST_TRANSPORT_DEFINES_H_
diff --git a/media/cast/transport/cast_transport_sender.h b/media/cast/net/cast_transport_sender.h
index e88f2f4f09..3f8a119655 100644
--- a/media/cast/transport/cast_transport_sender.h
+++ b/media/cast/net/cast_transport_sender.h
@@ -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.
 
@@ -20,8 +20,8 @@
 // This also works when the CastSender acts as a receiver for the RTCP packets
 // due to the weak pointers in the ReceivedPacket method in cast_sender_impl.cc.
 
-#ifndef MEDIA_CAST_TRANSPORT_CAST_TRANSPORT_SENDER_H_
-#define MEDIA_CAST_TRANSPORT_CAST_TRANSPORT_SENDER_H_
+#ifndef MEDIA_CAST_NET_CAST_TRANSPORT_SENDER_H_
+#define MEDIA_CAST_NET_CAST_TRANSPORT_SENDER_H_
 
 #include "base/basictypes.h"
 #include "base/callback.h"
@@ -29,8 +29,9 @@
 #include "base/threading/non_thread_safe.h"
 #include "base/time/tick_clock.h"
 #include "media/cast/logging/logging_defines.h"
-#include "media/cast/transport/cast_transport_config.h"
-#include "media/cast/transport/cast_transport_defines.h"
+#include "media/cast/net/cast_transport_config.h"
+#include "media/cast/net/cast_transport_defines.h"
+#include "net/base/ip_endpoint.h"
 
 namespace net {
 class NetLog;
@@ -38,7 +39,6 @@ class NetLog;
 
 namespace media {
 namespace cast {
-namespace transport {
 
 // Following the initialization of either audio or video an initialization
 // status will be sent via this callback.
@@ -65,9 +65,8 @@ class CastTransportSender : public base::NonThreadSafe {
   // Audio/Video initialization.
   // Encoded frames cannot be transmitted until the relevant initialize method
   // is called. Usually called by CastSender.
-  virtual void InitializeAudio(const CastTransportAudioConfig& config) = 0;
-
-  virtual void InitializeVideo(const CastTransportVideoConfig& config) = 0;
+  virtual void InitializeAudio(const CastTransportRtpConfig& config) = 0;
+  virtual void InitializeVideo(const CastTransportRtpConfig& config) = 0;
 
   // Sets the Cast packet receiver. Should be called after creation on the
   // Cast sender. Packets won't be received until this function is called.
@@ -106,8 +105,7 @@ class CastTransportSender : public base::NonThreadSafe {
       base::TimeDelta dedupe_window) = 0;
 };
 
-}  // namespace transport
 }  // namespace cast
 }  // namespace media
 
-#endif  // MEDIA_CAST_TRANSPORT_CAST_TRANSPORT_SENDER_H_
+#endif  // MEDIA_CAST_NET_CAST_TRANSPORT_SENDER_H_
diff --git a/media/cast/transport/cast_transport_sender_impl.cc b/media/cast/net/cast_transport_sender_impl.cc
index 6fd848f27b..973f34155f 100644
--- a/media/cast/transport/cast_transport_sender_impl.cc
+++ b/media/cast/net/cast_transport_sender_impl.cc
@@ -2,16 +2,16 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-#include "media/cast/transport/cast_transport_sender_impl.h"
+#include "media/cast/net/cast_transport_sender_impl.h"
 
 #include "base/single_thread_task_runner.h"
-#include "media/cast/transport/cast_transport_config.h"
-#include "media/cast/transport/cast_transport_defines.h"
+#include "media/cast/net/cast_transport_config.h"
+#include "media/cast/net/cast_transport_defines.h"
+#include "media/cast/net/udp_transport.h"
 #include "net/base/net_util.h"
 
 namespace media {
 namespace cast {
-namespace transport {
 
 scoped_ptr<CastTransportSender> CastTransportSender::Create(
     net::NetLog* net_log,
@@ -80,18 +80,18 @@ CastTransportSenderImpl::~CastTransportSenderImpl() {
 }
 
 void CastTransportSenderImpl::InitializeAudio(
-    const CastTransportAudioConfig& config) {
-  LOG_IF(WARNING, config.rtp.config.aes_key.empty() ||
-                  config.rtp.config.aes_iv_mask.empty())
+    const CastTransportRtpConfig& config) {
+  LOG_IF(WARNING, config.aes_key.empty() || config.aes_iv_mask.empty())
       << "Unsafe to send audio with encryption DISABLED.";
-  if (!audio_encryptor_.Initialize(config.rtp.config.aes_key,
-                                   config.rtp.config.aes_iv_mask)) {
+  if (!audio_encryptor_.Initialize(config.aes_key, config.aes_iv_mask)) {
     status_callback_.Run(TRANSPORT_AUDIO_UNINITIALIZED);
     return;
   }
   audio_sender_.reset(new RtpSender(clock_, transport_task_runner_, &pacer_));
-  if (audio_sender_->InitializeAudio(config)) {
-    pacer_.RegisterAudioSsrc(config.rtp.config.ssrc);
+  if (audio_sender_->Initialize(config)) {
+    // Audio packets have a higher priority.
+    pacer_.RegisterAudioSsrc(config.ssrc);
+    pacer_.RegisterPrioritySsrc(config.ssrc);
     status_callback_.Run(TRANSPORT_AUDIO_INITIALIZED);
   } else {
     audio_sender_.reset();
@@ -100,18 +100,16 @@ void CastTransportSenderImpl::InitializeAudio(
 }
 
 void CastTransportSenderImpl::InitializeVideo(
-    const CastTransportVideoConfig& config) {
-  LOG_IF(WARNING, config.rtp.config.aes_key.empty() ||
-                  config.rtp.config.aes_iv_mask.empty())
+    const CastTransportRtpConfig& config) {
+  LOG_IF(WARNING, config.aes_key.empty() || config.aes_iv_mask.empty())
       << "Unsafe to send video with encryption DISABLED.";
-  if (!video_encryptor_.Initialize(config.rtp.config.aes_key,
-                                   config.rtp.config.aes_iv_mask)) {
+  if (!video_encryptor_.Initialize(config.aes_key, config.aes_iv_mask)) {
     status_callback_.Run(TRANSPORT_VIDEO_UNINITIALIZED);
     return;
   }
   video_sender_.reset(new RtpSender(clock_, transport_task_runner_, &pacer_));
-  if (video_sender_->InitializeVideo(config)) {
-    pacer_.RegisterVideoSsrc(config.rtp.config.ssrc);
+  if (video_sender_->Initialize(config)) {
+    pacer_.RegisterVideoSsrc(config.ssrc);
     status_callback_.Run(TRANSPORT_VIDEO_INITIALIZED);
   } else {
     video_sender_.reset();
@@ -128,7 +126,7 @@ namespace {
 void EncryptAndSendFrame(const EncodedFrame& frame,
                          TransportEncryptionHandler* encryptor,
                          RtpSender* sender) {
-  if (encryptor->initialized()) {
+  if (encryptor->is_activated()) {
     EncodedFrame encrypted_frame;
     frame.CopyMetadataTo(&encrypted_frame);
     if (encryptor->Encrypt(frame.frame_id, frame.data, &encrypted_frame.data)) {
@@ -207,6 +205,5 @@ void CastTransportSenderImpl::SendRawEvents() {
   raw_events_callback_.Run(packet_events);
 }
 
-}  // namespace transport
 }  // namespace cast
 }  // namespace media
diff --git a/media/cast/transport/cast_transport_sender_impl.h b/media/cast/net/cast_transport_sender_impl.h
index 035ef844b6..5d34fdeb01 100644
--- a/media/cast/transport/cast_transport_sender_impl.h
+++ b/media/cast/net/cast_transport_sender_impl.h
@@ -2,8 +2,8 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-#ifndef MEDIA_CAST_TRANSPORT_CAST_TRANSPORT_IMPL_H_
-#define MEDIA_CAST_TRANSPORT_CAST_TRANSPORT_IMPL_H_
+#ifndef MEDIA_CAST_NET_CAST_TRANSPORT_IMPL_H_
+#define MEDIA_CAST_NET_CAST_TRANSPORT_IMPL_H_
 
 #include "base/callback.h"
 #include "base/memory/ref_counted.h"
@@ -11,18 +11,19 @@
 #include "base/time/tick_clock.h"
 #include "base/time/time.h"
 #include "base/timer/timer.h"
+#include "media/cast/common/transport_encryption_handler.h"
 #include "media/cast/logging/logging_defines.h"
 #include "media/cast/logging/simple_event_subscriber.h"
-#include "media/cast/transport/cast_transport_config.h"
-#include "media/cast/transport/cast_transport_sender.h"
-#include "media/cast/transport/pacing/paced_sender.h"
-#include "media/cast/transport/rtcp/rtcp_builder.h"
-#include "media/cast/transport/rtp_sender/rtp_sender.h"
-#include "media/cast/transport/utility/transport_encryption_handler.h"
+#include "media/cast/net/cast_transport_config.h"
+#include "media/cast/net/cast_transport_sender.h"
+#include "media/cast/net/pacing/paced_sender.h"
+#include "media/cast/net/rtcp/rtcp_builder.h"
+#include "media/cast/net/rtp/rtp_sender.h"
 
 namespace media {
 namespace cast {
-namespace transport {
+
+class UdpTransport;
 
 class CastTransportSenderImpl : public CastTransportSender {
  public:
@@ -46,9 +47,8 @@ class CastTransportSenderImpl : public CastTransportSender {
 
   virtual ~CastTransportSenderImpl();
 
-  virtual void InitializeAudio(const CastTransportAudioConfig& config) OVERRIDE;
-
-  virtual void InitializeVideo(const CastTransportVideoConfig& config) OVERRIDE;
+  virtual void InitializeAudio(const CastTransportRtpConfig& config) OVERRIDE;
+  virtual void InitializeVideo(const CastTransportRtpConfig& config) OVERRIDE;
 
   // CastTransportSender implementation.
   virtual void SetPacketReceiver(const PacketReceiverCallback& packet_receiver)
@@ -103,8 +103,7 @@ class CastTransportSenderImpl : public CastTransportSender {
   DISALLOW_COPY_AND_ASSIGN(CastTransportSenderImpl);
 };
 
-}  // namespace transport
 }  // namespace cast
 }  // namespace media
 
-#endif  // MEDIA_CAST_TRANSPORT_CAST_TRANSPORT_IMPL_H_
+#endif  // MEDIA_CAST_NET_CAST_TRANSPORT_IMPL_H_
diff --git a/media/cast/transport/cast_transport_sender_impl_unittest.cc b/media/cast/net/cast_transport_sender_impl_unittest.cc
index 67eb39a47a..b1d29b9ef9 100644
--- a/media/cast/transport/cast_transport_sender_impl_unittest.cc
+++ b/media/cast/net/cast_transport_sender_impl_unittest.cc
@@ -12,19 +12,18 @@
 #include "base/run_loop.h"
 #include "base/test/simple_test_tick_clock.h"
 #include "media/cast/cast_config.h"
-#include "media/cast/rtcp/rtcp.h"
+#include "media/cast/net/cast_transport_config.h"
+#include "media/cast/net/cast_transport_sender_impl.h"
+#include "media/cast/net/rtcp/rtcp.h"
 #include "media/cast/test/fake_single_thread_task_runner.h"
-#include "media/cast/transport/cast_transport_config.h"
-#include "media/cast/transport/cast_transport_sender_impl.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
 namespace media {
 namespace cast {
-namespace transport {
 
 static const int64 kStartMillisecond = INT64_C(12345678900000);
 
-class FakePacketSender : public transport::PacketSender {
+class FakePacketSender : public PacketSender {
  public:
   FakePacketSender() {}
 
@@ -78,7 +77,7 @@ class CastTransportSenderImplTest : public ::testing::Test {
     }
   }
 
-  static void UpdateCastTransportStatus(transport::CastTransportStatus status) {
+  static void UpdateCastTransportStatus(CastTransportStatus status) {
   }
 
   base::SimpleTestTickClock testing_clock_;
@@ -108,6 +107,5 @@ TEST_F(CastTransportSenderImplTest, InitWithLogging) {
   EXPECT_GT(num_times_callback_called_, 1);
 }
 
-}  // namespace transport
 }  // namespace cast
 }  // namespace media
diff --git a/media/cast/transport/frame_id_wrap_helper_test.cc b/media/cast/net/frame_id_wrap_helper_test.cc
index 3a2060d3aa..92a8443533 100644
--- a/media/cast/transport/frame_id_wrap_helper_test.cc
+++ b/media/cast/net/frame_id_wrap_helper_test.cc
@@ -1,13 +1,12 @@
-// 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 <gtest/gtest.h>
-#include "media/cast/transport/cast_transport_defines.h"
+#include "media/cast/net/cast_transport_defines.h"
 
 namespace media {
 namespace cast {
-namespace transport {
 
 class FrameIdWrapHelperTest : public ::testing::Test {
  protected:
@@ -47,6 +46,5 @@ TEST_F(FrameIdWrapHelperTest, OutOfOrder) {
   EXPECT_EQ(257u, new_frame_id);
 }
 
-}  // namespace transport
 }  // namespace cast
 }  // namespace media
diff --git a/media/cast/transport/pacing/mock_paced_packet_sender.cc b/media/cast/net/pacing/mock_paced_packet_sender.cc
index 5e325f0233..3219ba2ba4 100644
--- a/media/cast/transport/pacing/mock_paced_packet_sender.cc
+++ b/media/cast/net/pacing/mock_paced_packet_sender.cc
@@ -1,17 +1,15 @@
-// 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/transport/pacing/mock_paced_packet_sender.h"
+#include "media/cast/net/pacing/mock_paced_packet_sender.h"
 
 namespace media {
 namespace cast {
-namespace transport {
 
 MockPacedPacketSender::MockPacedPacketSender() {}
 
 MockPacedPacketSender::~MockPacedPacketSender() {}
 
-}  // namespace transport
 }  // namespace cast
 }  // namespace media
diff --git a/media/cast/transport/pacing/mock_paced_packet_sender.h b/media/cast/net/pacing/mock_paced_packet_sender.h
index 20b7647035..2f7f1b2650 100644
--- a/media/cast/transport/pacing/mock_paced_packet_sender.h
+++ b/media/cast/net/pacing/mock_paced_packet_sender.h
@@ -1,16 +1,15 @@
-// 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_TRANSPORT_PACING_MOCK_PACED_PACKET_SENDER_H_
-#define MEDIA_CAST_TRANSPORT_PACING_MOCK_PACED_PACKET_SENDER_H_
+#ifndef MEDIA_CAST_NET_PACING_MOCK_PACED_PACKET_SENDER_H_
+#define MEDIA_CAST_NET_PACING_MOCK_PACED_PACKET_SENDER_H_
 
-#include "media/cast/transport/pacing/paced_sender.h"
+#include "media/cast/net/pacing/paced_sender.h"
 #include "testing/gmock/include/gmock/gmock.h"
 
 namespace media {
 namespace cast {
-namespace transport {
 
 class MockPacedPacketSender : public PacedPacketSender {
  public:
@@ -24,8 +23,7 @@ class MockPacedPacketSender : public PacedPacketSender {
   MOCK_METHOD1(CancelSendingPacket, void(const PacketKey& packet_key));
 };
 
-}  // namespace transport
 }  // namespace cast
 }  // namespace media
 
-#endif  // MEDIA_CAST_TRANSPORT_PACING_MOCK_PACED_PACKET_SENDER_H_
+#endif  // MEDIA_CAST_NET_PACING_MOCK_PACED_PACKET_SENDER_H_
diff --git a/media/cast/transport/pacing/paced_sender.cc b/media/cast/net/pacing/paced_sender.cc
index 20cbde85be..89a69272dc 100644
--- a/media/cast/transport/pacing/paced_sender.cc
+++ b/media/cast/net/pacing/paced_sender.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/transport/pacing/paced_sender.h"
+#include "media/cast/net/pacing/paced_sender.h"
 
 #include "base/big_endian.h"
 #include "base/bind.h"
 #include "base/message_loop/message_loop.h"
+#include "media/cast/logging/logging_impl.h"
 
 namespace media {
 namespace cast {
-namespace transport {
 
 namespace {
 
@@ -60,13 +60,24 @@ void PacedSender::RegisterVideoSsrc(uint32 video_ssrc) {
   video_ssrc_ = video_ssrc;
 }
 
+void PacedSender::RegisterPrioritySsrc(uint32 ssrc) {
+  priority_ssrcs_.push_back(ssrc);
+}
+
 bool PacedSender::SendPackets(const SendPacketVector& packets) {
   if (packets.empty()) {
     return true;
   }
+  const bool high_priority = IsHighPriority(packets.begin()->first);
   for (size_t i = 0; i < packets.size(); i++) {
-    packet_list_[packets[i].first] =
-        make_pair(PacketType_Normal, packets[i].second);
+    DCHECK(IsHighPriority(packets[i].first) == high_priority);
+    if (high_priority) {
+      priority_packet_list_[packets[i].first] =
+          make_pair(PacketType_Normal, packets[i].second);
+    } else {
+      packet_list_[packets[i].first] =
+          make_pair(PacketType_Normal, packets[i].second);
+    }
   }
   if (state_ == State_Unblocked) {
     SendStoredPackets();
@@ -79,6 +90,7 @@ bool PacedSender::ResendPackets(const SendPacketVector& packets,
   if (packets.empty()) {
     return true;
   }
+  const bool high_priority = IsHighPriority(packets.begin()->first);
   base::TimeTicks now = clock_->NowTicks();
   for (size_t i = 0; i < packets.size(); i++) {
     std::map<PacketKey, base::TimeTicks>::const_iterator j =
@@ -89,8 +101,14 @@ bool PacedSender::ResendPackets(const SendPacketVector& packets,
       continue;
     }
 
-    packet_list_[packets[i].first] =
-        make_pair(PacketType_Resend, packets[i].second);
+    DCHECK(IsHighPriority(packets[i].first) == high_priority);
+    if (high_priority) {
+      priority_packet_list_[packets[i].first] =
+          make_pair(PacketType_Resend, packets[i].second);
+    } else {
+      packet_list_[packets[i].first] =
+          make_pair(PacketType_Resend, packets[i].second);
+    }
   }
   if (state_ == State_Unblocked) {
     SendStoredPackets();
@@ -100,7 +118,8 @@ bool PacedSender::ResendPackets(const SendPacketVector& packets,
 
 bool PacedSender::SendRtcpPacket(uint32 ssrc, PacketRef packet) {
   if (state_ == State_TransportBlocked) {
-    packet_list_[PacedPacketSender::MakePacketKey(base::TimeTicks(), ssrc, 0)] =
+    priority_packet_list_[
+        PacedPacketSender::MakePacketKey(base::TimeTicks(), ssrc, 0)] =
         make_pair(PacketType_RTCP, packet);
   } else {
     // We pass the RTCP packets straight through.
@@ -110,33 +129,39 @@ bool PacedSender::SendRtcpPacket(uint32 ssrc, PacketRef packet) {
                        weak_factory_.GetWeakPtr()))) {
       state_ = State_TransportBlocked;
     }
-
   }
   return true;
 }
 
 void PacedSender::CancelSendingPacket(const PacketKey& packet_key) {
   packet_list_.erase(packet_key);
+  priority_packet_list_.erase(packet_key);
 }
 
-PacketRef PacedSender::GetNextPacket(PacketType* packet_type,
+PacketRef PacedSender::PopNextPacket(PacketType* packet_type,
                                      PacketKey* packet_key) {
-  std::map<PacketKey, std::pair<PacketType, PacketRef> >::iterator i;
-  i = packet_list_.begin();
-  DCHECK(i != packet_list_.end());
+  PacketList* list = !priority_packet_list_.empty() ?
+      &priority_packet_list_ : &packet_list_;
+  DCHECK(!list->empty());
+  PacketList::iterator i = list->begin();
   *packet_type = i->second.first;
   *packet_key = i->first;
   PacketRef ret = i->second.second;
-  packet_list_.erase(i);
+  list->erase(i);
   return ret;
 }
 
+bool PacedSender::IsHighPriority(const PacketKey& packet_key) const {
+  return std::find(priority_ssrcs_.begin(), priority_ssrcs_.end(),
+                   packet_key.second.first) != priority_ssrcs_.end();
+}
+
 bool PacedSender::empty() const {
-  return packet_list_.empty();
+  return packet_list_.empty() && priority_packet_list_.empty();
 }
 
 size_t PacedSender::size() const {
-  return packet_list_.size();
+  return packet_list_.size() + priority_packet_list_.size();
 }
 
 // This function can be called from three places:
@@ -199,7 +224,7 @@ void PacedSender::SendStoredPackets() {
     }
     PacketType packet_type;
     PacketKey packet_key;
-    PacketRef packet = GetNextPacket(&packet_type, &packet_key);
+    PacketRef packet = PopNextPacket(&packet_type, &packet_key);
     sent_time_[packet_key] = now;
     sent_time_buffer_[packet_key] = now;
 
@@ -255,6 +280,5 @@ void PacedSender::LogPacketEvent(const Packet& packet, CastLoggingEvent event) {
       packet);
 }
 
-}  // namespace transport
 }  // namespace cast
 }  // namespace media
diff --git a/media/cast/transport/pacing/paced_sender.h b/media/cast/net/pacing/paced_sender.h
index 9fc0c8b8b8..094e5299e1 100644
--- a/media/cast/transport/pacing/paced_sender.h
+++ b/media/cast/net/pacing/paced_sender.h
@@ -1,11 +1,11 @@
-// 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_TRANSPORT_PACING_PACED_SENDER_H_
-#define MEDIA_CAST_TRANSPORT_PACING_PACED_SENDER_H_
+#ifndef MEDIA_CAST_NET_PACING_PACED_SENDER_H_
+#define MEDIA_CAST_NET_PACING_PACED_SENDER_H_
 
-#include <list>
+#include <map>
 #include <vector>
 
 #include "base/basictypes.h"
@@ -16,16 +16,14 @@
 #include "base/time/default_tick_clock.h"
 #include "base/time/tick_clock.h"
 #include "base/time/time.h"
-#include "media/cast/transport/cast_transport_config.h"
-#include "media/cast/transport/transport/udp_transport.h"
+#include "media/cast/logging/logging_defines.h"
+#include "media/cast/net/cast_transport_config.h"
 
 namespace media {
 namespace cast {
 
 class LoggingImpl;
 
-namespace transport {
-
 // Use std::pair for free comparison operators.
 // { capture_time, ssrc, packet_id }
 // The PacketKey is designed to meet two criteria:
@@ -71,6 +69,12 @@ class PacedSender : public PacedPacketSender,
   void RegisterAudioSsrc(uint32 audio_ssrc);
   void RegisterVideoSsrc(uint32 video_ssrc);
 
+  // Register SSRC that has a higher priority for sending. Multiple SSRCs can
+  // be registered.
+  // Note that it is not expected to register many SSRCs with this method.
+  // Because IsHigherPriority() is determined in linear time.
+  void RegisterPrioritySsrc(uint32 ssrc);
+
   // PacedPacketSender implementation.
   virtual bool SendPackets(const SendPacketVector& packets) OVERRIDE;
   virtual bool ResendPackets(const SendPacketVector& packets,
@@ -110,16 +114,25 @@ class PacedSender : public PacedPacketSender,
   // Returns the next packet to send. RTCP packets have highest priority,
   // resend packets have second highest priority and then comes everything
   // else.
-  PacketRef GetNextPacket(PacketType* packet_type,
+  PacketRef PopNextPacket(PacketType* packet_type,
                           PacketKey* packet_key);
 
+  // Returns true if the packet should have a higher priority.
+  bool IsHighPriority(const PacketKey& packet_key) const;
+
   base::TickClock* const clock_;  // Not owned by this class.
   LoggingImpl* const logging_;    // Not owned by this class.
   PacketSender* transport_;       // Not owned by this class.
   scoped_refptr<base::SingleThreadTaskRunner> transport_task_runner_;
   uint32 audio_ssrc_;
   uint32 video_ssrc_;
-  std::map<PacketKey, std::pair<PacketType, PacketRef> > packet_list_;
+
+  // Set of SSRCs that have higher priority. This is a vector instead of a
+  // set because there's only very few in it (most likely 1).
+  std::vector<uint32> priority_ssrcs_;
+  typedef std::map<PacketKey, std::pair<PacketType, PacketRef> > PacketList;
+  PacketList packet_list_;
+  PacketList priority_packet_list_;
   std::map<PacketKey, base::TimeTicks> sent_time_;
   std::map<PacketKey, base::TimeTicks> sent_time_buffer_;
 
@@ -140,8 +153,7 @@ class PacedSender : public PacedPacketSender,
   DISALLOW_COPY_AND_ASSIGN(PacedSender);
 };
 
-}  // namespace transport
 }  // namespace cast
 }  // namespace media
 
-#endif  // MEDIA_CAST_TRANSPORT_PACING_PACED_SENDER_H_
+#endif  // MEDIA_CAST_NET_PACING_PACED_SENDER_H_
diff --git a/media/cast/transport/pacing/paced_sender_unittest.cc b/media/cast/net/pacing/paced_sender_unittest.cc
index 5e24fca4b5..43e7603385 100644
--- a/media/cast/transport/pacing/paced_sender_unittest.cc
+++ b/media/cast/net/pacing/paced_sender_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.
 
@@ -6,23 +6,24 @@
 
 #include "base/big_endian.h"
 #include "base/test/simple_test_tick_clock.h"
+#include "media/cast/logging/logging_impl.h"
 #include "media/cast/logging/simple_event_subscriber.h"
+#include "media/cast/net/pacing/paced_sender.h"
 #include "media/cast/test/fake_single_thread_task_runner.h"
-#include "media/cast/transport/pacing/paced_sender.h"
 #include "testing/gmock/include/gmock/gmock.h"
 
+using testing::_;
+
 namespace media {
 namespace cast {
-namespace transport {
-
-using testing::_;
+namespace {
 
 static const uint8 kValue = 123;
-static const size_t kSize1 = 100;
-static const size_t kSize2 = 101;
-static const size_t kSize3 = 102;
-static const size_t kSize4 = 103;
-static const size_t kNackSize = 104;
+static const size_t kSize1 = 101;
+static const size_t kSize2 = 102;
+static const size_t kSize3 = 103;
+static const size_t kSize4 = 104;
+static const size_t kNackSize = 105;
 static const int64 kStartMillisecond = INT64_C(12345678900000);
 static const uint32 kVideoSsrc = 0x1234;
 static const uint32 kAudioSsrc = 0x5678;
@@ -68,7 +69,7 @@ class PacedSenderTest : public ::testing::Test {
     logging_.RemoveRawEventSubscriber(&subscriber_);
   }
 
-  static void UpdateCastTransportStatus(transport::CastTransportStatus status) {
+  static void UpdateCastTransportStatus(CastTransportStatus status) {
     NOTREACHED();
   }
 
@@ -124,6 +125,8 @@ class PacedSenderTest : public ::testing::Test {
   DISALLOW_COPY_AND_ASSIGN(PacedSenderTest);
 };
 
+}  // namespace
+
 TEST_F(PacedSenderTest, PassThroughRtcp) {
   mock_transport_.AddExpectedSize(kSize1, 2);
   SendPacketVector packets = CreateSendPacketVector(kSize1, 1, true);
@@ -346,6 +349,56 @@ TEST_F(PacedSenderTest, PaceWith60fps) {
   EXPECT_TRUE(RunUntilEmpty(5));
 }
 
-}  // namespace transport
+TEST_F(PacedSenderTest, SendPriority) {
+  // Actual order to the network is:
+  // 1. Video packets x 10.
+  // 2. RTCP packet x 1.
+  // 3. Audio packet x 1.
+  // 4. Video retransmission packet x 10.
+  // 5. Video packet x 10.
+  mock_transport_.AddExpectedSize(kSize2, 10);  // Normal video packets.
+  mock_transport_.AddExpectedSize(kSize3, 1);  // RTCP packet.
+  mock_transport_.AddExpectedSize(kSize1, 1);  // Audio packet.
+  mock_transport_.AddExpectedSize(kSize4, 10);  // Resend video packets.
+  mock_transport_.AddExpectedSize(kSize2, 10);  // Normal video packets.
+
+  paced_sender_->RegisterPrioritySsrc(kAudioSsrc);
+
+  // Retransmission packets with the earlier timestamp.
+  SendPacketVector resend_packets =
+      CreateSendPacketVector(kSize4, 10, false);
+  testing_clock_.Advance(base::TimeDelta::FromMilliseconds(10));
+
+  // Send 20 normal video packets. Only 10 will be sent in this
+  // call, the rest will be sitting in the queue waiting for pacing.
+  EXPECT_TRUE(paced_sender_->SendPackets(
+      CreateSendPacketVector(kSize2, 20, false)));
+
+  testing_clock_.Advance(base::TimeDelta::FromMilliseconds(10));
+
+  // Send normal audio packet. This is queued and will be sent
+  // earlier than video packets.
+  EXPECT_TRUE(paced_sender_->SendPackets(
+      CreateSendPacketVector(kSize1, 1, true)));
+
+  // Send RTCP packet. This is queued and will be sent first.
+  EXPECT_TRUE(paced_sender_->SendRtcpPacket(
+      kVideoSsrc,
+      new base::RefCountedData<Packet>(Packet(kSize3, kValue))));
+
+  // Resend video packets. This is queued and will be sent
+  // earlier than normal video packets.
+  EXPECT_TRUE(paced_sender_->ResendPackets(
+      resend_packets, base::TimeDelta()));
+
+  // Roll the clock. Queued packets will be sent in this order:
+  // 1. RTCP packet x 1.
+  // 2. Audio packet x 1.
+  // 3. Video retransmission packet x 10.
+  // 4. Video packet x 10.
+  task_runner_->RunTasks();
+  EXPECT_TRUE(RunUntilEmpty(4));
+}
+
 }  // namespace cast
 }  // namespace media
diff --git a/media/cast/rtcp/mock_rtcp_receiver_feedback.cc b/media/cast/net/rtcp/mock_rtcp_receiver_feedback.cc
index 9ff2d48f03..0ea6606f82 100644
--- a/media/cast/rtcp/mock_rtcp_receiver_feedback.cc
+++ b/media/cast/net/rtcp/mock_rtcp_receiver_feedback.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/rtcp/mock_rtcp_receiver_feedback.h"
+#include "media/cast/net/rtcp/mock_rtcp_receiver_feedback.h"
 
 namespace media {
 namespace cast {
diff --git a/media/cast/rtcp/mock_rtcp_receiver_feedback.h b/media/cast/net/rtcp/mock_rtcp_receiver_feedback.h
index 56fe1ca699..38bf0eec28 100644
--- a/media/cast/rtcp/mock_rtcp_receiver_feedback.h
+++ b/media/cast/net/rtcp/mock_rtcp_receiver_feedback.h
@@ -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.
 
@@ -7,9 +7,9 @@
 
 #include <vector>
 
-#include "media/cast/rtcp/rtcp_defines.h"
-#include "media/cast/rtcp/rtcp_receiver.h"
-#include "media/cast/transport/cast_transport_defines.h"
+#include "media/cast/net/cast_transport_defines.h"
+#include "media/cast/net/rtcp/rtcp_defines.h"
+#include "media/cast/net/rtcp/rtcp_receiver.h"
 #include "testing/gmock/include/gmock/gmock.h"
 
 namespace media {
@@ -21,7 +21,7 @@ class MockRtcpReceiverFeedback : public RtcpReceiverFeedback {
   virtual ~MockRtcpReceiverFeedback();
 
   MOCK_METHOD1(OnReceivedSenderReport,
-               void(const transport::RtcpSenderInfo& remote_sender_info));
+               void(const RtcpSenderInfo& remote_sender_info));
 
   MOCK_METHOD1(OnReceiverReferenceTimeReport,
                void(const RtcpReceiverReferenceTimeReport& remote_time_report));
diff --git a/media/cast/rtcp/mock_rtcp_sender_feedback.cc b/media/cast/net/rtcp/mock_rtcp_sender_feedback.cc
index e44e0bfdef..2c51c7448d 100644
--- a/media/cast/rtcp/mock_rtcp_sender_feedback.cc
+++ b/media/cast/net/rtcp/mock_rtcp_sender_feedback.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/rtcp/mock_rtcp_sender_feedback.h"
+#include "media/cast/net/rtcp/mock_rtcp_sender_feedback.h"
 
 namespace media {
 namespace cast {
diff --git a/media/cast/rtcp/mock_rtcp_sender_feedback.h b/media/cast/net/rtcp/mock_rtcp_sender_feedback.h
index 40547e6283..a6af0aaa3e 100644
--- a/media/cast/rtcp/mock_rtcp_sender_feedback.h
+++ b/media/cast/net/rtcp/mock_rtcp_sender_feedback.h
@@ -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.
 
@@ -7,7 +7,7 @@
 
 #include <vector>
 
-#include "media/cast/rtcp/rtcp_receiver.h"
+#include "media/cast/net/rtcp/rtcp_receiver.h"
 #include "testing/gmock/include/gmock/gmock.h"
 
 namespace media {
diff --git a/media/cast/rtcp/receiver_rtcp_event_subscriber.cc b/media/cast/net/rtcp/receiver_rtcp_event_subscriber.cc
index 9a9c0aeeb7..a751ff94d3 100644
--- a/media/cast/rtcp/receiver_rtcp_event_subscriber.cc
+++ b/media/cast/net/rtcp/receiver_rtcp_event_subscriber.cc
@@ -2,7 +2,7 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-#include "media/cast/rtcp/receiver_rtcp_event_subscriber.h"
+#include "media/cast/net/rtcp/receiver_rtcp_event_subscriber.h"
 
 #include <utility>
 
diff --git a/media/cast/rtcp/receiver_rtcp_event_subscriber.h b/media/cast/net/rtcp/receiver_rtcp_event_subscriber.h
index 84af7cbaf3..7e80ffebc6 100644
--- a/media/cast/rtcp/receiver_rtcp_event_subscriber.h
+++ b/media/cast/net/rtcp/receiver_rtcp_event_subscriber.h
@@ -10,7 +10,7 @@
 #include "base/threading/thread_checker.h"
 #include "media/cast/logging/logging_defines.h"
 #include "media/cast/logging/raw_event_subscriber.h"
-#include "media/cast/rtcp/rtcp_defines.h"
+#include "media/cast/net/rtcp/rtcp_defines.h"
 
 namespace media {
 namespace cast {
diff --git a/media/cast/rtcp/receiver_rtcp_event_subscriber_unittest.cc b/media/cast/net/rtcp/receiver_rtcp_event_subscriber_unittest.cc
index e0d0f17216..8d975592f2 100644
--- a/media/cast/rtcp/receiver_rtcp_event_subscriber_unittest.cc
+++ b/media/cast/net/rtcp/receiver_rtcp_event_subscriber_unittest.cc
@@ -8,7 +8,7 @@
 #include "base/time/tick_clock.h"
 #include "media/cast/cast_environment.h"
 #include "media/cast/logging/logging_defines.h"
-#include "media/cast/rtcp/receiver_rtcp_event_subscriber.h"
+#include "media/cast/net/rtcp/receiver_rtcp_event_subscriber.h"
 #include "media/cast/test/fake_single_thread_task_runner.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
diff --git a/media/cast/rtcp/rtcp.cc b/media/cast/net/rtcp/rtcp.cc
index 480b2ac399..c0e6c19387 100644
--- a/media/cast/rtcp/rtcp.cc
+++ b/media/cast/net/rtcp/rtcp.cc
@@ -1,25 +1,26 @@
-// 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/rtcp/rtcp.h"
+#include "media/cast/net/rtcp/rtcp.h"
 
 #include "base/big_endian.h"
-#include "base/rand_util.h"
 #include "media/cast/cast_config.h"
 #include "media/cast/cast_defines.h"
 #include "media/cast/cast_environment.h"
-#include "media/cast/rtcp/rtcp_defines.h"
-#include "media/cast/rtcp/rtcp_receiver.h"
-#include "media/cast/rtcp/rtcp_sender.h"
-#include "media/cast/rtcp/rtcp_utility.h"
-#include "media/cast/transport/cast_transport_defines.h"
+#include "media/cast/net/cast_transport_defines.h"
+#include "media/cast/net/rtcp/rtcp_defines.h"
+#include "media/cast/net/rtcp/rtcp_receiver.h"
+#include "media/cast/net/rtcp/rtcp_sender.h"
+#include "media/cast/net/rtcp/rtcp_utility.h"
+
+using base::TimeDelta;
 
 namespace media {
 namespace cast {
 
-static const int kMaxRttMs = 10000;  // 10 seconds.
-static const int kMaxDelay = 2000;
+static const int32 kMaxRttMs = 10000;  // 10 seconds.
+static const int32 kMaxDelayMs = 2000;  // 2 seconds.
 
 class LocalRtcpRttFeedback : public RtcpRttFeedback {
  public:
@@ -43,7 +44,7 @@ class LocalRtcpReceiverFeedback : public RtcpReceiverFeedback {
       : rtcp_(rtcp), cast_environment_(cast_environment) {}
 
   virtual void OnReceivedSenderReport(
-      const transport::RtcpSenderInfo& remote_sender_info) OVERRIDE {
+      const RtcpSenderInfo& remote_sender_info) OVERRIDE {
     rtcp_->OnReceivedNtp(remote_sender_info.ntp_seconds,
                          remote_sender_info.ntp_fraction);
     if (remote_sender_info.send_packet_count != 0) {
@@ -75,8 +76,8 @@ class LocalRtcpReceiverFeedback : public RtcpReceiverFeedback {
 
 Rtcp::Rtcp(scoped_refptr<CastEnvironment> cast_environment,
            RtcpSenderFeedback* sender_feedback,
-           transport::CastTransportSender* const transport_sender,
-           transport::PacedPacketSender* paced_packet_sender,
+           CastTransportSender* const transport_sender,
+           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,
@@ -98,7 +99,7 @@ Rtcp::Rtcp(scoped_refptr<CastEnvironment> cast_environment,
       local_clock_ahead_by_(ClockDriftSmoother::GetDefaultTimeConstant()),
       lip_sync_rtp_timestamp_(0),
       lip_sync_ntp_timestamp_(0),
-      min_rtt_(base::TimeDelta::FromMilliseconds(kMaxRttMs)),
+      min_rtt_(TimeDelta::FromMilliseconds(kMaxRttMs)),
       number_of_rtt_in_avg_(0) {
   rtcp_receiver_.reset(new RtcpReceiver(cast_environment, sender_feedback,
                                         receiver_feedback_.get(),
@@ -114,8 +115,8 @@ bool Rtcp::IsRtcpPacket(const uint8* packet, size_t length) {
   if (length < kMinLengthOfRtcp) return false;
 
   uint8 packet_type = packet[1];
-  if (packet_type >= transport::kPacketTypeLow &&
-      packet_type <= transport::kPacketTypeHigh) {
+  if (packet_type >= kPacketTypeLow &&
+      packet_type <= kPacketTypeHigh) {
     return true;
   }
   return false;
@@ -156,23 +157,23 @@ void Rtcp::SendRtcpFromRtpReceiver(
   uint32 packet_type_flags = 0;
 
   base::TimeTicks now = cast_environment_->Clock()->NowTicks();
-  transport::RtcpReportBlock report_block;
+  RtcpReportBlock report_block;
   RtcpReceiverReferenceTimeReport rrtr;
 
   // Attach our NTP to all RTCP packets; with this information a "smart" sender
   // can make decisions based on how old the RTCP message is.
-  packet_type_flags |= transport::kRtcpRrtr;
+  packet_type_flags |= kRtcpRrtr;
   ConvertTimeTicksToNtp(now, &rrtr.ntp_seconds, &rrtr.ntp_fraction);
   SaveLastSentNtpTime(now, rrtr.ntp_seconds, rrtr.ntp_fraction);
 
   if (cast_message) {
-    packet_type_flags |= transport::kRtcpCast;
+    packet_type_flags |= kRtcpCast;
   }
   if (rtcp_events) {
-    packet_type_flags |= transport::kRtcpReceiverLog;
+    packet_type_flags |= kRtcpReceiverLog;
   }
   if (rtcp_mode_ == kRtcpCompound || now >= next_time_to_send_rtcp_) {
-    packet_type_flags |= transport::kRtcpRr;
+    packet_type_flags |= kRtcpRr;
 
     report_block.remote_ssrc = 0;            // Not needed to set send side.
     report_block.media_ssrc = remote_ssrc_;  // SSRC of the RTP packet sender.
@@ -201,13 +202,13 @@ void Rtcp::SendRtcpFromRtpReceiver(
                                         &rrtr,
                                         cast_message,
                                         rtcp_events,
-                                        target_delay_ms_);
+                                        target_delay_);
 }
 
 void Rtcp::SendRtcpFromRtpSender(base::TimeTicks current_time,
                                  uint32 current_time_as_rtp_timestamp) {
   DCHECK(transport_sender_);
-  uint32 packet_type_flags = transport::kRtcpSr;
+  uint32 packet_type_flags = kRtcpSr;
   uint32 current_ntp_seconds = 0;
   uint32 current_ntp_fractions = 0;
   ConvertTimeTicksToNtp(current_time, &current_ntp_seconds,
@@ -215,9 +216,9 @@ void Rtcp::SendRtcpFromRtpSender(base::TimeTicks current_time,
   SaveLastSentNtpTime(current_time, current_ntp_seconds,
                       current_ntp_fractions);
 
-  transport::RtcpDlrrReportBlock dlrr;
+  RtcpDlrrReportBlock dlrr;
   if (!time_last_report_received_.is_null()) {
-    packet_type_flags |= transport::kRtcpDlrr;
+    packet_type_flags |= kRtcpDlrr;
     dlrr.last_rr = last_report_truncated_ntp_;
     uint32 delay_seconds = 0;
     uint32 delay_fraction = 0;
@@ -302,8 +303,8 @@ void Rtcp::SetCastReceiverEventHistorySize(size_t size) {
 }
 
 void Rtcp::SetTargetDelay(base::TimeDelta target_delay) {
-  DCHECK(target_delay.InMilliseconds() < kMaxDelay);
-  target_delay_ms_ = static_cast<uint16>(target_delay.InMilliseconds());
+  DCHECK(target_delay < TimeDelta::FromMilliseconds(kMaxDelayMs));
+  target_delay_ = target_delay;
 }
 
 void Rtcp::OnReceivedDelaySinceLastReport(uint32 receivers_ssrc,
@@ -331,7 +332,7 @@ void Rtcp::SaveLastSentNtpTime(const base::TimeTicks& now,
   last_reports_sent_map_[last_report] = now;
   last_reports_sent_queue_.push(std::make_pair(last_report, now));
 
-  base::TimeTicks timeout = now - base::TimeDelta::FromMilliseconds(kMaxRttMs);
+  base::TimeTicks timeout = now - TimeDelta::FromMilliseconds(kMaxRttMs);
 
   // Cleanup old statistics older than |timeout|.
   while (!last_reports_sent_queue_.empty()) {
@@ -358,11 +359,12 @@ void Rtcp::UpdateRtt(const base::TimeDelta& sender_delay,
   // TODO(miu): Replace "average for all time" with an EWMA, or suitable
   // "average over recent past" mechanism.
   if (number_of_rtt_in_avg_ != 0) {
-    const double ac = static_cast<double>(number_of_rtt_in_avg_);
-    avg_rtt_ms_ = ((ac / (ac + 1.0)) * avg_rtt_ms_) +
-                  ((1.0 / (ac + 1.0)) * rtt.InMillisecondsF());
+    // Integer math equivalent of (ac/(ac+1.0))*avg_rtt_ + (1.0/(ac+1.0))*rtt).
+    // (TimeDelta only supports math with other TimeDeltas and int64s.)
+    avg_rtt_ = (avg_rtt_ * number_of_rtt_in_avg_ + rtt) /
+        (number_of_rtt_in_avg_ + 1);
   } else {
-    avg_rtt_ms_ = rtt.InMillisecondsF();
+    avg_rtt_ = rtt;
   }
   number_of_rtt_in_avg_++;
 }
@@ -377,19 +379,15 @@ bool Rtcp::Rtt(base::TimeDelta* rtt, base::TimeDelta* avg_rtt,
   if (number_of_rtt_in_avg_ == 0) return false;
 
   *rtt = rtt_;
-  *avg_rtt = base::TimeDelta::FromMillisecondsD(avg_rtt_ms_);
+  *avg_rtt = avg_rtt_;
   *min_rtt = min_rtt_;
   *max_rtt = max_rtt_;
   return true;
 }
 
 void Rtcp::UpdateNextTimeToSendRtcp() {
-  int random = base::RandInt(0, 999);
-  base::TimeDelta time_to_next =
-      (rtcp_interval_ / 2) + (rtcp_interval_ * random / 1000);
-
   base::TimeTicks now = cast_environment_->Clock()->NowTicks();
-  next_time_to_send_rtcp_ = now + time_to_next;
+  next_time_to_send_rtcp_ = now + rtcp_interval_;
 }
 
 void Rtcp::OnReceivedReceiverLog(const RtcpReceiverLogMessage& receiver_log) {
diff --git a/media/cast/rtcp/rtcp.h b/media/cast/net/rtcp/rtcp.h
index 9d0184f903..2bf0367185 100644
--- a/media/cast/rtcp/rtcp.h
+++ b/media/cast/net/rtcp/rtcp.h
@@ -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.
 
@@ -13,15 +13,15 @@
 #include "base/memory/scoped_ptr.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_defines.h"
 #include "media/cast/cast_environment.h"
-#include "media/cast/rtcp/receiver_rtcp_event_subscriber.h"
-#include "media/cast/rtcp/rtcp_defines.h"
-#include "media/cast/transport/cast_transport_defines.h"
-#include "media/cast/transport/cast_transport_sender.h"
-#include "media/cast/transport/pacing/paced_sender.h"
+#include "media/cast/common/clock_drift_smoother.h"
+#include "media/cast/net/cast_transport_defines.h"
+#include "media/cast/net/cast_transport_sender.h"
+#include "media/cast/net/pacing/paced_sender.h"
+#include "media/cast/net/rtcp/receiver_rtcp_event_subscriber.h"
+#include "media/cast/net/rtcp/rtcp_defines.h"
 
 namespace media {
 namespace cast {
@@ -60,8 +60,8 @@ class Rtcp {
   // be used by the Cast receivers and test applications.
   Rtcp(scoped_refptr<CastEnvironment> cast_environment,
        RtcpSenderFeedback* sender_feedback,
-       transport::CastTransportSender* const transport_sender,  // Send-side.
-       transport::PacedPacketSender* paced_packet_sender,       // Receive side.
+       CastTransportSender* const transport_sender,  // Send-side.
+       PacedPacketSender* paced_packet_sender,       // Receive side.
        RtpReceiverStatistics* rtp_receiver_statistics,
        RtcpMode rtcp_mode,
        const base::TimeDelta& rtcp_interval,
@@ -150,7 +150,7 @@ class Rtcp {
                            uint32 last_ntp_fraction);
 
   scoped_refptr<CastEnvironment> cast_environment_;
-  transport::CastTransportSender* const transport_sender_;
+  CastTransportSender* const transport_sender_;
   const base::TimeDelta rtcp_interval_;
   const RtcpMode rtcp_mode_;
   const uint32 local_ssrc_;
@@ -194,8 +194,8 @@ class Rtcp {
   base::TimeDelta min_rtt_;
   base::TimeDelta max_rtt_;
   int number_of_rtt_in_avg_;
-  double avg_rtt_ms_;
-  uint16 target_delay_ms_;
+  base::TimeDelta avg_rtt_;
+  base::TimeDelta target_delay_;
 
   DISALLOW_COPY_AND_ASSIGN(Rtcp);
 };
diff --git a/media/cast/transport/rtcp/rtcp_builder.cc b/media/cast/net/rtcp/rtcp_builder.cc
index b8875fc96b..a5673a33f2 100644
--- a/media/cast/transport/rtcp/rtcp_builder.cc
+++ b/media/cast/net/rtcp/rtcp_builder.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/transport/rtcp/rtcp_builder.h"
+#include "media/cast/net/rtcp/rtcp_builder.h"
 
 #include <algorithm>
 #include <string>
@@ -10,12 +10,11 @@
 
 #include "base/big_endian.h"
 #include "base/logging.h"
-#include "media/cast/transport/cast_transport_defines.h"
-#include "media/cast/transport/pacing/paced_sender.h"
+#include "media/cast/net/cast_transport_defines.h"
+#include "media/cast/net/pacing/paced_sender.h"
 
 namespace media {
 namespace cast {
-namespace transport {
 
 RtcpBuilder::RtcpBuilder(PacedSender* const outgoing_transport)
     : transport_(outgoing_transport),
@@ -192,6 +191,5 @@ bool RtcpBuilder::BuildDlrrRb(const RtcpDlrrReportBlock& dlrr,
   return true;
 }
 
-}  // namespace transport
 }  // namespace cast
 }  // namespace media
diff --git a/media/cast/transport/rtcp/rtcp_builder.h b/media/cast/net/rtcp/rtcp_builder.h
index f095ae9ee5..1a9b169be1 100644
--- a/media/cast/transport/rtcp/rtcp_builder.h
+++ b/media/cast/net/rtcp/rtcp_builder.h
@@ -1,20 +1,19 @@
-// 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_TRANSPORT_RTCP_RTCP_BUILDER_H_
-#define MEDIA_CAST_TRANSPORT_RTCP_RTCP_BUILDER_H_
+#ifndef MEDIA_CAST_NET_RTCP_RTCP_BUILDER_H_
+#define MEDIA_CAST_NET_RTCP_RTCP_BUILDER_H_
 
 #include <list>
 #include <string>
 #include <vector>
 
-#include "media/cast/transport/cast_transport_defines.h"
-#include "media/cast/transport/pacing/paced_sender.h"
+#include "media/cast/net/cast_transport_defines.h"
+#include "media/cast/net/pacing/paced_sender.h"
 
 namespace media {
 namespace cast {
-namespace transport {
 
 class RtcpBuilder {
  public:
@@ -42,8 +41,7 @@ class RtcpBuilder {
   DISALLOW_COPY_AND_ASSIGN(RtcpBuilder);
 };
 
-}  // namespace transport
 }  // namespace cast
 }  // namespace media
 
-#endif  // MEDIA_CAST_TRANSPORT_RTCP_RTCP_BUILDER_H_
+#endif  // MEDIA_CAST_NET_RTCP_RTCP_BUILDER_H_
diff --git a/media/cast/transport/rtcp/rtcp_builder_unittest.cc b/media/cast/net/rtcp/rtcp_builder_unittest.cc
index 0322612f27..1c471f5176 100644
--- a/media/cast/transport/rtcp/rtcp_builder_unittest.cc
+++ b/media/cast/net/rtcp/rtcp_builder_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.
 
@@ -6,11 +6,11 @@
 #include "base/test/simple_test_tick_clock.h"
 #include "media/cast/cast_defines.h"
 #include "media/cast/cast_environment.h"
-#include "media/cast/rtcp/rtcp_utility.h"
-#include "media/cast/rtcp/test_rtcp_packet_builder.h"
+#include "media/cast/net/pacing/paced_sender.h"
+#include "media/cast/net/rtcp/rtcp_builder.h"
+#include "media/cast/net/rtcp/rtcp_utility.h"
+#include "media/cast/net/rtcp/test_rtcp_packet_builder.h"
 #include "media/cast/test/fake_single_thread_task_runner.h"
-#include "media/cast/transport/pacing/paced_sender.h"
-#include "media/cast/transport/rtcp/rtcp_builder.h"
 #include "testing/gmock/include/gmock/gmock.h"
 
 namespace media {
diff --git a/media/cast/rtcp/rtcp_defines.cc b/media/cast/net/rtcp/rtcp_defines.cc
index 214100d4d9..63799a92df 100644
--- a/media/cast/rtcp/rtcp_defines.cc
+++ b/media/cast/net/rtcp/rtcp_defines.cc
@@ -2,7 +2,7 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-#include "media/cast/rtcp/rtcp_defines.h"
+#include "media/cast/net/rtcp/rtcp_defines.h"
 
 #include "media/cast/logging/logging_defines.h"
 
diff --git a/media/cast/rtcp/rtcp_defines.h b/media/cast/net/rtcp/rtcp_defines.h
index 31795648c6..28f69e1835 100644
--- a/media/cast/rtcp/rtcp_defines.h
+++ b/media/cast/net/rtcp/rtcp_defines.h
@@ -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.
 
@@ -11,7 +11,7 @@
 #include "media/cast/cast_config.h"
 #include "media/cast/cast_defines.h"
 #include "media/cast/logging/logging_defines.h"
-#include "media/cast/transport/cast_transport_defines.h"
+#include "media/cast/net/cast_transport_defines.h"
 
 namespace media {
 namespace cast {
diff --git a/media/cast/rtcp/rtcp_receiver.cc b/media/cast/net/rtcp/rtcp_receiver.cc
index 3be8e921c4..99fd9178ae 100644
--- a/media/cast/rtcp/rtcp_receiver.cc
+++ b/media/cast/net/rtcp/rtcp_receiver.cc
@@ -1,12 +1,12 @@
-// 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/rtcp/rtcp_receiver.h"
+#include "media/cast/net/rtcp/rtcp_receiver.h"
 
 #include "base/logging.h"
-#include "media/cast/rtcp/rtcp_utility.h"
-#include "media/cast/transport/cast_transport_defines.h"
+#include "media/cast/net/cast_transport_defines.h"
+#include "media/cast/net/rtcp/rtcp_utility.h"
 
 namespace {
 
@@ -135,7 +135,7 @@ void RtcpReceiver::HandleSenderReport(RtcpParser* rtcp_parser) {
   VLOG(2) << "Cast RTCP received SR from SSRC " << remote_ssrc;
 
   if (remote_ssrc_ == remote_ssrc) {
-    transport::RtcpSenderInfo remote_sender_info;
+    RtcpSenderInfo remote_sender_info;
     remote_sender_info.ntp_seconds =
         rtcp_field.sender_report.ntp_most_significant;
     remote_sender_info.ntp_fraction =
@@ -193,7 +193,7 @@ void RtcpReceiver::HandleReportBlock(const RtcpField* rtcp_field,
   }
   VLOG(2) << "Cast RTCP received RB from SSRC " << remote_ssrc;
 
-  transport::RtcpReportBlock report_block;
+  RtcpReportBlock report_block;
   report_block.remote_ssrc = remote_ssrc;
   report_block.media_ssrc = rb.ssrc;
   report_block.fraction_lost = rb.fraction_lost;
diff --git a/media/cast/rtcp/rtcp_receiver.h b/media/cast/net/rtcp/rtcp_receiver.h
index d3cef9e57b..aea45846a2 100644
--- a/media/cast/rtcp/rtcp_receiver.h
+++ b/media/cast/net/rtcp/rtcp_receiver.h
@@ -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 +8,10 @@
 #include <queue>
 
 #include "base/containers/hash_tables.h"
-#include "media/cast/rtcp/rtcp.h"
-#include "media/cast/rtcp/rtcp_defines.h"
-#include "media/cast/rtcp/rtcp_utility.h"
-#include "media/cast/transport/cast_transport_defines.h"
+#include "media/cast/net/cast_transport_defines.h"
+#include "media/cast/net/rtcp/rtcp.h"
+#include "media/cast/net/rtcp/rtcp_defines.h"
+#include "media/cast/net/rtcp/rtcp_utility.h"
 
 namespace media {
 namespace cast {
@@ -19,7 +19,7 @@ namespace cast {
 class RtcpReceiverFeedback {
  public:
   virtual void OnReceivedSenderReport(
-      const transport::RtcpSenderInfo& remote_sender_info) = 0;
+      const RtcpSenderInfo& remote_sender_info) = 0;
 
   virtual void OnReceiverReferenceTimeReport(
       const RtcpReceiverReferenceTimeReport& remote_time_report) = 0;
@@ -116,7 +116,7 @@ class RtcpReceiver {
   RtcpRttFeedback* const rtt_feedback_;
   scoped_refptr<CastEnvironment> cast_environment_;
 
-  transport::FrameIdWrapHelper ack_frame_id_wrap_helper_;
+  FrameIdWrapHelper ack_frame_id_wrap_helper_;
 
   // Maintains a history of receiver events.
   size_t receiver_event_history_size_;
diff --git a/media/cast/rtcp/rtcp_receiver_unittest.cc b/media/cast/net/rtcp/rtcp_receiver_unittest.cc
index 51026d1554..f4010cb36b 100644
--- a/media/cast/rtcp/rtcp_receiver_unittest.cc
+++ b/media/cast/net/rtcp/rtcp_receiver_unittest.cc
@@ -1,17 +1,17 @@
-// 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 "base/memory/scoped_ptr.h"
 #include "base/test/simple_test_tick_clock.h"
 #include "media/cast/cast_environment.h"
-#include "media/cast/rtcp/mock_rtcp_receiver_feedback.h"
-#include "media/cast/rtcp/mock_rtcp_sender_feedback.h"
-#include "media/cast/rtcp/rtcp_receiver.h"
-#include "media/cast/rtcp/rtcp_utility.h"
-#include "media/cast/rtcp/test_rtcp_packet_builder.h"
+#include "media/cast/net/cast_transport_defines.h"
+#include "media/cast/net/rtcp/mock_rtcp_receiver_feedback.h"
+#include "media/cast/net/rtcp/mock_rtcp_sender_feedback.h"
+#include "media/cast/net/rtcp/rtcp_receiver.h"
+#include "media/cast/net/rtcp/rtcp_utility.h"
+#include "media/cast/net/rtcp/test_rtcp_packet_builder.h"
 #include "media/cast/test/fake_single_thread_task_runner.h"
-#include "media/cast/transport/cast_transport_defines.h"
 #include "testing/gmock/include/gmock/gmock.h"
 
 namespace media {
@@ -22,7 +22,8 @@ using testing::_;
 static const uint32 kSenderSsrc = 0x10203;
 static const uint32 kSourceSsrc = 0x40506;
 static const uint32 kUnknownSsrc = 0xDEAD;
-static const uint16 kTargetDelayMs = 100;
+static const base::TimeDelta kTargetDelay =
+    base::TimeDelta::FromMilliseconds(100);
 static const std::string kCName("test@10.1.1.1");
 
 namespace {
@@ -72,7 +73,7 @@ class RtcpReceiverCastLogVerification : public RtcpReceiverFeedback {
         called_on_received_receiver_log_(false) {}
 
   virtual void OnReceivedSenderReport(
-      const transport::RtcpSenderInfo& remote_sender_info) OVERRIDE{};
+      const RtcpSenderInfo& remote_sender_info) OVERRIDE{};
 
   virtual void OnReceiverReferenceTimeReport(
       const RtcpReceiverReferenceTimeReport& remote_time_report) OVERRIDE{};
@@ -188,8 +189,8 @@ class RtcpReceiverTest : public ::testing::Test {
   MockRtcpRttFeedback mock_rtt_feedback_;
   MockRtcpSenderFeedback mock_sender_feedback_;
   scoped_ptr<RtcpReceiver> rtcp_receiver_;
-  transport::RtcpSenderInfo expected_sender_info_;
-  transport::RtcpReportBlock expected_report_block_;
+  RtcpSenderInfo expected_sender_info_;
+  RtcpReportBlock expected_report_block_;
   RtcpReceiverReferenceTimeReport expected_receiver_reference_report_;
 
   DISALLOW_COPY_AND_ASSIGN(RtcpReceiverTest);
@@ -374,7 +375,7 @@ TEST_F(RtcpReceiverTest, InjectReceiverReportPacketWithCastFeedback) {
   TestRtcpPacketBuilder p1;
   p1.AddRr(kSenderSsrc, 1);
   p1.AddRb(kUnknownSsrc);
-  p1.AddCast(kSenderSsrc, kUnknownSsrc, kTargetDelayMs);
+  p1.AddCast(kSenderSsrc, kUnknownSsrc, kTargetDelay);
 
   // Expected to be ignored since the source ssrc does not match our
   // local ssrc.
@@ -391,7 +392,7 @@ TEST_F(RtcpReceiverTest, InjectReceiverReportPacketWithCastFeedback) {
   TestRtcpPacketBuilder p2;
   p2.AddRr(kSenderSsrc, 1);
   p2.AddRb(kSourceSsrc);
-  p2.AddCast(kSenderSsrc, kSourceSsrc, kTargetDelayMs);
+  p2.AddCast(kSenderSsrc, kSourceSsrc, kTargetDelay);
 
   // Expected to be pass through since the sender ssrc match our local ssrc.
   InjectRtcpPacket(p2.Data(), p2.Length());
@@ -415,7 +416,7 @@ TEST_F(RtcpReceiverTest, InjectReceiverReportPacketWithCastVerification) {
   TestRtcpPacketBuilder p;
   p.AddRr(kSenderSsrc, 1);
   p.AddRb(kSourceSsrc);
-  p.AddCast(kSenderSsrc, kSourceSsrc, kTargetDelayMs);
+  p.AddCast(kSenderSsrc, kSourceSsrc, kTargetDelay);
 
   // Expected to be pass through since the sender ssrc match our local ssrc.
   RtcpParser rtcp_parser(p.Data(), p.Length());
diff --git a/media/cast/rtcp/rtcp_sender.cc b/media/cast/net/rtcp/rtcp_sender.cc
index bf7d30c84c..a44f87ebb0 100644
--- a/media/cast/rtcp/rtcp_sender.cc
+++ b/media/cast/net/rtcp/rtcp_sender.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/rtcp/rtcp_sender.h"
+#include "media/cast/net/rtcp/rtcp_sender.h"
 
 #include <stdint.h>
 
@@ -12,10 +12,10 @@
 #include "base/big_endian.h"
 #include "base/logging.h"
 #include "media/cast/cast_environment.h"
-#include "media/cast/rtcp/rtcp_defines.h"
-#include "media/cast/rtcp/rtcp_utility.h"
-#include "media/cast/transport/cast_transport_defines.h"
-#include "media/cast/transport/pacing/paced_sender.h"
+#include "media/cast/net/cast_transport_defines.h"
+#include "media/cast/net/pacing/paced_sender.h"
+#include "media/cast/net/rtcp/rtcp_defines.h"
+#include "media/cast/net/rtcp/rtcp_utility.h"
 
 namespace media {
 namespace cast {
@@ -148,7 +148,7 @@ class NackStringBuilder {
 
 // TODO(mikhal): This is only used by the receiver. Consider renaming.
 RtcpSender::RtcpSender(scoped_refptr<CastEnvironment> cast_environment,
-                       transport::PacedPacketSender* outgoing_transport,
+                       PacedPacketSender* outgoing_transport,
                        uint32 sending_ssrc,
                        const std::string& c_name)
     : ssrc_(sending_ssrc),
@@ -162,44 +162,44 @@ RtcpSender::~RtcpSender() {}
 
 void RtcpSender::SendRtcpFromRtpReceiver(
     uint32 packet_type_flags,
-    const transport::RtcpReportBlock* report_block,
+    const RtcpReportBlock* report_block,
     const RtcpReceiverReferenceTimeReport* rrtr,
     const RtcpCastMessage* cast_message,
     const ReceiverRtcpEventSubscriber::RtcpEventMultiMap* rtcp_events,
-    uint16 target_delay_ms) {
-  if (packet_type_flags & transport::kRtcpSr ||
-      packet_type_flags & transport::kRtcpDlrr ||
-      packet_type_flags & transport::kRtcpSenderLog) {
+    base::TimeDelta target_delay) {
+  if (packet_type_flags & kRtcpSr ||
+      packet_type_flags & kRtcpDlrr ||
+      packet_type_flags & kRtcpSenderLog) {
     NOTREACHED() << "Invalid argument";
   }
-  if (packet_type_flags & transport::kRtcpPli ||
-      packet_type_flags & transport::kRtcpRpsi ||
-      packet_type_flags & transport::kRtcpRemb ||
-      packet_type_flags & transport::kRtcpNack) {
+  if (packet_type_flags & kRtcpPli ||
+      packet_type_flags & kRtcpRpsi ||
+      packet_type_flags & kRtcpRemb ||
+      packet_type_flags & kRtcpNack) {
     // Implement these for webrtc interop.
     NOTIMPLEMENTED();
   }
-  transport::PacketRef packet(new base::RefCountedData<Packet>);
+  PacketRef packet(new base::RefCountedData<Packet>);
   packet->data.reserve(kMaxIpPacketSize);
 
-  if (packet_type_flags & transport::kRtcpRr) {
+  if (packet_type_flags & kRtcpRr) {
     BuildRR(report_block, &packet->data);
     if (!c_name_.empty()) {
       BuildSdec(&packet->data);
     }
   }
-  if (packet_type_flags & transport::kRtcpBye) {
+  if (packet_type_flags & kRtcpBye) {
     BuildBye(&packet->data);
   }
-  if (packet_type_flags & transport::kRtcpRrtr) {
+  if (packet_type_flags & kRtcpRrtr) {
     DCHECK(rrtr) << "Invalid argument";
     BuildRrtr(rrtr, &packet->data);
   }
-  if (packet_type_flags & transport::kRtcpCast) {
+  if (packet_type_flags & kRtcpCast) {
     DCHECK(cast_message) << "Invalid argument";
-    BuildCast(cast_message, target_delay_ms, &packet->data);
+    BuildCast(cast_message, target_delay, &packet->data);
   }
-  if (packet_type_flags & transport::kRtcpReceiverLog) {
+  if (packet_type_flags & kRtcpReceiverLog) {
     DCHECK(rtcp_events) << "Invalid argument";
     BuildReceiverLog(*rtcp_events, &packet->data);
   }
@@ -210,7 +210,7 @@ void RtcpSender::SendRtcpFromRtpReceiver(
   transport_->SendRtcpPacket(ssrc_, packet);
 }
 
-void RtcpSender::BuildRR(const transport::RtcpReportBlock* report_block,
+void RtcpSender::BuildRR(const RtcpReportBlock* report_block,
                          Packet* packet) const {
   size_t start_size = packet->size();
   DCHECK_LT(start_size + 32, kMaxIpPacketSize) << "Not enough buffer space";
@@ -223,7 +223,7 @@ void RtcpSender::BuildRR(const transport::RtcpReportBlock* report_block,
   base::BigEndianWriter big_endian_writer(
       reinterpret_cast<char*>(&((*packet)[start_size])), 8);
   big_endian_writer.WriteU8(0x80 + (report_block ? 1 : 0));
-  big_endian_writer.WriteU8(transport::kPacketTypeReceiverReport);
+  big_endian_writer.WriteU8(kPacketTypeReceiverReport);
   big_endian_writer.WriteU16(number_of_rows);
   big_endian_writer.WriteU32(ssrc_);
 
@@ -232,7 +232,7 @@ void RtcpSender::BuildRR(const transport::RtcpReportBlock* report_block,
   }
 }
 
-void RtcpSender::AddReportBlocks(const transport::RtcpReportBlock& report_block,
+void RtcpSender::AddReportBlocks(const RtcpReportBlock& report_block,
                                  Packet* packet) const {
   size_t start_size = packet->size();
   DCHECK_LT(start_size + 24, kMaxIpPacketSize) << "Not enough buffer space";
@@ -276,7 +276,7 @@ void RtcpSender::BuildSdec(Packet* packet) const {
       reinterpret_cast<char*>(&((*packet)[start_size])), 10);
   // We always need to add one SDES CNAME.
   big_endian_writer.WriteU8(0x80 + 1);
-  big_endian_writer.WriteU8(transport::kPacketTypeSdes);
+  big_endian_writer.WriteU8(kPacketTypeSdes);
 
   // Handle SDES length later on.
   uint32 sdes_length_position = static_cast<uint32>(start_size) + 3;
@@ -319,7 +319,7 @@ void RtcpSender::BuildPli(uint32 remote_ssrc, Packet* packet) const {
       reinterpret_cast<char*>(&((*packet)[start_size])), 12);
   uint8 FMT = 1;  // Picture loss indicator.
   big_endian_writer.WriteU8(0x80 + FMT);
-  big_endian_writer.WriteU8(transport::kPacketTypePayloadSpecific);
+  big_endian_writer.WriteU8(kPacketTypePayloadSpecific);
   big_endian_writer.WriteU16(2);            // Used fixed length of 2.
   big_endian_writer.WriteU32(ssrc_);        // Add our own SSRC.
   big_endian_writer.WriteU32(remote_ssrc);  // Add the remote SSRC.
@@ -346,7 +346,7 @@ void RtcpSender::BuildRpsi(const RtcpRpsiMessage* rpsi, Packet* packet) const {
       reinterpret_cast<char*>(&((*packet)[start_size])), 24);
   uint8 FMT = 3;  // Reference Picture Selection Indication.
   big_endian_writer.WriteU8(0x80 + FMT);
-  big_endian_writer.WriteU8(transport::kPacketTypePayloadSpecific);
+  big_endian_writer.WriteU8(kPacketTypePayloadSpecific);
 
   // Calculate length.
   uint32 bits_required = 7;
@@ -404,7 +404,7 @@ void RtcpSender::BuildRemb(const RtcpRembMessage* remb, Packet* packet) const {
   // Add application layer feedback.
   uint8 FMT = 15;
   big_endian_writer.WriteU8(0x80 + FMT);
-  big_endian_writer.WriteU8(transport::kPacketTypePayloadSpecific);
+  big_endian_writer.WriteU8(kPacketTypePayloadSpecific);
   big_endian_writer.WriteU8(0);
   big_endian_writer.WriteU8(static_cast<uint8>(remb->remb_ssrcs.size() + 4));
   big_endian_writer.WriteU32(ssrc_);  // Add our own SSRC.
@@ -442,7 +442,7 @@ void RtcpSender::BuildNack(const RtcpNackMessage* nack, Packet* packet) const {
 
   uint8 FMT = 1;
   big_endian_writer.WriteU8(0x80 + FMT);
-  big_endian_writer.WriteU8(transport::kPacketTypeGenericRtpFeedback);
+  big_endian_writer.WriteU8(kPacketTypeGenericRtpFeedback);
   big_endian_writer.WriteU8(0);
   size_t nack_size_pos = start_size + 3;
   big_endian_writer.WriteU8(3);
@@ -498,7 +498,7 @@ void RtcpSender::BuildBye(Packet* packet) const {
   base::BigEndianWriter big_endian_writer(
       reinterpret_cast<char*>(&((*packet)[start_size])), 8);
   big_endian_writer.WriteU8(0x80 + 1);
-  big_endian_writer.WriteU8(transport::kPacketTypeBye);
+  big_endian_writer.WriteU8(kPacketTypeBye);
   big_endian_writer.WriteU16(1);      // Length.
   big_endian_writer.WriteU32(ssrc_);  // Add our own SSRC.
 }
@@ -516,7 +516,7 @@ void RtcpSender::BuildRrtr(const RtcpReceiverReferenceTimeReport* rrtr,
       reinterpret_cast<char*>(&((*packet)[start_size])), 20);
 
   big_endian_writer.WriteU8(0x80);
-  big_endian_writer.WriteU8(transport::kPacketTypeXr);
+  big_endian_writer.WriteU8(kPacketTypeXr);
   big_endian_writer.WriteU16(4);      // Length.
   big_endian_writer.WriteU32(ssrc_);  // Add our own SSRC.
   big_endian_writer.WriteU8(4);       // Add block type.
@@ -529,7 +529,7 @@ void RtcpSender::BuildRrtr(const RtcpReceiverReferenceTimeReport* rrtr,
 }
 
 void RtcpSender::BuildCast(const RtcpCastMessage* cast,
-                           uint16 target_delay_ms,
+                           base::TimeDelta target_delay,
                            Packet* packet) const {
   size_t start_size = packet->size();
   DCHECK_LT(start_size + 20, kMaxIpPacketSize) << "Not enough buffer space";
@@ -542,7 +542,7 @@ void RtcpSender::BuildCast(const RtcpCastMessage* cast,
       reinterpret_cast<char*>(&((*packet)[start_size])), 20);
   uint8 FMT = 15;  // Application layer feedback.
   big_endian_writer.WriteU8(0x80 + FMT);
-  big_endian_writer.WriteU8(transport::kPacketTypePayloadSpecific);
+  big_endian_writer.WriteU8(kPacketTypePayloadSpecific);
   big_endian_writer.WriteU8(0);
   size_t cast_size_pos = start_size + 3;  // Save length position.
   big_endian_writer.WriteU8(4);
@@ -552,7 +552,9 @@ void RtcpSender::BuildCast(const RtcpCastMessage* cast,
   big_endian_writer.WriteU8(static_cast<uint8>(cast->ack_frame_id_));
   size_t cast_loss_field_pos = start_size + 17;  // Save loss field position.
   big_endian_writer.WriteU8(0);  // Overwritten with number_of_loss_fields.
-  big_endian_writer.WriteU16(target_delay_ms);
+  DCHECK_LE(target_delay.InMilliseconds(),
+            std::numeric_limits<uint16_t>::max());
+  big_endian_writer.WriteU16(target_delay.InMilliseconds());
 
   size_t number_of_loss_fields = 0;
   size_t max_number_of_loss_fields = std::min<size_t>(
@@ -641,7 +643,7 @@ void RtcpSender::BuildReceiverLog(
   base::BigEndianWriter big_endian_writer(
       reinterpret_cast<char*>(&((*packet)[packet_start_size])), rtcp_log_size);
   big_endian_writer.WriteU8(0x80 + kReceiverLogSubtype);
-  big_endian_writer.WriteU8(transport::kPacketTypeApplicationDefined);
+  big_endian_writer.WriteU8(kPacketTypeApplicationDefined);
   big_endian_writer.WriteU16(static_cast<uint16>(
       2 + 2 * number_of_frames + total_number_of_messages_to_send));
   big_endian_writer.WriteU32(ssrc_);  // Add our own SSRC.
diff --git a/media/cast/rtcp/rtcp_sender.h b/media/cast/net/rtcp/rtcp_sender.h
index f09a4fb0e5..fe5af0dcd3 100644
--- a/media/cast/rtcp/rtcp_sender.h
+++ b/media/cast/net/rtcp/rtcp_sender.h
@@ -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.
 
@@ -11,11 +11,11 @@
 
 #include "media/cast/cast_config.h"
 #include "media/cast/cast_defines.h"
-#include "media/cast/rtcp/receiver_rtcp_event_subscriber.h"
-#include "media/cast/rtcp/rtcp.h"
-#include "media/cast/rtcp/rtcp_defines.h"
-#include "media/cast/transport/cast_transport_defines.h"
-#include "media/cast/transport/rtcp/rtcp_builder.h"
+#include "media/cast/net/cast_transport_defines.h"
+#include "media/cast/net/rtcp/receiver_rtcp_event_subscriber.h"
+#include "media/cast/net/rtcp/rtcp.h"
+#include "media/cast/net/rtcp/rtcp_builder.h"
+#include "media/cast/net/rtcp/rtcp_defines.h"
 
 namespace media {
 namespace cast {
@@ -44,7 +44,7 @@ COMPILE_ASSERT(kSecondRedundancyOffset >
 class RtcpSender {
  public:
   RtcpSender(scoped_refptr<CastEnvironment> cast_environment,
-             transport::PacedPacketSender* outgoing_transport,
+             PacedPacketSender* outgoing_transport,
              uint32 sending_ssrc,
              const std::string& c_name);
 
@@ -52,17 +52,17 @@ class RtcpSender {
 
   void SendRtcpFromRtpReceiver(
       uint32 packet_type_flags,
-      const transport::RtcpReportBlock* report_block,
+      const RtcpReportBlock* report_block,
       const RtcpReceiverReferenceTimeReport* rrtr,
       const RtcpCastMessage* cast_message,
       const ReceiverRtcpEventSubscriber::RtcpEventMultiMap* rtcp_events,
-      uint16 target_delay_ms);
+      base::TimeDelta target_delay);
 
  private:
-  void BuildRR(const transport::RtcpReportBlock* report_block,
+  void BuildRR(const RtcpReportBlock* report_block,
                Packet* packet) const;
 
-  void AddReportBlocks(const transport::RtcpReportBlock& report_block,
+  void AddReportBlocks(const RtcpReportBlock& report_block,
                        Packet* packet) const;
 
   void BuildSdec(Packet* packet) const;
@@ -81,7 +81,7 @@ class RtcpSender {
                  Packet* packet) const;
 
   void BuildCast(const RtcpCastMessage* cast_message,
-                 uint16 target_delay_ms,
+                 base::TimeDelta target_delay,
                  Packet* packet) const;
 
   void BuildReceiverLog(
@@ -114,7 +114,7 @@ class RtcpSender {
   const std::string c_name_;
 
   // Not owned by this class.
-  transport::PacedPacketSender* const transport_;
+  PacedPacketSender* const transport_;
   scoped_refptr<CastEnvironment> cast_environment_;
 
   std::deque<RtcpReceiverLogMessage> rtcp_events_history_;
diff --git a/media/cast/rtcp/rtcp_sender_unittest.cc b/media/cast/net/rtcp/rtcp_sender_unittest.cc
index 0b0c7d3ab8..c0c321e2c8 100644
--- a/media/cast/rtcp/rtcp_sender_unittest.cc
+++ b/media/cast/net/rtcp/rtcp_sender_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.
 
@@ -6,13 +6,13 @@
 #include "base/test/simple_test_tick_clock.h"
 #include "media/cast/cast_defines.h"
 #include "media/cast/cast_environment.h"
-#include "media/cast/rtcp/receiver_rtcp_event_subscriber.h"
-#include "media/cast/rtcp/rtcp_sender.h"
-#include "media/cast/rtcp/rtcp_utility.h"
-#include "media/cast/rtcp/test_rtcp_packet_builder.h"
+#include "media/cast/net/cast_transport_defines.h"
+#include "media/cast/net/pacing/paced_sender.h"
+#include "media/cast/net/rtcp/receiver_rtcp_event_subscriber.h"
+#include "media/cast/net/rtcp/rtcp_sender.h"
+#include "media/cast/net/rtcp/rtcp_utility.h"
+#include "media/cast/net/rtcp/test_rtcp_packet_builder.h"
 #include "media/cast/test/fake_single_thread_task_runner.h"
-#include "media/cast/transport/cast_transport_defines.h"
-#include "media/cast/transport/pacing/paced_sender.h"
 #include "testing/gmock/include/gmock/gmock.h"
 
 namespace media {
@@ -21,11 +21,12 @@ namespace cast {
 namespace {
 static const uint32 kSendingSsrc = 0x12345678;
 static const uint32 kMediaSsrc = 0x87654321;
-static const int16 kDefaultDelay = 100;
+static const base::TimeDelta kDefaultDelay =
+    base::TimeDelta::FromMilliseconds(100);
 static const std::string kCName("test@10.1.1.1");
 
-transport::RtcpReportBlock GetReportBlock() {
-  transport::RtcpReportBlock report_block;
+RtcpReportBlock GetReportBlock() {
+  RtcpReportBlock report_block;
   // Initialize remote_ssrc to a "clearly illegal" value.
   report_block.remote_ssrc = 0xDEAD;
   report_block.media_ssrc = kMediaSsrc;  // SSRC of the RTP packet sender.
@@ -40,12 +41,12 @@ transport::RtcpReportBlock GetReportBlock() {
 
 }  // namespace
 
-class TestRtcpTransport : public transport::PacedPacketSender {
+class TestRtcpTransport : public PacedPacketSender {
  public:
   TestRtcpTransport() : packet_count_(0) {}
 
   virtual bool SendRtcpPacket(uint32 ssrc,
-                              transport::PacketRef packet) OVERRIDE {
+                              PacketRef packet) OVERRIDE {
     EXPECT_EQ(expected_packet_.size(), packet->data.size());
     EXPECT_EQ(0, memcmp(expected_packet_.data(),
                         packet->data.data(),
@@ -55,17 +56,17 @@ class TestRtcpTransport : public transport::PacedPacketSender {
   }
 
   virtual bool SendPackets(
-      const transport::SendPacketVector& packets) OVERRIDE {
+      const SendPacketVector& packets) OVERRIDE {
     return false;
   }
   virtual bool ResendPackets(
-      const transport::SendPacketVector& packets,
+      const SendPacketVector& packets,
       base::TimeDelta dedupe_window) OVERRIDE {
     return false;
   }
 
   virtual void CancelSendingPacket(
-      const transport::PacketKey& packet_key) OVERRIDE {
+      const PacketKey& packet_key) OVERRIDE {
   }
 
   void SetExpectedRtcpPacket(scoped_ptr<Packet> packet) {
@@ -113,7 +114,7 @@ TEST_F(RtcpSenderTest, RtcpReceiverReport) {
   test_transport_.SetExpectedRtcpPacket(p1.GetPacket());
 
   rtcp_sender_->SendRtcpFromRtpReceiver(
-      transport::kRtcpRr, NULL, NULL, NULL, NULL, kDefaultDelay);
+      kRtcpRr, NULL, NULL, NULL, NULL, kDefaultDelay);
 
   EXPECT_EQ(1, test_transport_.packet_count());
 
@@ -124,10 +125,10 @@ TEST_F(RtcpSenderTest, RtcpReceiverReport) {
   p2.AddSdesCname(kSendingSsrc, kCName);
   test_transport_.SetExpectedRtcpPacket(p2.GetPacket().Pass());
 
-  transport::RtcpReportBlock report_block = GetReportBlock();
+  RtcpReportBlock report_block = GetReportBlock();
 
   rtcp_sender_->SendRtcpFromRtpReceiver(
-      transport::kRtcpRr, &report_block, NULL, NULL, NULL, kDefaultDelay);
+      kRtcpRr, &report_block, NULL, NULL, NULL, kDefaultDelay);
 
   EXPECT_EQ(2, test_transport_.packet_count());
 }
@@ -142,14 +143,14 @@ TEST_F(RtcpSenderTest, RtcpReceiverReportWithRrtr) {
   p.AddXrRrtrBlock();
   test_transport_.SetExpectedRtcpPacket(p.GetPacket().Pass());
 
-  transport::RtcpReportBlock report_block = GetReportBlock();
+  RtcpReportBlock report_block = GetReportBlock();
 
   RtcpReceiverReferenceTimeReport rrtr;
   rrtr.ntp_seconds = kNtpHigh;
   rrtr.ntp_fraction = kNtpLow;
 
   rtcp_sender_->SendRtcpFromRtpReceiver(
-      transport::kRtcpRr | transport::kRtcpRrtr,
+      kRtcpRr | kRtcpRrtr,
       &report_block,
       &rrtr,
       NULL,
@@ -168,7 +169,7 @@ TEST_F(RtcpSenderTest, RtcpReceiverReportWithCast) {
   p.AddCast(kSendingSsrc, kMediaSsrc, kDefaultDelay);
   test_transport_.SetExpectedRtcpPacket(p.GetPacket().Pass());
 
-  transport::RtcpReportBlock report_block = GetReportBlock();
+  RtcpReportBlock report_block = GetReportBlock();
 
   RtcpCastMessage cast_message(kMediaSsrc);
   cast_message.ack_frame_id_ = kAckFrameId;
@@ -182,7 +183,7 @@ TEST_F(RtcpSenderTest, RtcpReceiverReportWithCast) {
       missing_packets;
 
   rtcp_sender_->SendRtcpFromRtpReceiver(
-      transport::kRtcpRr | transport::kRtcpCast,
+      kRtcpRr | kRtcpCast,
       &report_block,
       NULL,
       &cast_message,
@@ -202,7 +203,7 @@ TEST_F(RtcpSenderTest, RtcpReceiverReportWithRrtraAndCastMessage) {
   p.AddCast(kSendingSsrc, kMediaSsrc, kDefaultDelay);
   test_transport_.SetExpectedRtcpPacket(p.GetPacket().Pass());
 
-  transport::RtcpReportBlock report_block = GetReportBlock();
+  RtcpReportBlock report_block = GetReportBlock();
 
   RtcpReceiverReferenceTimeReport rrtr;
   rrtr.ntp_seconds = kNtpHigh;
@@ -220,7 +221,7 @@ TEST_F(RtcpSenderTest, RtcpReceiverReportWithRrtraAndCastMessage) {
       missing_packets;
 
   rtcp_sender_->SendRtcpFromRtpReceiver(
-      transport::kRtcpRr | transport::kRtcpRrtr | transport::kRtcpCast,
+      kRtcpRr | kRtcpRrtr | kRtcpCast,
       &report_block,
       &rrtr,
       &cast_message,
@@ -243,7 +244,7 @@ TEST_F(RtcpSenderTest, RtcpReceiverReportWithRrtrCastMessageAndLog) {
   p.AddCast(kSendingSsrc, kMediaSsrc, kDefaultDelay);
   test_transport_.SetExpectedRtcpPacket(p.GetPacket().Pass());
 
-  transport::RtcpReportBlock report_block = GetReportBlock();
+  RtcpReportBlock report_block = GetReportBlock();
 
   RtcpReceiverReferenceTimeReport rrtr;
   rrtr.ntp_seconds = kNtpHigh;
@@ -264,8 +265,8 @@ TEST_F(RtcpSenderTest, RtcpReceiverReportWithRrtrCastMessageAndLog) {
   ReceiverRtcpEventSubscriber::RtcpEventMultiMap rtcp_events;
 
   rtcp_sender_->SendRtcpFromRtpReceiver(
-      transport::kRtcpRr | transport::kRtcpRrtr | transport::kRtcpCast |
-          transport::kRtcpReceiverLog,
+      kRtcpRr | kRtcpRrtr | kRtcpCast |
+          kRtcpReceiverLog,
       &report_block,
       &rrtr,
       &cast_message,
@@ -301,8 +302,8 @@ TEST_F(RtcpSenderTest, RtcpReceiverReportWithRrtrCastMessageAndLog) {
   EXPECT_EQ(2u, rtcp_events.size());
 
   rtcp_sender_->SendRtcpFromRtpReceiver(
-      transport::kRtcpRr | transport::kRtcpRrtr | transport::kRtcpCast |
-          transport::kRtcpReceiverLog,
+      kRtcpRr | kRtcpRrtr | kRtcpCast |
+          kRtcpReceiverLog,
       &report_block,
       &rrtr,
       &cast_message,
@@ -321,7 +322,7 @@ TEST_F(RtcpSenderTest, RtcpReceiverReportWithOversizedFrameLog) {
   p.AddRb(kMediaSsrc);
   p.AddSdesCname(kSendingSsrc, kCName);
 
-  transport::RtcpReportBlock report_block = GetReportBlock();
+  RtcpReportBlock report_block = GetReportBlock();
 
   base::SimpleTestTickClock testing_clock;
   testing_clock.Advance(base::TimeDelta::FromMilliseconds(kTimeBaseMs));
@@ -370,7 +371,7 @@ TEST_F(RtcpSenderTest, RtcpReceiverReportWithOversizedFrameLog) {
   event_subscriber.GetRtcpEventsAndReset(&rtcp_events);
 
   rtcp_sender_->SendRtcpFromRtpReceiver(
-      transport::kRtcpRr | transport::kRtcpReceiverLog,
+      kRtcpRr | kRtcpReceiverLog,
       &report_block,
       NULL,
       NULL,
@@ -389,7 +390,7 @@ TEST_F(RtcpSenderTest, RtcpReceiverReportWithTooManyLogFrames) {
   p.AddRb(kMediaSsrc);
   p.AddSdesCname(kSendingSsrc, kCName);
 
-  transport::RtcpReportBlock report_block = GetReportBlock();
+  RtcpReportBlock report_block = GetReportBlock();
 
   base::SimpleTestTickClock testing_clock;
   testing_clock.Advance(base::TimeDelta::FromMilliseconds(kTimeBaseMs));
@@ -427,7 +428,7 @@ TEST_F(RtcpSenderTest, RtcpReceiverReportWithTooManyLogFrames) {
   event_subscriber.GetRtcpEventsAndReset(&rtcp_events);
 
   rtcp_sender_->SendRtcpFromRtpReceiver(
-      transport::kRtcpRr | transport::kRtcpReceiverLog,
+      kRtcpRr | kRtcpReceiverLog,
       &report_block,
       NULL,
       NULL,
@@ -445,7 +446,7 @@ TEST_F(RtcpSenderTest, RtcpReceiverReportWithOldLogFrames) {
   p.AddRb(kMediaSsrc);
   p.AddSdesCname(kSendingSsrc, kCName);
 
-  transport::RtcpReportBlock report_block = GetReportBlock();
+  RtcpReportBlock report_block = GetReportBlock();
 
   base::SimpleTestTickClock testing_clock;
   testing_clock.Advance(base::TimeDelta::FromMilliseconds(kTimeBaseMs));
@@ -478,7 +479,7 @@ TEST_F(RtcpSenderTest, RtcpReceiverReportWithOldLogFrames) {
   event_subscriber.GetRtcpEventsAndReset(&rtcp_events);
 
   rtcp_sender_->SendRtcpFromRtpReceiver(
-      transport::kRtcpRr | transport::kRtcpReceiverLog,
+      kRtcpRr | kRtcpReceiverLog,
       &report_block,
       NULL,
       NULL,
@@ -492,7 +493,7 @@ TEST_F(RtcpSenderTest, RtcpReceiverReportRedundancy) {
   uint32 time_base_ms = 12345678;
   int kTimeBetweenEventsMs = 10;
 
-  transport::RtcpReportBlock report_block = GetReportBlock();
+  RtcpReportBlock report_block = GetReportBlock();
 
   base::SimpleTestTickClock testing_clock;
   testing_clock.Advance(base::TimeDelta::FromMilliseconds(time_base_ms));
@@ -537,7 +538,7 @@ TEST_F(RtcpSenderTest, RtcpReceiverReportRedundancy) {
     event_subscriber.GetRtcpEventsAndReset(&rtcp_events);
 
     rtcp_sender_->SendRtcpFromRtpReceiver(
-        transport::kRtcpRr | transport::kRtcpReceiverLog,
+        kRtcpRr | kRtcpReceiverLog,
         &report_block,
         NULL,
         NULL,
diff --git a/media/cast/rtcp/rtcp_unittest.cc b/media/cast/net/rtcp/rtcp_unittest.cc
index 095e6d24df..25d05b2239 100644
--- a/media/cast/rtcp/rtcp_unittest.cc
+++ b/media/cast/net/rtcp/rtcp_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.
 
@@ -7,14 +7,14 @@
 #include "base/test/simple_test_tick_clock.h"
 #include "media/cast/cast_defines.h"
 #include "media/cast/cast_environment.h"
-#include "media/cast/rtcp/mock_rtcp_receiver_feedback.h"
-#include "media/cast/rtcp/mock_rtcp_sender_feedback.h"
-#include "media/cast/rtcp/rtcp.h"
-#include "media/cast/rtcp/test_rtcp_packet_builder.h"
+#include "media/cast/net/cast_transport_config.h"
+#include "media/cast/net/cast_transport_sender_impl.h"
+#include "media/cast/net/pacing/paced_sender.h"
+#include "media/cast/net/rtcp/mock_rtcp_receiver_feedback.h"
+#include "media/cast/net/rtcp/mock_rtcp_sender_feedback.h"
+#include "media/cast/net/rtcp/rtcp.h"
+#include "media/cast/net/rtcp/test_rtcp_packet_builder.h"
 #include "media/cast/test/fake_single_thread_task_runner.h"
-#include "media/cast/transport/cast_transport_config.h"
-#include "media/cast/transport/cast_transport_sender_impl.h"
-#include "media/cast/transport/pacing/paced_sender.h"
 #include "testing/gmock/include/gmock/gmock.h"
 
 namespace media {
@@ -29,7 +29,7 @@ static const uint32 kRtcpIntervalMs = 500;
 static const int64 kAddedDelay = 123;
 static const int64 kAddedShortDelay = 100;
 
-class RtcpTestPacketSender : public transport::PacketSender {
+class RtcpTestPacketSender : public PacketSender {
  public:
   explicit RtcpTestPacketSender(base::SimpleTestTickClock* testing_clock)
       : drop_packets_(false),
@@ -45,7 +45,7 @@ class RtcpTestPacketSender : public transport::PacketSender {
   void set_drop_packets(bool drop_packets) { drop_packets_ = drop_packets; }
 
   // A singular packet implies a RTCP packet.
-  virtual bool SendPacket(transport::PacketRef packet,
+  virtual bool SendPacket(PacketRef packet,
                           const base::Closure& cb) OVERRIDE {
     if (short_delay_) {
       testing_clock_->Advance(
@@ -69,7 +69,7 @@ class RtcpTestPacketSender : public transport::PacketSender {
   DISALLOW_COPY_AND_ASSIGN(RtcpTestPacketSender);
 };
 
-class LocalRtcpTransport : public transport::PacedPacketSender {
+class LocalRtcpTransport : public PacedPacketSender {
  public:
   LocalRtcpTransport(scoped_refptr<CastEnvironment> cast_environment,
                      base::SimpleTestTickClock* testing_clock)
@@ -84,7 +84,7 @@ class LocalRtcpTransport : public transport::PacedPacketSender {
   void set_drop_packets(bool drop_packets) { drop_packets_ = drop_packets; }
 
   virtual bool SendRtcpPacket(uint32 ssrc,
-                              transport::PacketRef packet) OVERRIDE {
+                              PacketRef packet) OVERRIDE {
     if (short_delay_) {
       testing_clock_->Advance(
           base::TimeDelta::FromMilliseconds(kAddedShortDelay));
@@ -99,18 +99,18 @@ class LocalRtcpTransport : public transport::PacedPacketSender {
   }
 
   virtual bool SendPackets(
-      const transport::SendPacketVector& packets) OVERRIDE {
+      const SendPacketVector& packets) OVERRIDE {
     return false;
   }
 
   virtual bool ResendPackets(
-      const transport::SendPacketVector& packets,
+      const SendPacketVector& packets,
       base::TimeDelta dedupe_window) OVERRIDE {
     return false;
   }
 
   virtual void CancelSendingPacket(
-      const transport::PacketKey& packet_key) OVERRIDE {
+      const PacketKey& packet_key) OVERRIDE {
   }
 
  private:
@@ -127,8 +127,8 @@ class RtcpPeer : public Rtcp {
  public:
   RtcpPeer(scoped_refptr<CastEnvironment> cast_environment,
            RtcpSenderFeedback* sender_feedback,
-           transport::CastTransportSender* const transport_sender,
-           transport::PacedPacketSender* paced_packet_sender,
+           CastTransportSender* const transport_sender,
+           PacedPacketSender* paced_packet_sender,
            RtpReceiverStatistics* rtp_receiver_statistics,
            RtcpMode rtcp_mode,
            const base::TimeDelta& rtcp_interval,
@@ -165,27 +165,28 @@ class RtcpTest : public ::testing::Test {
         receiver_to_sender_(cast_environment_, testing_clock_) {
     testing_clock_->Advance(base::TimeTicks::Now() - base::TimeTicks());
     net::IPEndPoint dummy_endpoint;
-    transport_sender_.reset(new transport::CastTransportSenderImpl(
+    transport_sender_.reset(new CastTransportSenderImpl(
         NULL,
         testing_clock_,
         dummy_endpoint,
         base::Bind(&UpdateCastTransportStatus),
-        transport::BulkRawEventsCallback(),
+        BulkRawEventsCallback(),
         base::TimeDelta(),
         task_runner_,
         &sender_to_receiver_));
-    transport::CastTransportAudioConfig config;
-    config.rtp.config.ssrc = kSenderSsrc;
-    config.rtp.max_outstanding_frames = 1;
+    CastTransportRtpConfig config;
+    config.ssrc = kSenderSsrc;
+    config.rtp_payload_type = 127;
+    config.stored_frames = 1;
     transport_sender_->InitializeAudio(config);
     EXPECT_CALL(mock_sender_feedback_, OnReceivedCastFeedback(_)).Times(0);
   }
 
   virtual ~RtcpTest() {}
 
-  static void UpdateCastTransportStatus(transport::CastTransportStatus status) {
-    bool result = (status == transport::TRANSPORT_AUDIO_INITIALIZED ||
-                   status == transport::TRANSPORT_VIDEO_INITIALIZED);
+  static void UpdateCastTransportStatus(CastTransportStatus status) {
+    bool result = (status == TRANSPORT_AUDIO_INITIALIZED ||
+                   status == TRANSPORT_VIDEO_INITIALIZED);
     EXPECT_TRUE(result);
   }
 
@@ -201,7 +202,7 @@ class RtcpTest : public ::testing::Test {
   scoped_refptr<test::FakeSingleThreadTaskRunner> task_runner_;
   scoped_refptr<CastEnvironment> cast_environment_;
   RtcpTestPacketSender sender_to_receiver_;
-  scoped_ptr<transport::CastTransportSenderImpl> transport_sender_;
+  scoped_ptr<CastTransportSenderImpl> transport_sender_;
   LocalRtcpTransport receiver_to_sender_;
   MockRtcpSenderFeedback mock_sender_feedback_;
 
diff --git a/media/cast/rtcp/rtcp_utility.cc b/media/cast/net/rtcp/rtcp_utility.cc
index e29f82e9cf..91fccc1968 100644
--- a/media/cast/rtcp/rtcp_utility.cc
+++ b/media/cast/net/rtcp/rtcp_utility.cc
@@ -1,12 +1,12 @@
-// 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/rtcp/rtcp_utility.h"
+#include "media/cast/net/rtcp/rtcp_utility.h"
 
 #include "base/big_endian.h"
 #include "base/logging.h"
-#include "media/cast/transport/cast_transport_defines.h"
+#include "media/cast/net/cast_transport_defines.h"
 
 namespace media {
 namespace cast {
@@ -106,44 +106,44 @@ void RtcpParser::IterateTopLevel() {
       return;  // Bad block!
 
     switch (header.PT) {
-      case transport::kPacketTypeSenderReport:
+      case kPacketTypeSenderReport:
         // number of Report blocks
         number_of_blocks_ = header.IC;
         ParseSR();
         return;
-      case transport::kPacketTypeReceiverReport:
+      case kPacketTypeReceiverReport:
         // number of Report blocks
         number_of_blocks_ = header.IC;
         ParseRR();
         return;
-      case transport::kPacketTypeSdes:
+      case kPacketTypeSdes:
         // number of Sdes blocks
         number_of_blocks_ = header.IC;
         if (!ParseSdes()) {
           break;  // Nothing supported found, continue to next block!
         }
         return;
-      case transport::kPacketTypeBye:
+      case kPacketTypeBye:
         number_of_blocks_ = header.IC;
         if (!ParseBye()) {
           // Nothing supported found, continue to next block!
           break;
         }
         return;
-      case transport::kPacketTypeApplicationDefined:
+      case kPacketTypeApplicationDefined:
         if (!ParseApplicationDefined(header.IC)) {
           // Nothing supported found, continue to next block!
           break;
         }
         return;
-      case transport::kPacketTypeGenericRtpFeedback:  // Fall through!
-      case transport::kPacketTypePayloadSpecific:
+      case kPacketTypeGenericRtpFeedback:  // Fall through!
+      case kPacketTypePayloadSpecific:
         if (!ParseFeedBackCommon(header)) {
           // Nothing supported found, continue to next block!
           break;
         }
         return;
-      case transport::kPacketTypeXr:
+      case kPacketTypeXr:
         if (!ParseExtendedReport()) {
           break;  // Nothing supported found, continue to next block!
         }
@@ -609,8 +609,8 @@ bool RtcpParser::ParseCastReceiverLogEventItem() {
 }
 
 bool RtcpParser::ParseFeedBackCommon(const RtcpCommonHeader& header) {
-  DCHECK((header.PT == transport::kPacketTypeGenericRtpFeedback) ||
-         (header.PT == transport::kPacketTypePayloadSpecific))
+  DCHECK((header.PT == kPacketTypeGenericRtpFeedback) ||
+         (header.PT == kPacketTypePayloadSpecific))
       << "Invalid state";
 
   ptrdiff_t length = rtcp_block_end_ - rtcp_data_;
@@ -630,7 +630,7 @@ bool RtcpParser::ParseFeedBackCommon(const RtcpCommonHeader& header) {
 
   rtcp_data_ += 12;
 
-  if (header.PT == transport::kPacketTypeGenericRtpFeedback) {
+  if (header.PT == kPacketTypeGenericRtpFeedback) {
     // Transport layer feedback
     switch (header.IC) {
       case 1:
@@ -663,7 +663,7 @@ bool RtcpParser::ParseFeedBackCommon(const RtcpCommonHeader& header) {
     EndCurrentBlock();
     return false;
 
-  } else if (header.PT == transport::kPacketTypePayloadSpecific) {
+  } else if (header.PT == kPacketTypePayloadSpecific) {
     // Payload specific feedback
     switch (header.IC) {
       case 1:
diff --git a/media/cast/rtcp/rtcp_utility.h b/media/cast/net/rtcp/rtcp_utility.h
index 34f3f25a88..bf6717071a 100644
--- a/media/cast/rtcp/rtcp_utility.h
+++ b/media/cast/net/rtcp/rtcp_utility.h
@@ -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.
 
@@ -8,7 +8,7 @@
 #include "media/cast/cast_config.h"
 #include "media/cast/cast_defines.h"
 #include "media/cast/logging/logging_defines.h"
-#include "media/cast/rtcp/rtcp_defines.h"
+#include "media/cast/net/rtcp/rtcp_defines.h"
 
 namespace media {
 namespace cast {
diff --git a/media/cast/rtcp/test_rtcp_packet_builder.cc b/media/cast/net/rtcp/test_rtcp_packet_builder.cc
index 8d0809d928..b6d7cf208e 100644
--- a/media/cast/rtcp/test_rtcp_packet_builder.cc
+++ b/media/cast/net/rtcp/test_rtcp_packet_builder.cc
@@ -1,11 +1,11 @@
-// 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/rtcp/test_rtcp_packet_builder.h"
+#include "media/cast/net/rtcp/test_rtcp_packet_builder.h"
 
 #include "base/logging.h"
-#include "media/cast/rtcp/rtcp_utility.h"
+#include "media/cast/net/rtcp/rtcp_utility.h"
 
 namespace media {
 namespace cast {
@@ -65,21 +65,7 @@ void TestRtcpPacketBuilder::AddSdesCname(uint32 sender_ssrc,
   for (size_t i = 0; i < c_name.size(); ++i) {
     big_endian_writer_.WriteU8(c_name.c_str()[i]);
   }
-  int padding;
-  switch (c_name.size() % 4) {
-    case 0:
-      padding = 2;
-      break;
-    case 1:
-      padding = 1;
-      break;
-    case 2:
-      padding = 4;
-      break;
-    case 3:
-      padding = 3;
-      break;
-  }
+  const int padding = 4 - ((c_name.size() + 2) % 4);
   for (int j = 0; j < padding; ++j) {
     big_endian_writer_.WriteU8(0);
   }
@@ -191,7 +177,7 @@ void TestRtcpPacketBuilder::AddRemb(uint32 sender_ssrc, uint32 media_ssrc) {
 
 void TestRtcpPacketBuilder::AddCast(uint32 sender_ssrc,
                                     uint32 media_ssrc,
-                                    uint16 target_delay_ms) {
+                                    base::TimeDelta target_delay) {
   AddRtcpHeader(206, 15);
   big_endian_writer_.WriteU32(sender_ssrc);
   big_endian_writer_.WriteU32(media_ssrc);
@@ -201,7 +187,7 @@ void TestRtcpPacketBuilder::AddCast(uint32 sender_ssrc,
   big_endian_writer_.WriteU8('T');
   big_endian_writer_.WriteU8(kAckFrameId);
   big_endian_writer_.WriteU8(3);     // Loss fields.
-  big_endian_writer_.WriteU16(target_delay_ms);
+  big_endian_writer_.WriteU16(target_delay.InMilliseconds());
   big_endian_writer_.WriteU8(kLostFrameId);
   big_endian_writer_.WriteU16(kRtcpCastAllPacketsLost);
   big_endian_writer_.WriteU8(0);  // Lost packet id mask.
diff --git a/media/cast/rtcp/test_rtcp_packet_builder.h b/media/cast/net/rtcp/test_rtcp_packet_builder.h
index d4266670ab..153ced1fd6 100644
--- a/media/cast/rtcp/test_rtcp_packet_builder.h
+++ b/media/cast/net/rtcp/test_rtcp_packet_builder.h
@@ -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.
 
@@ -9,7 +9,8 @@
 
 #include "base/big_endian.h"
 #include "media/cast/cast_config.h"
-#include "media/cast/rtcp/rtcp_defines.h"
+#include "media/cast/net/cast_transport_defines.h"
+#include "media/cast/net/rtcp/rtcp_defines.h"
 
 namespace media {
 namespace cast {
@@ -79,7 +80,9 @@ class TestRtcpPacketBuilder {
   void AddPli(uint32 sender_ssrc, uint32 media_ssrc);
   void AddRpsi(uint32 sender_ssrc, uint32 media_ssrc);
   void AddRemb(uint32 sender_ssrc, uint32 media_ssrc);
-  void AddCast(uint32 sender_ssrc, uint32 media_ssrc, uint16 target_delay_ms);
+  void AddCast(uint32 sender_ssrc,
+               uint32 media_ssrc,
+               base::TimeDelta target_delay);
   void AddReceiverLog(uint32 sender_ssrc);
   void AddReceiverFrameLog(uint32 rtp_timestamp,
                            int num_events,
diff --git a/media/cast/framer/cast_message_builder.cc b/media/cast/net/rtp/cast_message_builder.cc
index f3473f9690..bb36613d2c 100644
--- a/media/cast/framer/cast_message_builder.cc
+++ b/media/cast/net/rtp/cast_message_builder.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/framer/cast_message_builder.h"
+#include "media/cast/net/rtp/cast_message_builder.h"
 
 #include "media/cast/cast_defines.h"
 
diff --git a/media/cast/framer/cast_message_builder.h b/media/cast/net/rtp/cast_message_builder.h
index 9db88d4a99..0136fed4a0 100644
--- a/media/cast/framer/cast_message_builder.h
+++ b/media/cast/net/rtp/cast_message_builder.h
@@ -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.
 
@@ -10,9 +10,9 @@
 #include <deque>
 #include <map>
 
-#include "media/cast/framer/frame_id_map.h"
-#include "media/cast/rtcp/rtcp.h"
-#include "media/cast/rtp_receiver/rtp_receiver_defines.h"
+#include "media/cast/net/rtcp/rtcp.h"
+#include "media/cast/net/rtp/frame_id_map.h"
+#include "media/cast/net/rtp/rtp_receiver_defines.h"
 
 namespace media {
 namespace cast {
diff --git a/media/cast/framer/cast_message_builder_unittest.cc b/media/cast/net/rtp/cast_message_builder_unittest.cc
index ef75162a08..b0e30c44cd 100644
--- a/media/cast/framer/cast_message_builder_unittest.cc
+++ b/media/cast/net/rtp/cast_message_builder_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.
 
@@ -6,9 +6,9 @@
 
 #include "base/memory/scoped_ptr.h"
 #include "base/test/simple_test_tick_clock.h"
-#include "media/cast/framer/cast_message_builder.h"
-#include "media/cast/rtcp/rtcp.h"
-#include "media/cast/rtp_receiver/rtp_receiver_defines.h"
+#include "media/cast/net/rtcp/rtcp.h"
+#include "media/cast/net/rtp/cast_message_builder.h"
+#include "media/cast/net/rtp/rtp_receiver_defines.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
 namespace media {
diff --git a/media/cast/framer/frame_buffer.cc b/media/cast/net/rtp/frame_buffer.cc
index 0b6fa8332c..a419ab6a46 100644
--- a/media/cast/framer/frame_buffer.cc
+++ b/media/cast/net/rtp/frame_buffer.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/framer/frame_buffer.h"
+#include "media/cast/net/rtp/frame_buffer.h"
 
 #include "base/logging.h"
 
@@ -59,17 +59,17 @@ bool FrameBuffer::Complete() const {
   return num_packets_received_ - 1 == max_packet_id_;
 }
 
-bool FrameBuffer::AssembleEncodedFrame(transport::EncodedFrame* frame) const {
+bool FrameBuffer::AssembleEncodedFrame(EncodedFrame* frame) const {
   if (!Complete())
     return false;
 
   // Frame is complete -> construct.
   if (is_key_frame_)
-    frame->dependency = transport::EncodedFrame::KEY;
+    frame->dependency = EncodedFrame::KEY;
   else if (frame_id_ == last_referenced_frame_id_)
-    frame->dependency = transport::EncodedFrame::INDEPENDENT;
+    frame->dependency = EncodedFrame::INDEPENDENT;
   else
-    frame->dependency = transport::EncodedFrame::DEPENDENT;
+    frame->dependency = EncodedFrame::DEPENDENT;
   frame->frame_id = frame_id_;
   frame->referenced_frame_id = last_referenced_frame_id_;
   frame->rtp_timestamp = rtp_timestamp_;
diff --git a/media/cast/framer/frame_buffer.h b/media/cast/net/rtp/frame_buffer.h
index d4d5dedbbd..1309a8785a 100644
--- a/media/cast/framer/frame_buffer.h
+++ b/media/cast/net/rtp/frame_buffer.h
@@ -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.
 
@@ -9,7 +9,7 @@
 #include <vector>
 
 #include "media/cast/cast_config.h"
-#include "media/cast/rtp_receiver/rtp_receiver_defines.h"
+#include "media/cast/net/rtp/rtp_receiver_defines.h"
 
 namespace media {
 namespace cast {
@@ -29,7 +29,7 @@ class FrameBuffer {
   // and also copies the data from all packets into the data field in |frame|.
   // Returns true if the frame was complete; false if incomplete and |frame|
   // remains unchanged.
-  bool AssembleEncodedFrame(transport::EncodedFrame* frame) const;
+  bool AssembleEncodedFrame(EncodedFrame* frame) const;
 
   bool is_key_frame() const { return is_key_frame_; }
 
diff --git a/media/cast/framer/frame_buffer_unittest.cc b/media/cast/net/rtp/frame_buffer_unittest.cc
index d6844f3e95..69ce93dbbb 100644
--- a/media/cast/framer/frame_buffer_unittest.cc
+++ b/media/cast/net/rtp/frame_buffer_unittest.cc
@@ -1,8 +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.
 
-#include "media/cast/framer/frame_buffer.h"
+#include "media/cast/net/rtp/frame_buffer.h"
+#include "media/cast/net/cast_transport_defines.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
 namespace media {
@@ -29,9 +30,9 @@ TEST_F(FrameBufferTest, OnePacketInsertSanity) {
   rtp_header_.frame_id = 5;
   rtp_header_.reference_frame_id = 5;
   buffer_.InsertPacket(payload_.data(), payload_.size(), rtp_header_);
-  transport::EncodedFrame frame;
+  EncodedFrame frame;
   EXPECT_TRUE(buffer_.AssembleEncodedFrame(&frame));
-  EXPECT_EQ(transport::EncodedFrame::KEY, frame.dependency);
+  EXPECT_EQ(EncodedFrame::KEY, frame.dependency);
   EXPECT_EQ(5u, frame.frame_id);
   EXPECT_EQ(5u, frame.referenced_frame_id);
   EXPECT_EQ(3000u, frame.rtp_timestamp);
@@ -39,7 +40,7 @@ TEST_F(FrameBufferTest, OnePacketInsertSanity) {
 
 TEST_F(FrameBufferTest, EmptyBuffer) {
   EXPECT_FALSE(buffer_.Complete());
-  transport::EncodedFrame frame;
+  EncodedFrame frame;
   EXPECT_FALSE(buffer_.AssembleEncodedFrame(&frame));
 }
 
@@ -47,7 +48,7 @@ TEST_F(FrameBufferTest, DefaultOnePacketFrame) {
   buffer_.InsertPacket(payload_.data(), payload_.size(), rtp_header_);
   EXPECT_TRUE(buffer_.Complete());
   EXPECT_FALSE(buffer_.is_key_frame());
-  transport::EncodedFrame frame;
+  EncodedFrame frame;
   EXPECT_TRUE(buffer_.AssembleEncodedFrame(&frame));
   EXPECT_EQ(payload_.size(), frame.data.size());
 }
@@ -63,7 +64,7 @@ TEST_F(FrameBufferTest, MultiplePacketFrame) {
   ++rtp_header_.packet_id;
   EXPECT_TRUE(buffer_.Complete());
   EXPECT_TRUE(buffer_.is_key_frame());
-  transport::EncodedFrame frame;
+  EncodedFrame frame;
   EXPECT_TRUE(buffer_.AssembleEncodedFrame(&frame));
   EXPECT_EQ(3 * payload_.size(), frame.data.size());
 }
diff --git a/media/cast/framer/frame_id_map.cc b/media/cast/net/rtp/frame_id_map.cc
index b4389fd532..f0b433c2fb 100644
--- a/media/cast/framer/frame_id_map.cc
+++ b/media/cast/net/rtp/frame_id_map.cc
@@ -1,11 +1,11 @@
-// 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/framer/frame_id_map.h"
+#include "media/cast/net/rtp/frame_id_map.h"
 
 #include "base/logging.h"
-#include "media/cast/rtp_receiver/rtp_receiver_defines.h"
+#include "media/cast/net/rtp/rtp_receiver_defines.h"
 
 namespace media {
 namespace cast {
diff --git a/media/cast/framer/frame_id_map.h b/media/cast/net/rtp/frame_id_map.h
index 66e306f671..9c1b674ff4 100644
--- a/media/cast/framer/frame_id_map.h
+++ b/media/cast/net/rtp/frame_id_map.h
@@ -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.
 
@@ -11,8 +11,8 @@
 #include "base/memory/linked_ptr.h"
 #include "base/memory/scoped_ptr.h"
 #include "media/cast/cast_config.h"
-#include "media/cast/rtcp/rtcp_defines.h"
-#include "media/cast/rtp_receiver/rtp_receiver_defines.h"
+#include "media/cast/net/rtcp/rtcp_defines.h"
+#include "media/cast/net/rtp/rtp_receiver_defines.h"
 
 namespace media {
 namespace cast {
diff --git a/media/cast/framer/framer.cc b/media/cast/net/rtp/framer.cc
index de4451a3b4..c94dc0c712 100644
--- a/media/cast/framer/framer.cc
+++ b/media/cast/net/rtp/framer.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/framer/framer.h"
+#include "media/cast/net/rtp/framer.h"
 
 #include "base/logging.h"
 
@@ -62,7 +62,7 @@ bool Framer::InsertPacket(const uint8* payload_data,
 }
 
 // This does not release the frame.
-bool Framer::GetEncodedFrame(transport::EncodedFrame* frame,
+bool Framer::GetEncodedFrame(EncodedFrame* frame,
                              bool* next_frame,
                              bool* have_multiple_decodable_frames) {
   *have_multiple_decodable_frames = frame_id_map_.HaveMultipleDecodableFrames();
diff --git a/media/cast/framer/framer.h b/media/cast/net/rtp/framer.h
index 0b7249eff3..cf70ef191e 100644
--- a/media/cast/framer/framer.h
+++ b/media/cast/net/rtp/framer.h
@@ -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.
 
@@ -12,11 +12,11 @@
 #include "base/memory/scoped_ptr.h"
 #include "base/time/tick_clock.h"
 #include "base/time/time.h"
-#include "media/cast/framer/cast_message_builder.h"
-#include "media/cast/framer/frame_buffer.h"
-#include "media/cast/framer/frame_id_map.h"
-#include "media/cast/rtcp/rtcp.h"
-#include "media/cast/rtp_receiver/rtp_receiver_defines.h"
+#include "media/cast/net/rtcp/rtcp.h"
+#include "media/cast/net/rtp/cast_message_builder.h"
+#include "media/cast/net/rtp/frame_buffer.h"
+#include "media/cast/net/rtp/frame_id_map.h"
+#include "media/cast/net/rtp/rtp_receiver_defines.h"
 
 namespace media {
 namespace cast {
@@ -45,7 +45,7 @@ class Framer {
   // |next_frame| will be set to true if the returned frame is the very
   // next frame. |have_multiple_complete_frames| will be set to true
   // if there are more decodadble frames available.
-  bool GetEncodedFrame(transport::EncodedFrame* video_frame,
+  bool GetEncodedFrame(EncodedFrame* video_frame,
                        bool* next_frame,
                        bool* have_multiple_complete_frames);
 
diff --git a/media/cast/framer/framer_unittest.cc b/media/cast/net/rtp/framer_unittest.cc
index ad53ef06ee..95e60d438d 100644
--- a/media/cast/framer/framer_unittest.cc
+++ b/media/cast/net/rtp/framer_unittest.cc
@@ -1,10 +1,11 @@
-// 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 "base/test/simple_test_tick_clock.h"
-#include "media/cast/framer/framer.h"
-#include "media/cast/rtp_receiver/mock_rtp_payload_feedback.h"
+#include "media/cast/net/cast_transport_defines.h"
+#include "media/cast/net/rtp/framer.h"
+#include "media/cast/net/rtp/mock_rtp_payload_feedback.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
 namespace media {
@@ -33,14 +34,14 @@ class FramerTest : public ::testing::Test {
 };
 
 TEST_F(FramerTest, EmptyState) {
-  transport::EncodedFrame frame;
+  EncodedFrame frame;
   bool next_frame = false;
   bool multiple = false;
   EXPECT_FALSE(framer_.GetEncodedFrame(&frame, &next_frame, &multiple));
 }
 
 TEST_F(FramerTest, AlwaysStartWithKey) {
-  transport::EncodedFrame frame;
+  EncodedFrame frame;
   bool next_frame = false;
   bool complete = false;
   bool multiple = false;
@@ -60,14 +61,14 @@ TEST_F(FramerTest, AlwaysStartWithKey) {
   EXPECT_TRUE(framer_.GetEncodedFrame(&frame, &next_frame, &multiple));
   EXPECT_TRUE(next_frame);
   EXPECT_TRUE(multiple);
-  EXPECT_EQ(transport::EncodedFrame::KEY, frame.dependency);
+  EXPECT_EQ(EncodedFrame::KEY, frame.dependency);
   EXPECT_EQ(1u, frame.frame_id);
   EXPECT_EQ(1u, frame.referenced_frame_id);
   framer_.ReleaseFrame(frame.frame_id);
 }
 
 TEST_F(FramerTest, CompleteFrame) {
-  transport::EncodedFrame frame;
+  EncodedFrame frame;
   bool next_frame = false;
   bool complete = false;
   bool multiple = false;
@@ -81,7 +82,7 @@ TEST_F(FramerTest, CompleteFrame) {
   EXPECT_TRUE(framer_.GetEncodedFrame(&frame, &next_frame, &multiple));
   EXPECT_TRUE(next_frame);
   EXPECT_FALSE(multiple);
-  EXPECT_EQ(transport::EncodedFrame::KEY, frame.dependency);
+  EXPECT_EQ(EncodedFrame::KEY, frame.dependency);
   EXPECT_EQ(0u, frame.frame_id);
   EXPECT_EQ(0u, frame.referenced_frame_id);
   framer_.ReleaseFrame(frame.frame_id);
@@ -107,7 +108,7 @@ TEST_F(FramerTest, CompleteFrame) {
 }
 
 TEST_F(FramerTest, DuplicatePackets) {
-  transport::EncodedFrame frame;
+  EncodedFrame frame;
   bool next_frame = false;
   bool complete = false;
   bool multiple = false;
@@ -139,7 +140,7 @@ TEST_F(FramerTest, DuplicatePackets) {
   EXPECT_TRUE(complete);
   EXPECT_FALSE(duplicate);
   EXPECT_TRUE(framer_.GetEncodedFrame(&frame, &next_frame, &multiple));
-  EXPECT_EQ(transport::EncodedFrame::KEY, frame.dependency);
+  EXPECT_EQ(EncodedFrame::KEY, frame.dependency);
   EXPECT_FALSE(multiple);
   EXPECT_EQ(0u, frame.referenced_frame_id);
 
@@ -150,7 +151,7 @@ TEST_F(FramerTest, DuplicatePackets) {
   EXPECT_FALSE(complete);
   EXPECT_TRUE(duplicate);
   EXPECT_TRUE(framer_.GetEncodedFrame(&frame, &next_frame, &multiple));
-  EXPECT_EQ(transport::EncodedFrame::KEY, frame.dependency);
+  EXPECT_EQ(EncodedFrame::KEY, frame.dependency);
   EXPECT_EQ(0u, frame.frame_id);
   EXPECT_FALSE(multiple);
   EXPECT_EQ(0u, frame.referenced_frame_id);
@@ -184,7 +185,7 @@ TEST_F(FramerTest, DuplicatePackets) {
   EXPECT_TRUE(complete);
   EXPECT_FALSE(duplicate);
   EXPECT_TRUE(framer_.GetEncodedFrame(&frame, &next_frame, &multiple));
-  EXPECT_EQ(transport::EncodedFrame::DEPENDENT, frame.dependency);
+  EXPECT_EQ(EncodedFrame::DEPENDENT, frame.dependency);
   EXPECT_EQ(1u, frame.frame_id);
   EXPECT_EQ(0u, frame.referenced_frame_id);
   EXPECT_FALSE(multiple);
@@ -196,14 +197,14 @@ TEST_F(FramerTest, DuplicatePackets) {
   EXPECT_FALSE(complete);
   EXPECT_TRUE(duplicate);
   EXPECT_TRUE(framer_.GetEncodedFrame(&frame, &next_frame, &multiple));
-  EXPECT_EQ(transport::EncodedFrame::DEPENDENT, frame.dependency);
+  EXPECT_EQ(EncodedFrame::DEPENDENT, frame.dependency);
   EXPECT_EQ(1u, frame.frame_id);
   EXPECT_EQ(0u, frame.referenced_frame_id);
   EXPECT_FALSE(multiple);
 }
 
 TEST_F(FramerTest, ContinuousSequence) {
-  transport::EncodedFrame frame;
+  EncodedFrame frame;
   bool next_frame = false;
   bool complete = false;
   bool multiple = false;
@@ -217,7 +218,7 @@ TEST_F(FramerTest, ContinuousSequence) {
   EXPECT_TRUE(framer_.GetEncodedFrame(&frame, &next_frame, &multiple));
   EXPECT_TRUE(next_frame);
   EXPECT_FALSE(multiple);
-  EXPECT_EQ(transport::EncodedFrame::KEY, frame.dependency);
+  EXPECT_EQ(EncodedFrame::KEY, frame.dependency);
   EXPECT_EQ(0u, frame.frame_id);
   EXPECT_EQ(0u, frame.referenced_frame_id);
   framer_.ReleaseFrame(frame.frame_id);
@@ -234,7 +235,7 @@ TEST_F(FramerTest, ContinuousSequence) {
 
 TEST_F(FramerTest, Wrap) {
   // Insert key frame, frame_id = 255 (will jump to that)
-  transport::EncodedFrame frame;
+  EncodedFrame frame;
   bool next_frame = false;
   bool multiple = true;
   bool duplicate = false;
@@ -248,7 +249,7 @@ TEST_F(FramerTest, Wrap) {
   EXPECT_TRUE(framer_.GetEncodedFrame(&frame, &next_frame, &multiple));
   EXPECT_TRUE(next_frame);
   EXPECT_FALSE(multiple);
-  EXPECT_EQ(transport::EncodedFrame::KEY, frame.dependency);
+  EXPECT_EQ(EncodedFrame::KEY, frame.dependency);
   EXPECT_EQ(255u, frame.frame_id);
   EXPECT_EQ(255u, frame.referenced_frame_id);
   framer_.ReleaseFrame(frame.frame_id);
@@ -261,14 +262,14 @@ TEST_F(FramerTest, Wrap) {
   EXPECT_TRUE(framer_.GetEncodedFrame(&frame, &next_frame, &multiple));
   EXPECT_TRUE(next_frame);
   EXPECT_FALSE(multiple);
-  EXPECT_EQ(transport::EncodedFrame::DEPENDENT, frame.dependency);
+  EXPECT_EQ(EncodedFrame::DEPENDENT, frame.dependency);
   EXPECT_EQ(256u, frame.frame_id);
   EXPECT_EQ(255u, frame.referenced_frame_id);
   framer_.ReleaseFrame(frame.frame_id);
 }
 
 TEST_F(FramerTest, Reset) {
-  transport::EncodedFrame frame;
+  EncodedFrame frame;
   bool next_frame = false;
   bool complete = false;
   bool multiple = true;
@@ -284,7 +285,7 @@ TEST_F(FramerTest, Reset) {
 }
 
 TEST_F(FramerTest, RequireKeyAfterReset) {
-  transport::EncodedFrame frame;
+  EncodedFrame frame;
   bool next_frame = false;
   bool multiple = false;
   bool duplicate = false;
@@ -308,7 +309,7 @@ TEST_F(FramerTest, RequireKeyAfterReset) {
 }
 
 TEST_F(FramerTest, BasicNonLastReferenceId) {
-  transport::EncodedFrame frame;
+  EncodedFrame frame;
   bool next_frame = false;
   bool multiple = false;
   bool duplicate = false;
@@ -335,7 +336,7 @@ TEST_F(FramerTest, BasicNonLastReferenceId) {
 
 TEST_F(FramerTest, InOrderReferenceFrameSelection) {
   // Create pattern: 0, 1, 4, 5.
-  transport::EncodedFrame frame;
+  EncodedFrame frame;
   bool next_frame = false;
   bool multiple = false;
   bool duplicate = false;
@@ -360,7 +361,7 @@ TEST_F(FramerTest, InOrderReferenceFrameSelection) {
   framer_.InsertPacket(
       payload_.data(), payload_.size(), rtp_header_, &duplicate);
   EXPECT_TRUE(framer_.GetEncodedFrame(&frame, &next_frame, &multiple));
-  EXPECT_EQ(transport::EncodedFrame::KEY, frame.dependency);
+  EXPECT_EQ(EncodedFrame::KEY, frame.dependency);
   EXPECT_EQ(0u, frame.frame_id);
   EXPECT_EQ(0u, frame.referenced_frame_id);
   EXPECT_FALSE(multiple);
@@ -368,14 +369,14 @@ TEST_F(FramerTest, InOrderReferenceFrameSelection) {
   EXPECT_TRUE(framer_.GetEncodedFrame(&frame, &next_frame, &multiple));
   EXPECT_TRUE(next_frame);
   EXPECT_TRUE(multiple);
-  EXPECT_EQ(transport::EncodedFrame::DEPENDENT, frame.dependency);
+  EXPECT_EQ(EncodedFrame::DEPENDENT, frame.dependency);
   EXPECT_EQ(1u, frame.frame_id);
   EXPECT_EQ(0u, frame.referenced_frame_id);
   framer_.ReleaseFrame(frame.frame_id);
   EXPECT_TRUE(framer_.GetEncodedFrame(&frame, &next_frame, &multiple));
   EXPECT_FALSE(next_frame);
   EXPECT_FALSE(multiple);
-  EXPECT_EQ(transport::EncodedFrame::DEPENDENT, frame.dependency);
+  EXPECT_EQ(EncodedFrame::DEPENDENT, frame.dependency);
   EXPECT_EQ(4u, frame.frame_id);
   EXPECT_EQ(0u, frame.referenced_frame_id);
   framer_.ReleaseFrame(frame.frame_id);
@@ -394,14 +395,14 @@ TEST_F(FramerTest, InOrderReferenceFrameSelection) {
   EXPECT_TRUE(framer_.GetEncodedFrame(&frame, &next_frame, &multiple));
   EXPECT_TRUE(next_frame);
   EXPECT_FALSE(multiple);
-  EXPECT_EQ(transport::EncodedFrame::DEPENDENT, frame.dependency);
+  EXPECT_EQ(EncodedFrame::DEPENDENT, frame.dependency);
   EXPECT_EQ(5u, frame.frame_id);
   EXPECT_EQ(4u, frame.referenced_frame_id);
 }
 
 TEST_F(FramerTest, AudioWrap) {
   // All audio frames are marked as key frames.
-  transport::EncodedFrame frame;
+  EncodedFrame frame;
   bool next_frame = false;
   bool multiple = false;
   bool duplicate = false;
@@ -415,7 +416,7 @@ TEST_F(FramerTest, AudioWrap) {
   EXPECT_TRUE(framer_.GetEncodedFrame(&frame, &next_frame, &multiple));
   EXPECT_TRUE(next_frame);
   EXPECT_FALSE(multiple);
-  EXPECT_EQ(transport::EncodedFrame::KEY, frame.dependency);
+  EXPECT_EQ(EncodedFrame::KEY, frame.dependency);
   EXPECT_EQ(254u, frame.frame_id);
   EXPECT_EQ(254u, frame.referenced_frame_id);
   framer_.ReleaseFrame(frame.frame_id);
@@ -434,7 +435,7 @@ TEST_F(FramerTest, AudioWrap) {
   EXPECT_TRUE(framer_.GetEncodedFrame(&frame, &next_frame, &multiple));
   EXPECT_TRUE(next_frame);
   EXPECT_TRUE(multiple);
-  EXPECT_EQ(transport::EncodedFrame::KEY, frame.dependency);
+  EXPECT_EQ(EncodedFrame::KEY, frame.dependency);
   EXPECT_EQ(255u, frame.frame_id);
   EXPECT_EQ(255u, frame.referenced_frame_id);
   framer_.ReleaseFrame(frame.frame_id);
@@ -442,7 +443,7 @@ TEST_F(FramerTest, AudioWrap) {
   EXPECT_TRUE(framer_.GetEncodedFrame(&frame, &next_frame, &multiple));
   EXPECT_TRUE(next_frame);
   EXPECT_FALSE(multiple);
-  EXPECT_EQ(transport::EncodedFrame::KEY, frame.dependency);
+  EXPECT_EQ(EncodedFrame::KEY, frame.dependency);
   EXPECT_EQ(256u, frame.frame_id);
   EXPECT_EQ(256u, frame.referenced_frame_id);
   framer_.ReleaseFrame(frame.frame_id);
@@ -450,7 +451,7 @@ TEST_F(FramerTest, AudioWrap) {
 
 TEST_F(FramerTest, AudioWrapWithMissingFrame) {
   // All audio frames are marked as key frames.
-  transport::EncodedFrame frame;
+  EncodedFrame frame;
   bool next_frame = false;
   bool multiple = true;
   bool duplicate = false;
@@ -464,7 +465,7 @@ TEST_F(FramerTest, AudioWrapWithMissingFrame) {
   EXPECT_TRUE(framer_.GetEncodedFrame(&frame, &next_frame, &multiple));
   EXPECT_TRUE(next_frame);
   EXPECT_FALSE(multiple);
-  EXPECT_EQ(transport::EncodedFrame::KEY, frame.dependency);
+  EXPECT_EQ(EncodedFrame::KEY, frame.dependency);
   EXPECT_EQ(253u, frame.frame_id);
   EXPECT_EQ(253u, frame.referenced_frame_id);
   framer_.ReleaseFrame(frame.frame_id);
@@ -483,14 +484,14 @@ TEST_F(FramerTest, AudioWrapWithMissingFrame) {
   EXPECT_TRUE(framer_.GetEncodedFrame(&frame, &next_frame, &multiple));
   EXPECT_FALSE(next_frame);
   EXPECT_TRUE(multiple);
-  EXPECT_EQ(transport::EncodedFrame::KEY, frame.dependency);
+  EXPECT_EQ(EncodedFrame::KEY, frame.dependency);
   EXPECT_EQ(255u, frame.frame_id);
   EXPECT_EQ(255u, frame.referenced_frame_id);
   framer_.ReleaseFrame(frame.frame_id);
   EXPECT_TRUE(framer_.GetEncodedFrame(&frame, &next_frame, &multiple));
   EXPECT_TRUE(next_frame);
   EXPECT_FALSE(multiple);
-  EXPECT_EQ(transport::EncodedFrame::KEY, frame.dependency);
+  EXPECT_EQ(EncodedFrame::KEY, frame.dependency);
   EXPECT_EQ(256u, frame.frame_id);
   EXPECT_EQ(256u, frame.referenced_frame_id);
   framer_.ReleaseFrame(frame.frame_id);
diff --git a/media/cast/rtp_receiver/rtp_parser/include/mock/mock_rtp_feedback.h b/media/cast/net/rtp/mock_rtp_feedback.h
index f5edf7c43f..44624ffc71 100644
--- a/media/cast/rtp_receiver/rtp_parser/include/mock/mock_rtp_feedback.h
+++ b/media/cast/net/rtp/mock_rtp_feedback.h
@@ -1,11 +1,11 @@
-// 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_RTP_INCLUDE_MOCK_RTP_FEEDBACK_H_
 #define MEDIA_CAST_RTP_INCLUDE_MOCK_RTP_FEEDBACK_H_
 
-#include "media/cast/rtp_receiver/rtp_parser/rtp_feedback.h"
+#include "media/cast/net/rtp/rtp_parser/rtp_feedback.h"
 #include "testing/gmock/include/gmock/gmock.h"
 
 namespace media {
diff --git a/media/cast/rtp_receiver/mock_rtp_payload_feedback.cc b/media/cast/net/rtp/mock_rtp_payload_feedback.cc
index 02b6c0be45..cde8e3880a 100644
--- a/media/cast/rtp_receiver/mock_rtp_payload_feedback.cc
+++ b/media/cast/net/rtp/mock_rtp_payload_feedback.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/rtp_receiver/mock_rtp_payload_feedback.h"
+#include "media/cast/net/rtp/mock_rtp_payload_feedback.h"
 
 namespace media {
 namespace cast {
diff --git a/media/cast/rtp_receiver/mock_rtp_payload_feedback.h b/media/cast/net/rtp/mock_rtp_payload_feedback.h
index 14e48673bd..f46d318476 100644
--- a/media/cast/rtp_receiver/mock_rtp_payload_feedback.h
+++ b/media/cast/net/rtp/mock_rtp_payload_feedback.h
@@ -1,11 +1,11 @@
-// 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_RTP_COMMON_MOCK_RTP_PAYLOAD_FEEDBACK_H_
 #define MEDIA_CAST_RTP_COMMON_MOCK_RTP_PAYLOAD_FEEDBACK_H_
 
-#include "media/cast/rtp_receiver/rtp_receiver_defines.h"
+#include "media/cast/net/rtp/rtp_receiver_defines.h"
 #include "testing/gmock/include/gmock/gmock.h"
 
 namespace media {
diff --git a/media/cast/transport/rtp_sender/packet_storage/packet_storage.cc b/media/cast/net/rtp/packet_storage.cc
index a748baa27a..59ac9ce44e 100644
--- a/media/cast/transport/rtp_sender/packet_storage/packet_storage.cc
+++ b/media/cast/net/rtp/packet_storage.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/transport/rtp_sender/packet_storage/packet_storage.h"
+#include "media/cast/net/rtp/packet_storage.h"
 
 #include <string>
 
 #include "base/logging.h"
+#include "media/cast/cast_defines.h"
 
 namespace media {
 namespace cast {
-namespace transport {
 
 PacketStorage::PacketStorage(size_t stored_frames)
     : max_stored_frames_(stored_frames),
@@ -60,6 +60,5 @@ const SendPacketVector* PacketStorage::GetFrame8(uint8 frame_id_8bits) const {
   return &(frames_[index_8bits]);
 }
 
-}  // namespace transport
 }  // namespace cast
 }  // namespace media
diff --git a/media/cast/transport/rtp_sender/packet_storage/packet_storage.h b/media/cast/net/rtp/packet_storage.h
index 037ead1edf..9330a6a0ae 100644
--- a/media/cast/transport/rtp_sender/packet_storage/packet_storage.h
+++ b/media/cast/net/rtp/packet_storage.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_TRANSPORT_RTP_SENDER_PACKET_STORAGE_PACKET_STORAGE_H_
-#define MEDIA_CAST_TRANSPORT_RTP_SENDER_PACKET_STORAGE_PACKET_STORAGE_H_
+#ifndef MEDIA_CAST_NET_RTP_SENDER_PACKET_STORAGE_PACKET_STORAGE_H_
+#define MEDIA_CAST_NET_RTP_SENDER_PACKET_STORAGE_PACKET_STORAGE_H_
 
 #include <deque>
 #include <list>
@@ -15,13 +15,12 @@
 #include "base/memory/scoped_ptr.h"
 #include "base/time/tick_clock.h"
 #include "base/time/time.h"
-#include "media/cast/transport/cast_transport_config.h"
-#include "media/cast/transport/cast_transport_defines.h"
-#include "media/cast/transport/pacing/paced_sender.h"
+#include "media/cast/net/cast_transport_config.h"
+#include "media/cast/net/cast_transport_defines.h"
+#include "media/cast/net/pacing/paced_sender.h"
 
 namespace media {
 namespace cast {
-namespace transport {
 
 // Stores a list of frames. Each frame consists a list of packets.
 typedef std::deque<SendPacketVector> FrameQueue;
@@ -55,8 +54,7 @@ class PacketStorage {
   DISALLOW_COPY_AND_ASSIGN(PacketStorage);
 };
 
-}  // namespace transport
 }  // namespace cast
 }  // namespace media
 
-#endif  // MEDIA_CAST_TRANSPORT_RTP_SENDER_PACKET_STORAGE_PACKET_STORAGE_H_
+#endif  // MEDIA_CAST_NET_RTP_SENDER_PACKET_STORAGE_PACKET_STORAGE_H_
diff --git a/media/cast/transport/rtp_sender/packet_storage/packet_storage_unittest.cc b/media/cast/net/rtp/packet_storage_unittest.cc
index 298942c80a..5e9393da5d 100644
--- a/media/cast/transport/rtp_sender/packet_storage/packet_storage_unittest.cc
+++ b/media/cast/net/rtp/packet_storage_unittest.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/transport/rtp_sender/packet_storage/packet_storage.h"
+#include "media/cast/net/rtp/packet_storage.h"
 
 #include <stdint.h>
 
@@ -14,7 +14,6 @@
 
 namespace media {
 namespace cast {
-namespace transport {
 
 static size_t kStoredFrames = 10;
 
@@ -110,6 +109,5 @@ TEST(PacketStorageTest, GetFrameTooOld) {
   }
 }
 
-}  // namespace transport
 }  // namespace cast
 }  // namespace media
diff --git a/media/cast/rtp_receiver/receiver_stats.cc b/media/cast/net/rtp/receiver_stats.cc
index 7eff86763f..416cdd8ea5 100644
--- a/media/cast/rtp_receiver/receiver_stats.cc
+++ b/media/cast/net/rtp/receiver_stats.cc
@@ -1,11 +1,11 @@
-// 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/rtp_receiver/receiver_stats.h"
+#include "media/cast/net/rtp/receiver_stats.h"
 
 #include "base/logging.h"
-#include "media/cast/rtp_receiver/rtp_receiver_defines.h"
+#include "media/cast/net/rtp/rtp_receiver_defines.h"
 
 namespace media {
 namespace cast {
diff --git a/media/cast/rtp_receiver/receiver_stats.h b/media/cast/net/rtp/receiver_stats.h
index 05a067f787..9de6b22c20 100644
--- a/media/cast/rtp_receiver/receiver_stats.h
+++ b/media/cast/net/rtp/receiver_stats.h
@@ -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.
 
@@ -7,8 +7,8 @@
 
 #include "base/time/tick_clock.h"
 #include "base/time/time.h"
-#include "media/cast/rtcp/rtcp.h"
-#include "media/cast/rtp_receiver/rtp_receiver_defines.h"
+#include "media/cast/net/rtcp/rtcp.h"
+#include "media/cast/net/rtp/rtp_receiver_defines.h"
 
 namespace media {
 namespace cast {
diff --git a/media/cast/rtp_receiver/receiver_stats_unittest.cc b/media/cast/net/rtp/receiver_stats_unittest.cc
index 98059cdde7..caedf84c06 100644
--- a/media/cast/rtp_receiver/receiver_stats_unittest.cc
+++ b/media/cast/net/rtp/receiver_stats_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.
 
@@ -8,8 +8,8 @@
 
 #include "base/test/simple_test_tick_clock.h"
 #include "base/time/time.h"
-#include "media/cast/rtp_receiver/receiver_stats.h"
-#include "media/cast/rtp_receiver/rtp_receiver_defines.h"
+#include "media/cast/net/rtp/receiver_stats.h"
+#include "media/cast/net/rtp/rtp_receiver_defines.h"
 
 namespace media {
 namespace cast {
diff --git a/media/cast/transport/rtp_sender/rtp_packetizer/test/rtp_header_parser.cc b/media/cast/net/rtp/rtp_header_parser.cc
index ddadcb2cf2..2d70f9827d 100644
--- a/media/cast/transport/rtp_sender/rtp_packetizer/test/rtp_header_parser.cc
+++ b/media/cast/net/rtp/rtp_header_parser.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/transport/rtp_sender/rtp_packetizer/test/rtp_header_parser.h"
+#include "media/cast/net/rtp/rtp_header_parser.h"
 
 #include <cstddef>
 
@@ -10,7 +10,6 @@
 
 namespace media {
 namespace cast {
-namespace transport {
 
 static const uint8 kCastKeyFrameBitMask = 0x80;
 static const uint8 kCastReferenceFrameIdBitMask = 0x40;
@@ -100,6 +99,5 @@ bool RtpHeaderParser::ParseCast(RtpCastTestHeader* parsed_packet) const {
   return true;
 }
 
-}  // namespace transport
 }  // namespace cast
 }  // namespace media
diff --git a/media/cast/transport/rtp_sender/rtp_packetizer/test/rtp_header_parser.h b/media/cast/net/rtp/rtp_header_parser.h
index bb6e0fae54..fd235c2aa5 100644
--- a/media/cast/transport/rtp_sender/rtp_packetizer/test/rtp_header_parser.h
+++ b/media/cast/net/rtp/rtp_header_parser.h
@@ -1,17 +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.
 
 // Utility parser for rtp packetizer unittests
-#ifndef MEDIA_CAST_TRANSPORT_RTP_SENDER_RTP_PACKETIZER_TEST_RTP_HEADER_PARSER_H_
-#define MEDIA_CAST_TRANSPORT_RTP_SENDER_RTP_PACKETIZER_TEST_RTP_HEADER_PARSER_H_
+#ifndef MEDIA_CAST_NET_RTP_RTP_HEADER_PARSER_H_
+#define MEDIA_CAST_NET_RTP_RTP_HEADER_PARSER_H_
 
 #include "base/basictypes.h"
-#include "media/cast/transport/cast_transport_defines.h"
+#include "media/cast/net/cast_transport_defines.h"
 
 namespace media {
 namespace cast {
-namespace transport {
 
 // TODO(miu): Kill this and use RtpCastHeader instead.
 struct RtpCastTestHeader {
@@ -51,14 +50,13 @@ class RtpHeaderParser {
   const uint8* const rtp_data_begin_;
   size_t length_;
 
-  mutable transport::FrameIdWrapHelper frame_id_wrap_helper_;
-  mutable transport::FrameIdWrapHelper reference_frame_id_wrap_helper_;
+  mutable FrameIdWrapHelper frame_id_wrap_helper_;
+  mutable FrameIdWrapHelper reference_frame_id_wrap_helper_;
 
   DISALLOW_COPY_AND_ASSIGN(RtpHeaderParser);
 };
 
-}  // namespace transport
 }  // namespace cast
 }  // namespace media
 
-#endif  // MEDIA_CAST_TRANSPORT_RTP_SENDER_RTP_PACKETIZER_TEST_RTP_HEADER_PARSER_H_
+#endif  // MEDIA_CAST_NET_RTP_RTP_HEADER_PARSER_H_
diff --git a/media/cast/rtp_receiver/rtp_parser/test/rtp_packet_builder.cc b/media/cast/net/rtp/rtp_packet_builder.cc
index b8ab3baa29..2f279f4e96 100644
--- a/media/cast/rtp_receiver/rtp_parser/test/rtp_packet_builder.cc
+++ b/media/cast/net/rtp/rtp_packet_builder.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/rtp_receiver/rtp_parser/test/rtp_packet_builder.h"
+#include "media/cast/net/rtp/rtp_packet_builder.h"
 
 #include "base/big_endian.h"
 #include "base/logging.h"
diff --git a/media/cast/rtp_receiver/rtp_parser/test/rtp_packet_builder.h b/media/cast/net/rtp/rtp_packet_builder.h
index 7337798e31..3d6579720b 100644
--- a/media/cast/rtp_receiver/rtp_parser/test/rtp_packet_builder.h
+++ b/media/cast/net/rtp/rtp_packet_builder.h
@@ -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.
 
@@ -7,7 +7,7 @@
 #ifndef MEDIA_CAST_RTP_RECEIVER_RTP_PARSER_TEST_RTP_PACKET_BUILDER_H_
 #define MEDIA_CAST_RTP_RECEIVER_RTP_PARSER_TEST_RTP_PACKET_BUILDER_H_
 
-#include "media/cast/rtp_receiver/rtp_receiver_defines.h"
+#include "media/cast/net/rtp/rtp_receiver_defines.h"
 
 namespace media {
 namespace cast {
diff --git a/media/cast/transport/rtp_sender/rtp_packetizer/rtp_packetizer.cc b/media/cast/net/rtp/rtp_packetizer.cc
index d40f99f144..dcfcc8bba4 100644
--- a/media/cast/transport/rtp_sender/rtp_packetizer/rtp_packetizer.cc
+++ b/media/cast/net/rtp/rtp_packetizer.cc
@@ -1,16 +1,15 @@
-// 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/transport/rtp_sender/rtp_packetizer/rtp_packetizer.h"
+#include "media/cast/net/rtp/rtp_packetizer.h"
 
 #include "base/big_endian.h"
 #include "base/logging.h"
-#include "media/cast/transport/pacing/paced_sender.h"
+#include "media/cast/net/pacing/paced_sender.h"
 
 namespace media {
 namespace cast {
-namespace transport {
 
 static const uint16 kCommonRtpHeaderLength = 12;
 static const uint16 kCastRtpHeaderLength = 7;
@@ -19,13 +18,10 @@ static const uint8 kCastReferenceFrameIdBitMask = 0x40;
 static const uint8 kRtpMarkerBitMask = 0x80;
 
 RtpPacketizerConfig::RtpPacketizerConfig()
-    : audio(false),
-      payload_type(-1),
+    : payload_type(-1),
       max_payload_length(kMaxIpPacketSize - 28),  // Default is IP-v4/UDP.
       sequence_number(0),
-      frequency(8000),
-      ssrc(0),
-      channels(0) {}
+      ssrc(0) {}
 
 RtpPacketizerConfig::~RtpPacketizerConfig() {}
 
@@ -132,6 +128,5 @@ void RtpPacketizer::BuildCommonRTPheader(Packet* packet,
   ++sequence_number_;
 }
 
-}  // namespace transport
 }  // namespace cast
 }  // namespace media
diff --git a/media/cast/transport/rtp_sender/rtp_packetizer/rtp_packetizer.h b/media/cast/net/rtp/rtp_packetizer.h
index ebdbf01018..034a74a2a5 100644
--- a/media/cast/transport/rtp_sender/rtp_packetizer/rtp_packetizer.h
+++ b/media/cast/net/rtp/rtp_packetizer.h
@@ -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.
 
-#ifndef MEDIA_CAST_TRANSPORT_RTP_SENDER_RTP_PACKETIZER_RTP_PACKETIZER_H_
-#define MEDIA_CAST_TRANSPORT_RTP_SENDER_RTP_PACKETIZER_RTP_PACKETIZER_H_
+#ifndef MEDIA_CAST_NET_RTP_RTP_PACKETIZER_H_
+#define MEDIA_CAST_NET_RTP_RTP_PACKETIZER_H_
 
 #include <cmath>
 #include <list>
 #include <map>
 
 #include "base/time/time.h"
-#include "media/cast/transport/rtp_sender/packet_storage/packet_storage.h"
+#include "media/cast/net/rtp/packet_storage.h"
 
 namespace base {
 class TickClock;
@@ -19,8 +19,6 @@ class TickClock;
 namespace media {
 namespace cast {
 
-namespace transport {
-
 class PacedSender;
 
 struct RtpPacketizerConfig {
@@ -28,21 +26,12 @@ struct RtpPacketizerConfig {
   ~RtpPacketizerConfig();
 
   // General.
-  bool audio;
   int payload_type;
   uint16 max_payload_length;
   uint16 sequence_number;
-  int frequency;
 
   // SSRC.
   unsigned int ssrc;
-
-  // Video.
-  VideoCodec video_codec;
-
-  // Audio.
-  uint8 channels;
-  AudioCodec audio_codec;
 };
 
 // This object is only called from the main cast thread.
@@ -79,8 +68,7 @@ class RtpPacketizer {
   size_t send_octet_count_;
 };
 
-}  // namespace transport
 }  // namespace cast
 }  // namespace media
 
-#endif  // MEDIA_CAST_TRANSPORT_RTP_SENDER_RTP_PACKETIZER_RTP_PACKETIZER_H_
+#endif  // MEDIA_CAST_NET_RTP_RTP_PACKETIZER_H_
diff --git a/media/cast/transport/rtp_sender/rtp_packetizer/rtp_packetizer_unittest.cc b/media/cast/net/rtp/rtp_packetizer_unittest.cc
index 64def4ce7f..1e39e5924f 100644
--- a/media/cast/transport/rtp_sender/rtp_packetizer/rtp_packetizer_unittest.cc
+++ b/media/cast/net/rtp/rtp_packetizer_unittest.cc
@@ -1,23 +1,23 @@
-// 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/transport/rtp_sender/rtp_packetizer/rtp_packetizer.h"
+#include "media/cast/net/rtp/rtp_packetizer.h"
 
 #include <stdint.h>
 
 #include "base/memory/scoped_ptr.h"
 #include "base/test/simple_test_tick_clock.h"
+#include "media/cast/logging/logging_impl.h"
 #include "media/cast/logging/simple_event_subscriber.h"
+#include "media/cast/net/pacing/paced_sender.h"
+#include "media/cast/net/rtp/packet_storage.h"
+#include "media/cast/net/rtp/rtp_header_parser.h"
 #include "media/cast/test/fake_single_thread_task_runner.h"
-#include "media/cast/transport/pacing/paced_sender.h"
-#include "media/cast/transport/rtp_sender/packet_storage/packet_storage.h"
-#include "media/cast/transport/rtp_sender/rtp_packetizer/test/rtp_header_parser.h"
 #include "testing/gmock/include/gmock/gmock.h"
 
 namespace media {
 namespace cast {
-namespace transport {
 
 namespace {
 static const int kPayload = 127;
@@ -26,7 +26,6 @@ static const uint16 kSeqNum = 33;
 static const int kMaxPacketLength = 1500;
 static const int kSsrc = 0x12345;
 static const unsigned int kFrameSize = 5000;
-static const uint32 kStartFrameId = UINT32_C(0xffffffff);
 }
 
 class TestRtpPacketTransport : public PacketSender {
@@ -170,6 +169,5 @@ TEST_F(RtpPacketizerTest, Stats) {
   EXPECT_EQ(expected_num_of_packets, transport_->number_of_packets_received());
 }
 
-}  // namespace transport
 }  // namespace cast
 }  // namespace media
diff --git a/media/cast/rtp_receiver/rtp_parser/rtp_parser.cc b/media/cast/net/rtp/rtp_parser.cc
index f44e82dac2..bed1d7c8d8 100644
--- a/media/cast/rtp_receiver/rtp_parser/rtp_parser.cc
+++ b/media/cast/net/rtp/rtp_parser.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/rtp_receiver/rtp_parser/rtp_parser.h"
+#include "media/cast/net/rtp/rtp_parser.h"
 
 #include "base/big_endian.h"
 #include "base/logging.h"
diff --git a/media/cast/rtp_receiver/rtp_parser/rtp_parser.h b/media/cast/net/rtp/rtp_parser.h
index 35118cf144..64586d27c9 100644
--- a/media/cast/rtp_receiver/rtp_parser/rtp_parser.h
+++ b/media/cast/net/rtp/rtp_parser.h
@@ -1,12 +1,12 @@
-// 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_RTP_RECEIVER_RTP_PARSER_RTP_PARSER_H_
-#define MEDIA_CAST_RTP_RECEIVER_RTP_PARSER_RTP_PARSER_H_
+#ifndef MEDIA_CAST_NET_RTP_RTP_PARSER_H_
+#define MEDIA_CAST_NET_RTP_RTP_PARSER_H_
 
-#include "media/cast/rtp_receiver/rtp_receiver_defines.h"
-#include "media/cast/transport/cast_transport_defines.h"
+#include "media/cast/net/cast_transport_defines.h"
+#include "media/cast/net/rtp/rtp_receiver_defines.h"
 
 namespace media {
 namespace cast {
@@ -36,7 +36,7 @@ class RtpParser {
  private:
   const uint32 expected_sender_ssrc_;
   const uint8 expected_payload_type_;
-  transport::FrameIdWrapHelper frame_id_wrap_helper_;
+  FrameIdWrapHelper frame_id_wrap_helper_;
 
   DISALLOW_COPY_AND_ASSIGN(RtpParser);
 };
@@ -44,4 +44,4 @@ class RtpParser {
 }  // namespace cast
 }  // namespace media
 
-#endif  // MEDIA_CAST_RTP_RECEIVER_RTP_PARSER_RTP_PARSER_H_
+#endif  // MEDIA_CAST_NET_RTP_RTP_PARSER_H_
diff --git a/media/cast/rtp_receiver/rtp_parser/rtp_parser_unittest.cc b/media/cast/net/rtp/rtp_parser_unittest.cc
index 47c79139ff..e5c08fe60e 100644
--- a/media/cast/rtp_receiver/rtp_parser/rtp_parser_unittest.cc
+++ b/media/cast/net/rtp/rtp_parser_unittest.cc
@@ -1,12 +1,12 @@
-// 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 "base/memory/scoped_ptr.h"
 #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_defines.h"
+#include "media/cast/net/rtp/rtp_packet_builder.h"
+#include "media/cast/net/rtp/rtp_parser.h"
+#include "media/cast/net/rtp/rtp_receiver_defines.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
 namespace media {
@@ -39,7 +39,7 @@ class RtpParserTest : public ::testing::Test {
   void ExpectParsesPacket() {
     RtpCastHeader parsed_header;
     const uint8* payload = NULL;
-    size_t payload_size = -1;
+    size_t payload_size = static_cast<size_t>(-1);
     EXPECT_TRUE(rtp_parser_.ParsePacket(
         packet_, kPacketLength, &parsed_header, &payload, &payload_size));
 
@@ -63,7 +63,7 @@ class RtpParserTest : public ::testing::Test {
   void ExpectDoesNotParsePacket() {
     RtpCastHeader parsed_header;
     const uint8* payload = NULL;
-    size_t payload_size = -1;
+    size_t payload_size = static_cast<size_t>(-1);
     EXPECT_FALSE(rtp_parser_.ParsePacket(
         packet_, kPacketLength, &parsed_header, &payload, &payload_size));
   }
diff --git a/media/cast/rtp_receiver/rtp_receiver_defines.cc b/media/cast/net/rtp/rtp_receiver_defines.cc
index e42b2b733c..9b20b5f904 100644
--- a/media/cast/rtp_receiver/rtp_receiver_defines.cc
+++ b/media/cast/net/rtp/rtp_receiver_defines.cc
@@ -2,7 +2,7 @@
 // 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_defines.h"
+#include "media/cast/net/rtp/rtp_receiver_defines.h"
 
 namespace media {
 namespace cast {
diff --git a/media/cast/rtp_receiver/rtp_receiver_defines.h b/media/cast/net/rtp/rtp_receiver_defines.h
index d907436f48..86fbd2296f 100644
--- a/media/cast/rtp_receiver/rtp_receiver_defines.h
+++ b/media/cast/net/rtp/rtp_receiver_defines.h
@@ -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.
 
@@ -7,7 +7,7 @@
 
 #include "base/basictypes.h"
 #include "media/cast/cast_config.h"
-#include "media/cast/rtcp/rtcp_defines.h"
+#include "media/cast/net/rtcp/rtcp_defines.h"
 
 namespace media {
 namespace cast {
diff --git a/media/cast/transport/rtp_sender/rtp_sender.cc b/media/cast/net/rtp/rtp_sender.cc
index b807b34757..0f88c444ec 100644
--- a/media/cast/transport/rtp_sender/rtp_sender.cc
+++ b/media/cast/net/rtp/rtp_sender.cc
@@ -1,18 +1,17 @@
-// 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/transport/rtp_sender/rtp_sender.h"
+#include "media/cast/net/rtp/rtp_sender.h"
 
 #include "base/big_endian.h"
 #include "base/logging.h"
 #include "base/rand_util.h"
-#include "media/cast/transport/cast_transport_defines.h"
-#include "media/cast/transport/pacing/paced_sender.h"
+#include "media/cast/net/cast_transport_defines.h"
+#include "media/cast/net/pacing/paced_sender.h"
 
 namespace media {
 namespace cast {
-namespace transport {
 
 namespace {
 
@@ -41,30 +40,13 @@ RtpSender::RtpSender(
 
 RtpSender::~RtpSender() {}
 
-bool RtpSender::InitializeAudio(const CastTransportAudioConfig& config) {
-  storage_.reset(new PacketStorage(config.rtp.max_outstanding_frames));
+bool RtpSender::Initialize(const CastTransportRtpConfig& config) {
+  storage_.reset(new PacketStorage(config.stored_frames));
   if (!storage_->IsValid()) {
     return false;
   }
-  config_.audio = true;
-  config_.ssrc = config.rtp.config.ssrc;
-  config_.payload_type = config.rtp.config.payload_type;
-  config_.frequency = config.frequency;
-  config_.audio_codec = config.codec;
-  packetizer_.reset(new RtpPacketizer(transport_, storage_.get(), config_));
-  return true;
-}
-
-bool RtpSender::InitializeVideo(const CastTransportVideoConfig& config) {
-  storage_.reset(new PacketStorage(config.rtp.max_outstanding_frames));
-  if (!storage_->IsValid()) {
-    return false;
-  }
-  config_.audio = false;
-  config_.ssrc = config.rtp.config.ssrc;
-  config_.payload_type = config.rtp.config.payload_type;
-  config_.frequency = kVideoFrequency;
-  config_.video_codec = config.codec;
+  config_.ssrc = config.ssrc;
+  config_.payload_type = config.rtp_payload_type;
   packetizer_.reset(new RtpPacketizer(transport_, storage_.get(), config_));
   return true;
 }
@@ -145,6 +127,5 @@ void RtpSender::UpdateSequenceNumber(Packet* packet) {
   big_endian_writer.WriteU16(packetizer_->NextSequenceNumber());
 }
 
-}  // namespace transport
 }  //  namespace cast
 }  // namespace media
diff --git a/media/cast/transport/rtp_sender/rtp_sender.h b/media/cast/net/rtp/rtp_sender.h
index e65326abf1..4dd7966a68 100644
--- a/media/cast/transport/rtp_sender/rtp_sender.h
+++ b/media/cast/net/rtp/rtp_sender.h
@@ -1,32 +1,30 @@
-// 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.
 
 // This file contains the interface to the cast RTP sender.
 
-#ifndef MEDIA_CAST_TRANSPORT_RTP_SENDER_RTP_SENDER_H_
-#define MEDIA_CAST_TRANSPORT_RTP_SENDER_RTP_SENDER_H_
+#ifndef MEDIA_CAST_NET_RTP_RTP_SENDER_H_
+#define MEDIA_CAST_NET_RTP_RTP_SENDER_H_
 
 #include <map>
 #include <set>
 
 #include "base/memory/scoped_ptr.h"
+#include "base/memory/weak_ptr.h"
 #include "base/time/tick_clock.h"
 #include "base/time/time.h"
-#include "base/memory/weak_ptr.h"
 #include "media/cast/cast_config.h"
 #include "media/cast/cast_environment.h"
-#include "media/cast/transport/cast_transport_defines.h"
-#include "media/cast/transport/cast_transport_sender.h"
-#include "media/cast/transport/pacing/paced_sender.h"
-#include "media/cast/transport/rtp_sender/packet_storage/packet_storage.h"
-#include "media/cast/transport/rtp_sender/rtp_packetizer/rtp_packetizer.h"
+#include "media/cast/net/cast_transport_defines.h"
+#include "media/cast/net/cast_transport_sender.h"
+#include "media/cast/net/pacing/paced_sender.h"
+#include "media/cast/net/rtp/packet_storage.h"
+#include "media/cast/net/rtp/rtp_packetizer.h"
 
 namespace media {
 namespace cast {
 
-namespace transport {
-
 // This object is only called from the main cast thread.
 // This class handles splitting encoded audio and video frames into packets and
 // add an RTP header to each packet. The sent packets are stored until they are
@@ -40,13 +38,9 @@ class RtpSender {
 
   ~RtpSender();
 
-  // Initialize audio stack. Audio must be initialized prior to sending encoded
-  // audio frames. Returns false if configuration is invalid.
-  bool InitializeAudio(const CastTransportAudioConfig& config);
-
-  // Initialize video stack. Video must be initialized prior to sending encoded
-  // video frames. Returns false if configuration is invalid.
-  bool InitializeVideo(const CastTransportVideoConfig& config);
+  // This must be called before sending any frames. Returns false if
+  // configuration is invalid.
+  bool Initialize(const CastTransportRtpConfig& config);
 
   void SendFrame(const EncodedFrame& frame);
 
@@ -78,8 +72,7 @@ class RtpSender {
   DISALLOW_COPY_AND_ASSIGN(RtpSender);
 };
 
-}  // namespace transport
 }  // namespace cast
 }  // namespace media
 
-#endif  // MEDIA_CAST_TRANSPORT_RTP_SENDER_RTP_SENDER_H_
+#endif  // MEDIA_CAST_NET_RTP_SENDER_RTP_SENDER_H_
diff --git a/media/cast/transport/transport/udp_transport.cc b/media/cast/net/udp_transport.cc
index 9669b17d43..ae0593d3ac 100644
--- a/media/cast/transport/transport/udp_transport.cc
+++ b/media/cast/net/udp_transport.cc
@@ -2,7 +2,7 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-#include "media/cast/transport/transport/udp_transport.h"
+#include "media/cast/net/udp_transport.h"
 
 #include <algorithm>
 #include <string>
@@ -19,7 +19,6 @@
 
 namespace media {
 namespace cast {
-namespace transport {
 
 namespace {
 const int kMaxPacketSize = 1500;
@@ -237,6 +236,5 @@ void UdpTransport::OnSent(const scoped_refptr<net::IOBuffer>& buf,
   }
 }
 
-}  // namespace transport
 }  // namespace cast
 }  // namespace media
diff --git a/media/cast/transport/transport/udp_transport.h b/media/cast/net/udp_transport.h
index 1a568501d5..951e9c12d3 100644
--- a/media/cast/transport/transport/udp_transport.h
+++ b/media/cast/net/udp_transport.h
@@ -2,15 +2,15 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-#ifndef MEDIA_CAST_TRANSPORT_TRANSPORT_UDP_TRANSPORT_H_
-#define MEDIA_CAST_TRANSPORT_TRANSPORT_UDP_TRANSPORT_H_
+#ifndef MEDIA_CAST_NET_UDP_TRANSPORT_H_
+#define MEDIA_CAST_NET_UDP_TRANSPORT_H_
 
 #include "base/memory/ref_counted.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/memory/weak_ptr.h"
 #include "media/cast/cast_environment.h"
-#include "media/cast/transport/cast_transport_config.h"
-#include "media/cast/transport/cast_transport_sender.h"
+#include "media/cast/net/cast_transport_config.h"
+#include "media/cast/net/cast_transport_sender.h"
 #include "net/base/ip_endpoint.h"
 #include "net/base/net_util.h"
 #include "net/udp/udp_socket.h"
@@ -23,7 +23,6 @@ class NetLog;
 
 namespace media {
 namespace cast {
-namespace transport {
 
 // This class implements UDP transport mechanism for Cast.
 class UdpTransport : public PacketSender {
@@ -90,8 +89,7 @@ class UdpTransport : public PacketSender {
   DISALLOW_COPY_AND_ASSIGN(UdpTransport);
 };
 
-}  // namespace transport
 }  // namespace cast
 }  // namespace media
 
-#endif  // MEDIA_CAST_TRANSPORT_TRANSPORT_UDP_TRANSPORT_H_
+#endif  // MEDIA_CAST_NET_UDP_TRANSPORT_H_
diff --git a/media/cast/transport/transport/udp_transport_unittest.cc b/media/cast/net/udp_transport_unittest.cc
index 26879492f0..7ae938b4ca 100644
--- a/media/cast/transport/transport/udp_transport_unittest.cc
+++ b/media/cast/net/udp_transport_unittest.cc
@@ -2,7 +2,7 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-#include "media/cast/transport/transport/udp_transport.h"
+#include "media/cast/net/udp_transport.h"
 
 #include <algorithm>
 #include <string>
@@ -12,14 +12,13 @@
 #include "base/callback.h"
 #include "base/message_loop/message_loop.h"
 #include "base/run_loop.h"
+#include "media/cast/net/cast_transport_config.h"
 #include "media/cast/test/utility/net_utility.h"
-#include "media/cast/transport/cast_transport_config.h"
 #include "net/base/net_util.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
 namespace media {
 namespace cast {
-namespace transport {
 
 class MockPacketReceiver {
  public:
@@ -33,7 +32,7 @@ class MockPacketReceiver {
   }
 
   std::string packet() const { return packet_; }
-  transport::PacketReceiverCallback packet_receiver() {
+  PacketReceiverCallback packet_receiver() {
     return base::Bind(&MockPacketReceiver::ReceivedPacket,
                       base::Unretained(this));
   }
@@ -50,7 +49,7 @@ void SendPacket(UdpTransport* transport, Packet packet) {
   transport->SendPacket(new base::RefCountedData<Packet>(packet), cb);
 }
 
-static void UpdateCastTransportStatus(transport::CastTransportStatus status) {
+static void UpdateCastTransportStatus(CastTransportStatus status) {
   NOTREACHED();
 }
 
@@ -95,6 +94,5 @@ TEST(UdpTransport, SendAndReceive) {
       std::equal(packet.begin(), packet.end(), receiver2.packet().begin()));
 }
 
-}  // namespace transport
 }  // namespace cast
 }  // namespace media
diff --git a/media/cast/receiver/audio_decoder.cc b/media/cast/receiver/audio_decoder.cc
index a4d1896835..dac0a5e9a1 100644
--- a/media/cast/receiver/audio_decoder.cc
+++ b/media/cast/receiver/audio_decoder.cc
@@ -23,7 +23,7 @@ class AudioDecoder::ImplBase
     : public base::RefCountedThreadSafe<AudioDecoder::ImplBase> {
  public:
   ImplBase(const scoped_refptr<CastEnvironment>& cast_environment,
-           transport::AudioCodec codec,
+           Codec codec,
            int num_channels,
            int sampling_rate)
       : cast_environment_(cast_environment),
@@ -39,7 +39,7 @@ class AudioDecoder::ImplBase
     return cast_initialization_status_;
   }
 
-  void DecodeFrame(scoped_ptr<transport::EncodedFrame> encoded_frame,
+  void DecodeFrame(scoped_ptr<EncodedFrame> encoded_frame,
                    const DecodeFrameCallback& callback) {
     DCHECK_EQ(cast_initialization_status_, STATUS_AUDIO_INITIALIZED);
 
@@ -77,7 +77,7 @@ class AudioDecoder::ImplBase
   virtual scoped_ptr<AudioBus> Decode(uint8* data, int len) = 0;
 
   const scoped_refptr<CastEnvironment> cast_environment_;
-  const transport::AudioCodec codec_;
+  const Codec codec_;
   const int num_channels_;
 
   // Subclass' ctor is expected to set this to STATUS_AUDIO_INITIALIZED.
@@ -96,7 +96,7 @@ class AudioDecoder::OpusImpl : public AudioDecoder::ImplBase {
            int num_channels,
            int sampling_rate)
       : ImplBase(cast_environment,
-                 transport::kOpus,
+                 CODEC_AUDIO_OPUS,
                  num_channels,
                  sampling_rate),
         decoder_memory_(new uint8[opus_decoder_get_size(num_channels)]),
@@ -166,7 +166,7 @@ class AudioDecoder::Pcm16Impl : public AudioDecoder::ImplBase {
             int num_channels,
             int sampling_rate)
       : ImplBase(cast_environment,
-                 transport::kPcm16,
+                 CODEC_AUDIO_PCM16,
                  num_channels,
                  sampling_rate) {
     if (ImplBase::cast_initialization_status_ != STATUS_AUDIO_UNINITIALIZED)
@@ -202,13 +202,13 @@ AudioDecoder::AudioDecoder(
     const scoped_refptr<CastEnvironment>& cast_environment,
     int channels,
     int sampling_rate,
-    transport::AudioCodec codec)
+    Codec codec)
     : cast_environment_(cast_environment) {
   switch (codec) {
-    case transport::kOpus:
+    case CODEC_AUDIO_OPUS:
       impl_ = new OpusImpl(cast_environment, channels, sampling_rate);
       break;
-    case transport::kPcm16:
+    case CODEC_AUDIO_PCM16:
       impl_ = new Pcm16Impl(cast_environment, channels, sampling_rate);
       break;
     default:
@@ -226,7 +226,7 @@ CastInitializationStatus AudioDecoder::InitializationResult() const {
 }
 
 void AudioDecoder::DecodeFrame(
-    scoped_ptr<transport::EncodedFrame> encoded_frame,
+    scoped_ptr<EncodedFrame> encoded_frame,
     const DecodeFrameCallback& callback) {
   DCHECK(encoded_frame.get());
   DCHECK(!callback.is_null());
diff --git a/media/cast/receiver/audio_decoder.h b/media/cast/receiver/audio_decoder.h
index c66735e4e6..0b13eae6a8 100644
--- a/media/cast/receiver/audio_decoder.h
+++ b/media/cast/receiver/audio_decoder.h
@@ -10,7 +10,7 @@
 #include "media/base/audio_bus.h"
 #include "media/cast/cast_config.h"
 #include "media/cast/cast_environment.h"
-#include "media/cast/transport/cast_transport_config.h"
+#include "media/cast/net/cast_transport_config.h"
 
 namespace media {
 namespace cast {
@@ -29,7 +29,7 @@ class AudioDecoder {
   AudioDecoder(const scoped_refptr<CastEnvironment>& cast_environment,
                int channels,
                int sampling_rate,
-               transport::AudioCodec codec);
+               Codec codec);
   virtual ~AudioDecoder();
 
   // Returns STATUS_AUDIO_INITIALIZED if the decoder was successfully
@@ -44,7 +44,7 @@ class AudioDecoder {
   // monotonically-increasing by 1 for each successive call to this method.
   // When it is not, the decoder will assume one or more frames have been
   // dropped (e.g., due to packet loss), and will perform recovery actions.
-  void DecodeFrame(scoped_ptr<transport::EncodedFrame> encoded_frame,
+  void DecodeFrame(scoped_ptr<EncodedFrame> encoded_frame,
                    const DecodeFrameCallback& callback);
 
  private:
diff --git a/media/cast/receiver/audio_decoder_unittest.cc b/media/cast/receiver/audio_decoder_unittest.cc
index 6985a69423..576e9dbce5 100644
--- a/media/cast/receiver/audio_decoder_unittest.cc
+++ b/media/cast/receiver/audio_decoder_unittest.cc
@@ -20,11 +20,11 @@ namespace cast {
 
 namespace {
 struct TestScenario {
-  transport::AudioCodec codec;
+  Codec codec;
   int num_channels;
   int sampling_rate;
 
-  TestScenario(transport::AudioCodec c, int n, int s)
+  TestScenario(Codec c, int n, int s)
       : codec(c), num_channels(n), sampling_rate(s) {}
 };
 }  // namespace
@@ -51,7 +51,7 @@ class AudioDecoderTest : public ::testing::TestWithParam<TestScenario> {
     last_frame_id_ = 0;
     seen_a_decoded_frame_ = false;
 
-    if (GetParam().codec == transport::kOpus) {
+    if (GetParam().codec == CODEC_AUDIO_OPUS) {
       opus_encoder_memory_.reset(
           new uint8[opus_encoder_get_size(GetParam().num_channels)]);
       OpusEncoder* const opus_encoder =
@@ -73,9 +73,9 @@ class AudioDecoderTest : public ::testing::TestWithParam<TestScenario> {
   void FeedMoreAudio(const base::TimeDelta& duration,
                      int num_dropped_frames) {
     // Prepare a simulated EncodedFrame to feed into the AudioDecoder.
-    scoped_ptr<transport::EncodedFrame> encoded_frame(
-        new transport::EncodedFrame());
-    encoded_frame->dependency = transport::EncodedFrame::KEY;
+    scoped_ptr<EncodedFrame> encoded_frame(
+        new EncodedFrame());
+    encoded_frame->dependency = EncodedFrame::KEY;
     encoded_frame->frame_id = last_frame_id_ + 1 + num_dropped_frames;
     encoded_frame->referenced_frame_id = encoded_frame->frame_id;
     last_frame_id_ = encoded_frame->frame_id;
@@ -88,13 +88,13 @@ class AudioDecoderTest : public ::testing::TestWithParam<TestScenario> {
     std::vector<int16> interleaved(num_elements);
     audio_bus->ToInterleaved(
         audio_bus->frames(), sizeof(int16), &interleaved.front());
-    if (GetParam().codec == transport::kPcm16) {
+    if (GetParam().codec == CODEC_AUDIO_PCM16) {
       encoded_frame->data.resize(num_elements * sizeof(int16));
       int16* const pcm_data =
           reinterpret_cast<int16*>(encoded_frame->mutable_bytes());
       for (size_t i = 0; i < interleaved.size(); ++i)
         pcm_data[i] = static_cast<int16>(base::HostToNet16(interleaved[i]));
-    } else if (GetParam().codec == transport::kOpus) {
+    } else if (GetParam().codec == CODEC_AUDIO_OPUS) {
       OpusEncoder* const opus_encoder =
           reinterpret_cast<OpusEncoder*>(opus_encoder_memory_.get());
       const int kOpusEncodeBufferSize = 4000;
@@ -154,7 +154,7 @@ class AudioDecoderTest : public ::testing::TestWithParam<TestScenario> {
     // first frame seen at the start (and immediately after dropped packet
     // recovery) because it introduces a tiny, significant delay.
     bool examine_signal = true;
-    if (GetParam().codec == transport::kOpus) {
+    if (GetParam().codec == CODEC_AUDIO_OPUS) {
       examine_signal = seen_a_decoded_frame_ && should_be_continuous;
       seen_a_decoded_frame_ = true;
     }
@@ -229,13 +229,14 @@ TEST_P(AudioDecoderTest, RecoversFromDroppedFrames) {
   WaitForAllAudioToBeDecoded();
 }
 
-INSTANTIATE_TEST_CASE_P(AudioDecoderTestScenarios,
-                        AudioDecoderTest,
-                        ::testing::Values(
-                             TestScenario(transport::kPcm16, 1, 8000),
-                             TestScenario(transport::kPcm16, 2, 48000),
-                             TestScenario(transport::kOpus, 1, 8000),
-                             TestScenario(transport::kOpus, 2, 48000)));
+INSTANTIATE_TEST_CASE_P(
+    AudioDecoderTestScenarios,
+    AudioDecoderTest,
+    ::testing::Values(
+         TestScenario(CODEC_AUDIO_PCM16, 1, 8000),
+         TestScenario(CODEC_AUDIO_PCM16, 2, 48000),
+         TestScenario(CODEC_AUDIO_OPUS, 1, 8000),
+         TestScenario(CODEC_AUDIO_OPUS, 2, 48000)));
 
 }  // namespace cast
 }  // namespace media
diff --git a/media/cast/receiver/cast_receiver_impl.cc b/media/cast/receiver/cast_receiver_impl.cc
index 7cff354c14..36669b9e62 100644
--- a/media/cast/receiver/cast_receiver_impl.cc
+++ b/media/cast/receiver/cast_receiver_impl.cc
@@ -20,7 +20,7 @@ scoped_ptr<CastReceiver> CastReceiver::Create(
     scoped_refptr<CastEnvironment> cast_environment,
     const FrameReceiverConfig& audio_config,
     const FrameReceiverConfig& video_config,
-    transport::PacketSender* const packet_sender) {
+    PacketSender* const packet_sender) {
   return scoped_ptr<CastReceiver>(new CastReceiverImpl(
       cast_environment, audio_config, video_config, packet_sender));
 }
@@ -29,7 +29,7 @@ CastReceiverImpl::CastReceiverImpl(
     scoped_refptr<CastEnvironment> cast_environment,
     const FrameReceiverConfig& audio_config,
     const FrameReceiverConfig& video_config,
-    transport::PacketSender* const packet_sender)
+    PacketSender* const packet_sender)
     : cast_environment_(cast_environment),
       pacer_(cast_environment->Clock(),
              cast_environment->Logging(),
@@ -41,8 +41,8 @@ CastReceiverImpl::CastReceiverImpl(
       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) {}
+      audio_codec_(audio_config.codec),
+      video_codec_(video_config.codec) {}
 
 CastReceiverImpl::~CastReceiverImpl() {}
 
@@ -76,7 +76,7 @@ void CastReceiverImpl::DispatchReceivedPacket(scoped_ptr<Packet> packet) {
                  base::Passed(&packet)));
 }
 
-transport::PacketReceiverCallback CastReceiverImpl::packet_receiver() {
+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
@@ -122,7 +122,7 @@ void CastReceiverImpl::RequestEncodedVideoFrame(
 
 void CastReceiverImpl::DecodeEncodedAudioFrame(
     const AudioFrameDecodedCallback& callback,
-    scoped_ptr<transport::EncodedFrame> encoded_frame) {
+    scoped_ptr<EncodedFrame> encoded_frame) {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
   if (!encoded_frame) {
     callback.Run(make_scoped_ptr<AudioBus>(NULL), base::TimeTicks(), false);
@@ -150,7 +150,7 @@ void CastReceiverImpl::DecodeEncodedAudioFrame(
 
 void CastReceiverImpl::DecodeEncodedVideoFrame(
     const VideoFrameDecodedCallback& callback,
-    scoped_ptr<transport::EncodedFrame> encoded_frame) {
+    scoped_ptr<EncodedFrame> encoded_frame) {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
   if (!encoded_frame) {
     callback.Run(
diff --git a/media/cast/receiver/cast_receiver_impl.h b/media/cast/receiver/cast_receiver_impl.h
index c0dd5f38d1..3443561257 100644
--- a/media/cast/receiver/cast_receiver_impl.h
+++ b/media/cast/receiver/cast_receiver_impl.h
@@ -10,8 +10,8 @@
 #include "media/cast/cast_config.h"
 #include "media/cast/cast_environment.h"
 #include "media/cast/cast_receiver.h"
+#include "media/cast/net/pacing/paced_sender.h"
 #include "media/cast/receiver/frame_receiver.h"
-#include "media/cast/transport/pacing/paced_sender.h"
 
 namespace media {
 namespace cast {
@@ -27,12 +27,12 @@ class CastReceiverImpl : public CastReceiver {
   CastReceiverImpl(scoped_refptr<CastEnvironment> cast_environment,
                    const FrameReceiverConfig& audio_config,
                    const FrameReceiverConfig& video_config,
-                   transport::PacketSender* const packet_sender);
+                   PacketSender* const packet_sender);
 
   virtual ~CastReceiverImpl();
 
   // CastReceiver implementation.
-  virtual transport::PacketReceiverCallback packet_receiver() OVERRIDE;
+  virtual PacketReceiverCallback packet_receiver() OVERRIDE;
   virtual void RequestDecodedAudioFrame(
       const AudioFrameDecodedCallback& callback) OVERRIDE;
   virtual void RequestEncodedAudioFrame(
@@ -51,13 +51,13 @@ class CastReceiverImpl : public CastReceiver {
   // uses this as a callback for RequestEncodedAudioFrame().
   void DecodeEncodedAudioFrame(
       const AudioFrameDecodedCallback& callback,
-      scoped_ptr<transport::EncodedFrame> encoded_frame);
+      scoped_ptr<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);
+      scoped_ptr<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
@@ -88,7 +88,7 @@ class CastReceiverImpl : public CastReceiver {
       bool is_continuous);
 
   const scoped_refptr<CastEnvironment> cast_environment_;
-  transport::PacedSender pacer_;
+  PacedSender pacer_;
   FrameReceiver audio_receiver_;
   FrameReceiver video_receiver_;
 
@@ -102,8 +102,8 @@ class CastReceiverImpl : public CastReceiver {
   // 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_;
+  const Codec audio_codec_;
+  const Codec video_codec_;
 
   // Created on-demand to decode frames from |audio_receiver_| into AudioBuses
   // for playback.
diff --git a/media/cast/receiver/frame_receiver.cc b/media/cast/receiver/frame_receiver.cc
index e189cc99a7..951e958b80 100644
--- a/media/cast/receiver/frame_receiver.cc
+++ b/media/cast/receiver/frame_receiver.cc
@@ -23,7 +23,7 @@ FrameReceiver::FrameReceiver(
     const scoped_refptr<CastEnvironment>& cast_environment,
     const FrameReceiverConfig& config,
     EventMediaType event_media_type,
-    transport::PacedPacketSender* const packet_sender)
+    PacedPacketSender* const packet_sender)
     : cast_environment_(cast_environment),
       packet_parser_(config.incoming_ssrc, config.rtp_payload_type),
       stats_(cast_environment->Clock()),
@@ -193,8 +193,8 @@ void FrameReceiver::EmitAvailableEncodedFrames() {
     // 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());
+    scoped_ptr<EncodedFrame> encoded_frame(
+        new EncodedFrame());
     bool is_consecutively_next_frame = false;
     bool have_multiple_complete_frames = false;
     if (!framer_.GetEncodedFrame(encoded_frame.get(),
@@ -239,7 +239,7 @@ void FrameReceiver::EmitAvailableEncodedFrames() {
     }
 
     // Decrypt the payload data in the frame, if crypto is being used.
-    if (decryptor_.initialized()) {
+    if (decryptor_.is_activated()) {
       std::string decrypted_data;
       if (!decryptor_.Decrypt(encoded_frame->frame_id,
                               encoded_frame->data,
diff --git a/media/cast/receiver/frame_receiver.h b/media/cast/receiver/frame_receiver.h
index ac14ab1e0f..f8e4481db2 100644
--- a/media/cast/receiver/frame_receiver.h
+++ b/media/cast/receiver/frame_receiver.h
@@ -9,17 +9,17 @@
 #include "base/memory/scoped_ptr.h"
 #include "base/memory/weak_ptr.h"
 #include "base/time/time.h"
-#include "media/cast/base/clock_drift_smoother.h"
 #include "media/cast/cast_config.h"
 #include "media/cast/cast_receiver.h"
-#include "media/cast/framer/framer.h"
+#include "media/cast/common/clock_drift_smoother.h"
+#include "media/cast/common/transport_encryption_handler.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/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"
+#include "media/cast/net/rtcp/receiver_rtcp_event_subscriber.h"
+#include "media/cast/net/rtcp/rtcp.h"
+#include "media/cast/net/rtp/framer.h"
+#include "media/cast/net/rtp/receiver_stats.h"
+#include "media/cast/net/rtp/rtp_parser.h"
+#include "media/cast/net/rtp/rtp_receiver_defines.h"
 
 namespace media {
 namespace cast {
@@ -50,7 +50,7 @@ class FrameReceiver : public RtpPayloadFeedback,
   FrameReceiver(const scoped_refptr<CastEnvironment>& cast_environment,
                 const FrameReceiverConfig& config,
                 EventMediaType event_media_type,
-                transport::PacedPacketSender* const packet_sender);
+                PacedPacketSender* const packet_sender);
 
   virtual ~FrameReceiver();
 
@@ -152,7 +152,7 @@ class FrameReceiver : public RtpPayloadFeedback,
   Rtcp rtcp_;
 
   // Decrypts encrypted frames.
-  transport::TransportEncryptionHandler decryptor_;
+  TransportEncryptionHandler decryptor_;
 
   // Outstanding callbacks to run to deliver on client requests for frames.
   std::list<ReceiveEncodedFrameCallback> frame_request_queue_;
diff --git a/media/cast/receiver/frame_receiver_unittest.cc b/media/cast/receiver/frame_receiver_unittest.cc
index 4d8273e132..121d12e812 100644
--- a/media/cast/receiver/frame_receiver_unittest.cc
+++ b/media/cast/receiver/frame_receiver_unittest.cc
@@ -12,11 +12,11 @@
 #include "media/cast/cast_defines.h"
 #include "media/cast/cast_environment.h"
 #include "media/cast/logging/simple_event_subscriber.h"
+#include "media/cast/net/pacing/mock_paced_packet_sender.h"
+#include "media/cast/net/rtcp/test_rtcp_packet_builder.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::_;
@@ -41,7 +41,7 @@ class FakeFrameClient {
         std::make_pair(expected_frame_id, expected_playout_time));
   }
 
-  void DeliverEncodedFrame(scoped_ptr<transport::EncodedFrame> frame) {
+  void DeliverEncodedFrame(scoped_ptr<EncodedFrame> frame) {
     SCOPED_TRACE(::testing::Message() << "num_called_ is " << num_called_);
     ASSERT_FALSE(!frame)
         << "If at shutdown: There were unsatisfied requests enqueued.";
@@ -136,7 +136,7 @@ class FrameReceiverTest : public ::testing::Test {
   RtpCastHeader rtp_header_;
   base::SimpleTestTickClock* testing_clock_;  // Owned by CastEnvironment.
   base::TimeTicks start_time_;
-  transport::MockPacedPacketSender mock_transport_;
+  MockPacedPacketSender mock_transport_;
   scoped_refptr<test::FakeSingleThreadTaskRunner> task_runner_;
   scoped_refptr<CastEnvironment> cast_environment_;
   FakeFrameClient frame_client_;
diff --git a/media/cast/receiver/video_decoder.cc b/media/cast/receiver/video_decoder.cc
index 6db3fd35f3..f4de9c3b31 100644
--- a/media/cast/receiver/video_decoder.cc
+++ b/media/cast/receiver/video_decoder.cc
@@ -30,7 +30,7 @@ class VideoDecoder::ImplBase
     : public base::RefCountedThreadSafe<VideoDecoder::ImplBase> {
  public:
   ImplBase(const scoped_refptr<CastEnvironment>& cast_environment,
-           transport::VideoCodec codec)
+           Codec codec)
       : cast_environment_(cast_environment),
         codec_(codec),
         cast_initialization_status_(STATUS_VIDEO_UNINITIALIZED),
@@ -40,7 +40,7 @@ class VideoDecoder::ImplBase
     return cast_initialization_status_;
   }
 
-  void DecodeFrame(scoped_ptr<transport::EncodedFrame> encoded_frame,
+  void DecodeFrame(scoped_ptr<EncodedFrame> encoded_frame,
                    const DecodeFrameCallback& callback) {
     DCHECK_EQ(cast_initialization_status_, STATUS_VIDEO_INITIALIZED);
 
@@ -77,7 +77,7 @@ class VideoDecoder::ImplBase
   virtual scoped_refptr<VideoFrame> Decode(uint8* data, int len) = 0;
 
   const scoped_refptr<CastEnvironment> cast_environment_;
-  const transport::VideoCodec codec_;
+  const Codec codec_;
 
   // Subclass' ctor is expected to set this to STATUS_VIDEO_INITIALIZED.
   CastInitializationStatus cast_initialization_status_;
@@ -92,7 +92,7 @@ class VideoDecoder::ImplBase
 class VideoDecoder::Vp8Impl : public VideoDecoder::ImplBase {
  public:
   explicit Vp8Impl(const scoped_refptr<CastEnvironment>& cast_environment)
-      : ImplBase(cast_environment, transport::kVp8) {
+      : ImplBase(cast_environment, CODEC_VIDEO_VP8) {
     if (ImplBase::cast_initialization_status_ != STATUS_VIDEO_UNINITIALIZED)
       return;
 
@@ -173,7 +173,7 @@ class VideoDecoder::Vp8Impl : public VideoDecoder::ImplBase {
 class VideoDecoder::FakeImpl : public VideoDecoder::ImplBase {
  public:
   explicit FakeImpl(const scoped_refptr<CastEnvironment>& cast_environment)
-      : ImplBase(cast_environment, transport::kFakeSoftwareVideo),
+      : ImplBase(cast_environment, CODEC_VIDEO_FAKE),
         last_decoded_id_(-1) {
     if (ImplBase::cast_initialization_status_ != STATUS_VIDEO_UNINITIALIZED)
       return;
@@ -184,9 +184,14 @@ class VideoDecoder::FakeImpl : public VideoDecoder::ImplBase {
   virtual ~FakeImpl() {}
 
   virtual scoped_refptr<VideoFrame> Decode(uint8* data, int len) OVERRIDE {
+    // Make sure this is a JSON string.
+    if (!len || data[0] != '{')
+      return NULL;
     base::JSONReader reader;
     scoped_ptr<base::Value> values(
-        reader.Read(base::StringPiece(reinterpret_cast<char*>(data))));
+        reader.Read(base::StringPiece(reinterpret_cast<char*>(data), len)));
+    if (!values)
+      return NULL;
     base::DictionaryValue* dict = NULL;
     values->GetAsDictionary(&dict);
 
@@ -209,18 +214,18 @@ class VideoDecoder::FakeImpl : public VideoDecoder::ImplBase {
 
 VideoDecoder::VideoDecoder(
     const scoped_refptr<CastEnvironment>& cast_environment,
-    transport::VideoCodec codec)
+    Codec codec)
     : cast_environment_(cast_environment) {
   switch (codec) {
 #ifndef OFFICIAL_BUILD
-    case transport::kFakeSoftwareVideo:
+    case CODEC_VIDEO_FAKE:
       impl_ = new FakeImpl(cast_environment);
       break;
 #endif
-    case transport::kVp8:
+    case CODEC_VIDEO_VP8:
       impl_ = new Vp8Impl(cast_environment);
       break;
-    case transport::kH264:
+    case CODEC_VIDEO_H264:
       // TODO(miu): Need implementation.
       NOTIMPLEMENTED();
       break;
@@ -239,7 +244,7 @@ CastInitializationStatus VideoDecoder::InitializationResult() const {
 }
 
 void VideoDecoder::DecodeFrame(
-    scoped_ptr<transport::EncodedFrame> encoded_frame,
+    scoped_ptr<EncodedFrame> encoded_frame,
     const DecodeFrameCallback& callback) {
   DCHECK(encoded_frame.get());
   DCHECK(!callback.is_null());
diff --git a/media/cast/receiver/video_decoder.h b/media/cast/receiver/video_decoder.h
index 66dc36bb2a..f3d8ca202e 100644
--- a/media/cast/receiver/video_decoder.h
+++ b/media/cast/receiver/video_decoder.h
@@ -10,7 +10,7 @@
 #include "base/memory/scoped_ptr.h"
 #include "media/base/video_frame.h"
 #include "media/cast/cast_config.h"
-#include "media/cast/transport/cast_transport_config.h"
+#include "media/cast/net/cast_transport_config.h"
 
 namespace media {
 namespace cast {
@@ -28,7 +28,7 @@ class VideoDecoder {
                               bool is_continuous)> DecodeFrameCallback;
 
   VideoDecoder(const scoped_refptr<CastEnvironment>& cast_environment,
-               transport::VideoCodec codec);
+               Codec codec);
   virtual ~VideoDecoder();
 
   // Returns STATUS_VIDEO_INITIALIZED if the decoder was successfully
@@ -43,7 +43,7 @@ class VideoDecoder {
   // monotonically-increasing by 1 for each successive call to this method.
   // When it is not, the decoder will assume one or more frames have been
   // dropped (e.g., due to packet loss), and will perform recovery actions.
-  void DecodeFrame(scoped_ptr<transport::EncodedFrame> encoded_frame,
+  void DecodeFrame(scoped_ptr<EncodedFrame> encoded_frame,
                    const DecodeFrameCallback& callback);
 
  private:
diff --git a/media/cast/receiver/video_decoder_unittest.cc b/media/cast/receiver/video_decoder_unittest.cc
index 1d16534b96..c7c39ad2d0 100644
--- a/media/cast/receiver/video_decoder_unittest.cc
+++ b/media/cast/receiver/video_decoder_unittest.cc
@@ -11,9 +11,9 @@
 #include "base/time/time.h"
 #include "media/cast/cast_config.h"
 #include "media/cast/receiver/video_decoder.h"
+#include "media/cast/sender/vp8_encoder.h"
 #include "media/cast/test/utility/standalone_cast_environment.h"
 #include "media/cast/test/utility/video_utility.h"
-#include "media/cast/video_sender/codecs/vp8/vp8_encoder.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
 namespace media {
@@ -35,8 +35,7 @@ VideoSenderConfig GetVideoSenderConfigForTest() {
 
 }  // namespace
 
-class VideoDecoderTest
-    : public ::testing::TestWithParam<transport::VideoCodec> {
+class VideoDecoderTest : public ::testing::TestWithParam<Codec> {
  public:
   VideoDecoderTest()
       : cast_environment_(new StandaloneCastEnvironment()),
@@ -74,9 +73,10 @@ class VideoDecoderTest
     PopulateVideoFrame(video_frame, 0);
 
     // Encode |frame| into |encoded_frame->data|.
-    scoped_ptr<transport::EncodedFrame> encoded_frame(
-        new transport::EncodedFrame());
-    CHECK_EQ(transport::kVp8, GetParam());  // Only support VP8 test currently.
+    scoped_ptr<EncodedFrame> encoded_frame(
+        new EncodedFrame());
+    // Test only supports VP8, currently.
+    CHECK_EQ(CODEC_VIDEO_VP8, GetParam());
     vp8_encoder_.Encode(video_frame, encoded_frame.get());
     encoded_frame->frame_id = last_frame_id_ + 1 + num_dropped_frames;
     last_frame_id_ = encoded_frame->frame_id;
@@ -177,7 +177,7 @@ TEST_P(VideoDecoderTest, RecoversFromDroppedFrames) {
 
 INSTANTIATE_TEST_CASE_P(VideoDecoderTestScenarios,
                         VideoDecoderTest,
-                        ::testing::Values(transport::kVp8));
+                        ::testing::Values(CODEC_VIDEO_VP8));
 
 }  // namespace cast
 }  // namespace media
diff --git a/media/cast/audio_sender/audio_encoder.cc b/media/cast/sender/audio_encoder.cc
index 8860c7dd2d..83bd594fb6 100644
--- a/media/cast/audio_sender/audio_encoder.cc
+++ b/media/cast/sender/audio_encoder.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_sender/audio_encoder.h"
+#include "media/cast/sender/audio_encoder.h"
 
 #include <algorithm>
 
@@ -44,7 +44,7 @@ class AudioEncoder::ImplBase
     : public base::RefCountedThreadSafe<AudioEncoder::ImplBase> {
  public:
   ImplBase(const scoped_refptr<CastEnvironment>& cast_environment,
-           transport::AudioCodec codec,
+           Codec codec,
            int num_channels,
            int sampling_rate,
            const FrameEncodedCallback& callback)
@@ -117,9 +117,9 @@ class AudioEncoder::ImplBase
       if (buffer_fill_end_ < samples_per_frame_)
         break;
 
-      scoped_ptr<transport::EncodedFrame> audio_frame(
-          new transport::EncodedFrame());
-      audio_frame->dependency = transport::EncodedFrame::KEY;
+      scoped_ptr<EncodedFrame> audio_frame(
+          new EncodedFrame());
+      audio_frame->dependency = EncodedFrame::KEY;
       audio_frame->frame_id = frame_id_;
       audio_frame->referenced_frame_id = frame_id_;
       audio_frame->rtp_timestamp = frame_rtp_timestamp_;
@@ -151,7 +151,7 @@ class AudioEncoder::ImplBase
   virtual bool EncodeFromFilledBuffer(std::string* out) = 0;
 
   const scoped_refptr<CastEnvironment> cast_environment_;
-  const transport::AudioCodec codec_;
+  const Codec codec_;
   const int num_channels_;
   const int samples_per_frame_;
   const FrameEncodedCallback callback_;
@@ -193,7 +193,7 @@ class AudioEncoder::OpusImpl : public AudioEncoder::ImplBase {
            int bitrate,
            const FrameEncodedCallback& callback)
       : ImplBase(cast_environment,
-                 transport::kOpus,
+                 CODEC_AUDIO_OPUS,
                  num_channels,
                  sampling_rate,
                  callback),
@@ -283,7 +283,7 @@ class AudioEncoder::Pcm16Impl : public AudioEncoder::ImplBase {
             int sampling_rate,
             const FrameEncodedCallback& callback)
       : ImplBase(cast_environment,
-                 transport::kPcm16,
+                 CODEC_AUDIO_PCM16,
                  num_channels,
                  sampling_rate,
                  callback),
@@ -326,24 +326,27 @@ class AudioEncoder::Pcm16Impl : public AudioEncoder::ImplBase {
 
 AudioEncoder::AudioEncoder(
     const scoped_refptr<CastEnvironment>& cast_environment,
-    const AudioSenderConfig& audio_config,
+    int num_channels,
+    int sampling_rate,
+    int bitrate,
+    Codec codec,
     const FrameEncodedCallback& frame_encoded_callback)
     : cast_environment_(cast_environment) {
   // Note: It doesn't matter which thread constructs AudioEncoder, just so long
   // as all calls to InsertAudio() are by the same thread.
   insert_thread_checker_.DetachFromThread();
-  switch (audio_config.codec) {
-    case transport::kOpus:
+  switch (codec) {
+    case CODEC_AUDIO_OPUS:
       impl_ = new OpusImpl(cast_environment,
-                           audio_config.channels,
-                           audio_config.frequency,
-                           audio_config.bitrate,
+                           num_channels,
+                           sampling_rate,
+                           bitrate,
                            frame_encoded_callback);
       break;
-    case transport::kPcm16:
+    case CODEC_AUDIO_PCM16:
       impl_ = new Pcm16Impl(cast_environment,
-                            audio_config.channels,
-                            audio_config.frequency,
+                            num_channels,
+                            sampling_rate,
                             frame_encoded_callback);
       break;
     default:
diff --git a/media/cast/audio_sender/audio_encoder.h b/media/cast/sender/audio_encoder.h
index 2297672b74..5f080c6cb6 100644
--- a/media/cast/audio_sender/audio_encoder.h
+++ b/media/cast/sender/audio_encoder.h
@@ -1,15 +1,14 @@
-// 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_SENDER_AUDIO_ENCODER_H_
-#define MEDIA_CAST_AUDIO_SENDER_AUDIO_ENCODER_H_
+#ifndef MEDIA_CAST_SENDER_AUDIO_ENCODER_H_
+#define MEDIA_CAST_SENDER_AUDIO_ENCODER_H_
 
 #include "base/memory/ref_counted.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/threading/thread_checker.h"
 #include "media/base/audio_bus.h"
-#include "media/cast/cast_config.h"
 #include "media/cast/cast_environment.h"
 
 namespace base {
@@ -21,11 +20,14 @@ namespace cast {
 
 class AudioEncoder {
  public:
-  typedef base::Callback<void(scoped_ptr<transport::EncodedFrame>)>
+  typedef base::Callback<void(scoped_ptr<EncodedFrame>)>
       FrameEncodedCallback;
 
   AudioEncoder(const scoped_refptr<CastEnvironment>& cast_environment,
-               const AudioSenderConfig& audio_config,
+               int num_channels,
+               int sampling_rate,
+               int bitrate,
+               Codec codec,
                const FrameEncodedCallback& frame_encoded_callback);
   virtual ~AudioEncoder();
 
@@ -51,4 +53,4 @@ class AudioEncoder {
 }  // namespace cast
 }  // namespace media
 
-#endif  // MEDIA_CAST_AUDIO_SENDER_AUDIO_ENCODER_H_
+#endif  // MEDIA_CAST_SENDER_AUDIO_ENCODER_H_
diff --git a/media/cast/audio_sender/audio_encoder_unittest.cc b/media/cast/sender/audio_encoder_unittest.cc
index b521099243..0764148a4c 100644
--- a/media/cast/audio_sender/audio_encoder_unittest.cc
+++ b/media/cast/sender/audio_encoder_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.
 
@@ -12,9 +12,8 @@
 #include "base/memory/scoped_ptr.h"
 #include "media/base/audio_bus.h"
 #include "media/base/media.h"
-#include "media/cast/audio_sender/audio_encoder.h"
-#include "media/cast/cast_config.h"
 #include "media/cast/cast_environment.h"
+#include "media/cast/sender/audio_encoder.h"
 #include "media/cast/test/fake_single_thread_task_runner.h"
 #include "media/cast/test/utility/audio_utility.h"
 #include "testing/gtest/include/gtest/gtest.h"
@@ -22,13 +21,13 @@
 namespace media {
 namespace cast {
 
-static const int64 kStartMillisecond = INT64_C(12345678900000);
+static const int kNumChannels = 2;
 
 namespace {
 
 class TestEncodedAudioFrameReceiver {
  public:
-  explicit TestEncodedAudioFrameReceiver(transport::AudioCodec codec)
+  explicit TestEncodedAudioFrameReceiver(Codec codec)
       : codec_(codec), frames_received_(0), rtp_lower_bound_(0) {}
   virtual ~TestEncodedAudioFrameReceiver() {}
 
@@ -40,8 +39,8 @@ class TestEncodedAudioFrameReceiver {
     upper_bound_ = upper_bound;
   }
 
-  void FrameEncoded(scoped_ptr<transport::EncodedFrame> encoded_frame) {
-    EXPECT_EQ(encoded_frame->dependency, transport::EncodedFrame::KEY);
+  void FrameEncoded(scoped_ptr<EncodedFrame> encoded_frame) {
+    EXPECT_EQ(encoded_frame->dependency, EncodedFrame::KEY);
     EXPECT_EQ(static_cast<uint8>(frames_received_ & 0xff),
               encoded_frame->frame_id);
     EXPECT_EQ(encoded_frame->frame_id, encoded_frame->referenced_frame_id);
@@ -62,7 +61,7 @@ class TestEncodedAudioFrameReceiver {
   }
 
  private:
-  const transport::AudioCodec codec_;
+  const Codec codec_;
   int frames_received_;
   uint32 rtp_lower_bound_;
   base::TimeTicks lower_bound_;
@@ -96,8 +95,7 @@ class AudioEncoderTest : public ::testing::TestWithParam<TestScenario> {
   AudioEncoderTest() {
     InitializeMediaLibraryForTesting();
     testing_clock_ = new base::SimpleTestTickClock();
-    testing_clock_->Advance(
-        base::TimeDelta::FromMilliseconds(kStartMillisecond));
+    testing_clock_->Advance(base::TimeTicks::Now() - base::TimeTicks());
   }
 
   virtual void SetUp() {
@@ -111,7 +109,7 @@ class AudioEncoderTest : public ::testing::TestWithParam<TestScenario> {
 
   virtual ~AudioEncoderTest() {}
 
-  void RunTestForCodec(transport::AudioCodec codec) {
+  void RunTestForCodec(Codec codec) {
     const TestScenario& scenario = GetParam();
     SCOPED_TRACE(::testing::Message() << "Durations: " << scenario.ToString());
 
@@ -144,18 +142,10 @@ class AudioEncoderTest : public ::testing::TestWithParam<TestScenario> {
   }
 
  private:
-  void CreateObjectsForCodec(transport::AudioCodec codec) {
-    AudioSenderConfig audio_config;
-    audio_config.codec = codec;
-    audio_config.use_external_encoder = false;
-    audio_config.frequency = kDefaultAudioSamplingRate;
-    audio_config.channels = 2;
-    audio_config.bitrate = kDefaultAudioEncoderBitrate;
-    audio_config.rtp_config.payload_type = 127;
-
+  void CreateObjectsForCodec(Codec codec) {
     audio_bus_factory_.reset(
-        new TestAudioBusFactory(audio_config.channels,
-                                audio_config.frequency,
+        new TestAudioBusFactory(kNumChannels,
+                                kDefaultAudioSamplingRate,
                                 TestAudioBusFactory::kMiddleANoteFreq,
                                 0.5f));
 
@@ -163,7 +153,10 @@ class AudioEncoderTest : public ::testing::TestWithParam<TestScenario> {
 
     audio_encoder_.reset(new AudioEncoder(
         cast_environment_,
-        audio_config,
+        kNumChannels,
+        kDefaultAudioSamplingRate,
+        kDefaultAudioEncoderBitrate,
+        codec,
         base::Bind(&TestEncodedAudioFrameReceiver::FrameEncoded,
                    base::Unretained(receiver_.get()))));
   }
@@ -178,9 +171,13 @@ class AudioEncoderTest : public ::testing::TestWithParam<TestScenario> {
   DISALLOW_COPY_AND_ASSIGN(AudioEncoderTest);
 };
 
-TEST_P(AudioEncoderTest, EncodeOpus) { RunTestForCodec(transport::kOpus); }
+TEST_P(AudioEncoderTest, EncodeOpus) {
+  RunTestForCodec(CODEC_AUDIO_OPUS);
+}
 
-TEST_P(AudioEncoderTest, EncodePcm16) { RunTestForCodec(transport::kPcm16); }
+TEST_P(AudioEncoderTest, EncodePcm16) {
+  RunTestForCodec(CODEC_AUDIO_PCM16);
+}
 
 static const int64 kOneCall_3Millis[] = {3};
 static const int64 kOneCall_10Millis[] = {10};
diff --git a/media/cast/audio_sender/audio_sender.cc b/media/cast/sender/audio_sender.cc
index 878f3456c8..7c7c696125 100644
--- a/media/cast/audio_sender/audio_sender.cc
+++ b/media/cast/sender/audio_sender.cc
@@ -1,19 +1,20 @@
-// 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_sender/audio_sender.h"
+#include "media/cast/sender/audio_sender.h"
 
 #include "base/bind.h"
 #include "base/logging.h"
 #include "base/message_loop/message_loop.h"
-#include "media/cast/audio_sender/audio_encoder.h"
 #include "media/cast/cast_defines.h"
-#include "media/cast/rtcp/rtcp_defines.h"
-#include "media/cast/transport/cast_transport_config.h"
+#include "media/cast/net/cast_transport_config.h"
+#include "media/cast/net/rtcp/rtcp_defines.h"
+#include "media/cast/sender/audio_encoder.h"
 
 namespace media {
 namespace cast {
+namespace {
 
 const int kNumAggressiveReportsSentAtStart = 100;
 const int kMinSchedulingDelayMs = 1;
@@ -23,18 +24,25 @@ const int kMinSchedulingDelayMs = 1;
 // well.
 const int kAudioFrameRate = 100;
 
+// Helper function to compute the maximum unacked audio frames that is sent.
+int GetMaxUnackedFrames(base::TimeDelta target_delay) {
+  // As long as it doesn't go over |kMaxUnackedFrames|, it is okay to send more
+  // audio data than the target delay would suggest. Audio packets are tiny and
+  // receiver has the ability to drop any one of the packets.
+  // We send up to three times of the target delay of audio frames.
+  int frames =
+      1 + 3 * target_delay * kAudioFrameRate / base::TimeDelta::FromSeconds(1);
+  return std::min(kMaxUnackedFrames, frames);
+}
+}  // namespace
+
 AudioSender::AudioSender(scoped_refptr<CastEnvironment> cast_environment,
                          const AudioSenderConfig& audio_config,
-                         transport::CastTransportSender* const transport_sender)
+                         CastTransportSender* const transport_sender)
     : cast_environment_(cast_environment),
-      target_playout_delay_(base::TimeDelta::FromMilliseconds(
-          audio_config.rtp_config.max_delay_ms)),
+      target_playout_delay_(audio_config.target_playout_delay),
       transport_sender_(transport_sender),
-      max_unacked_frames_(
-          std::min(kMaxUnackedFrames,
-                   1 + static_cast<int>(target_playout_delay_ *
-                                        kAudioFrameRate /
-                                        base::TimeDelta::FromSeconds(1)))),
+      max_unacked_frames_(GetMaxUnackedFrames(target_playout_delay_)),
       configured_encoder_bitrate_(audio_config.bitrate),
       rtcp_(cast_environment,
             this,
@@ -43,7 +51,7 @@ AudioSender::AudioSender(scoped_refptr<CastEnvironment> cast_environment,
             NULL,
             audio_config.rtcp_mode,
             base::TimeDelta::FromMilliseconds(audio_config.rtcp_interval),
-            audio_config.rtp_config.ssrc,
+            audio_config.ssrc,
             audio_config.incoming_feedback_ssrc,
             audio_config.rtcp_c_name,
             AUDIO_EVENT),
@@ -60,7 +68,10 @@ AudioSender::AudioSender(scoped_refptr<CastEnvironment> cast_environment,
   if (!audio_config.use_external_encoder) {
     audio_encoder_.reset(
         new AudioEncoder(cast_environment,
-                         audio_config,
+                         audio_config.channels,
+                         audio_config.frequency,
+                         audio_config.bitrate,
+                         audio_config.codec,
                          base::Bind(&AudioSender::SendEncodedAudioFrame,
                                     weak_factory_.GetWeakPtr())));
     cast_initialization_status_ = audio_encoder_->InitializationResult();
@@ -69,12 +80,14 @@ AudioSender::AudioSender(scoped_refptr<CastEnvironment> cast_environment,
     cast_initialization_status_ = STATUS_AUDIO_UNINITIALIZED;
   }
 
-  media::cast::transport::CastTransportAudioConfig transport_config;
-  transport_config.codec = audio_config.codec;
-  transport_config.rtp.config = audio_config.rtp_config;
-  transport_config.frequency = audio_config.frequency;
-  transport_config.channels = audio_config.channels;
-  transport_config.rtp.max_outstanding_frames = max_unacked_frames_;
+  media::cast::CastTransportRtpConfig transport_config;
+  transport_config.ssrc = audio_config.ssrc;
+  transport_config.rtp_payload_type = audio_config.rtp_payload_type;
+  // TODO(miu): AudioSender needs to be like VideoSender in providing an upper
+  // limit on the number of in-flight frames.
+  transport_config.stored_frames = max_unacked_frames_;
+  transport_config.aes_key = audio_config.aes_key;
+  transport_config.aes_iv_mask = audio_config.aes_iv_mask;
   transport_sender_->InitializeAudio(transport_config);
 
   rtcp_.SetCastReceiverEventHistorySize(kReceiverRtcpEventHistorySize);
@@ -102,7 +115,7 @@ void AudioSender::InsertAudio(scoped_ptr<AudioBus> audio_bus,
 }
 
 void AudioSender::SendEncodedAudioFrame(
-    scoped_ptr<transport::EncodedFrame> encoded_frame) {
+    scoped_ptr<EncodedFrame> encoded_frame) {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
 
   const uint32 frame_id = encoded_frame->frame_id;
@@ -121,7 +134,7 @@ void AudioSender::SendEncodedAudioFrame(
   cast_environment_->Logging()->InsertEncodedFrameEvent(
       last_send_time_, FRAME_ENCODED, AUDIO_EVENT, encoded_frame->rtp_timestamp,
       frame_id, static_cast<int>(encoded_frame->data.size()),
-      encoded_frame->dependency == transport::EncodedFrame::KEY,
+      encoded_frame->dependency == EncodedFrame::KEY,
       configured_encoder_bitrate_);
   // Only use lowest 8 bits as key.
   frame_id_to_rtp_timestamp_[frame_id & 0xff] = encoded_frame->rtp_timestamp;
diff --git a/media/cast/audio_sender/audio_sender.h b/media/cast/sender/audio_sender.h
index 80cf8a4e9e..efaa2b3d03 100644
--- a/media/cast/audio_sender/audio_sender.h
+++ b/media/cast/sender/audio_sender.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_SENDER_H_
-#define MEDIA_CAST_AUDIO_SENDER_H_
+#ifndef MEDIA_CAST_SENDER_AUDIO_SENDER_H_
+#define MEDIA_CAST_SENDER_AUDIO_SENDER_H_
 
 #include "base/callback.h"
 #include "base/memory/ref_counted.h"
@@ -16,8 +16,8 @@
 #include "media/cast/cast_config.h"
 #include "media/cast/cast_environment.h"
 #include "media/cast/logging/logging_defines.h"
-#include "media/cast/rtcp/rtcp.h"
-#include "media/cast/rtp_timestamp_helper.h"
+#include "media/cast/net/rtcp/rtcp.h"
+#include "media/cast/sender/rtp_timestamp_helper.h"
 
 namespace media {
 namespace cast {
@@ -36,7 +36,7 @@ class AudioSender : public RtcpSenderFeedback,
  public:
   AudioSender(scoped_refptr<CastEnvironment> cast_environment,
               const AudioSenderConfig& audio_config,
-              transport::CastTransportSender* const transport_sender);
+              CastTransportSender* const transport_sender);
 
   virtual ~AudioSender();
 
@@ -82,7 +82,7 @@ class AudioSender : public RtcpSenderFeedback,
   bool AreTooManyFramesInFlight() const;
 
   // Called by the |audio_encoder_| with the next EncodedFrame to send.
-  void SendEncodedAudioFrame(scoped_ptr<transport::EncodedFrame> audio_frame);
+  void SendEncodedAudioFrame(scoped_ptr<EncodedFrame> audio_frame);
 
   const scoped_refptr<CastEnvironment> cast_environment_;
 
@@ -99,7 +99,7 @@ class AudioSender : public RtcpSenderFeedback,
   // process to the browser process over IPC, with the browser process being
   // responsible for "packetizing" the frames and pushing packets into the
   // network layer.
-  transport::CastTransportSender* const transport_sender_;
+  CastTransportSender* const transport_sender_;
 
   // Maximum number of outstanding frames before the encoding and sending of
   // new frames shall halt.
@@ -159,4 +159,4 @@ class AudioSender : public RtcpSenderFeedback,
 }  // namespace cast
 }  // namespace media
 
-#endif  // MEDIA_CAST_AUDIO_SENDER_H_
+#endif  // MEDIA_CAST_SENDER_AUDIO_SENDER_H_
diff --git a/media/cast/audio_sender/audio_sender_unittest.cc b/media/cast/sender/audio_sender_unittest.cc
index 51edd49602..ab923811aa 100644
--- a/media/cast/audio_sender/audio_sender_unittest.cc
+++ b/media/cast/sender/audio_sender_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.
 
@@ -9,24 +9,24 @@
 #include "base/memory/scoped_ptr.h"
 #include "base/test/simple_test_tick_clock.h"
 #include "media/base/media.h"
-#include "media/cast/audio_sender/audio_sender.h"
 #include "media/cast/cast_config.h"
 #include "media/cast/cast_environment.h"
-#include "media/cast/rtcp/rtcp.h"
+#include "media/cast/net/cast_transport_config.h"
+#include "media/cast/net/cast_transport_sender_impl.h"
+#include "media/cast/net/rtcp/rtcp.h"
+#include "media/cast/sender/audio_sender.h"
 #include "media/cast/test/fake_single_thread_task_runner.h"
 #include "media/cast/test/utility/audio_utility.h"
-#include "media/cast/transport/cast_transport_config.h"
-#include "media/cast/transport/cast_transport_sender_impl.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
 namespace media {
 namespace cast {
 
-class TestPacketSender : public transport::PacketSender {
+class TestPacketSender : public PacketSender {
  public:
   TestPacketSender() : number_of_rtp_packets_(0), number_of_rtcp_packets_(0) {}
 
-  virtual bool SendPacket(transport::PacketRef packet,
+  virtual bool SendPacket(PacketRef packet,
                           const base::Closure& cb) OVERRIDE {
     if (Rtcp::IsRtcpPacket(&packet->data[0], packet->data.size())) {
       ++number_of_rtcp_packets_;
@@ -65,21 +65,21 @@ class AudioSenderTest : public ::testing::Test {
                             task_runner_,
                             task_runner_,
                             task_runner_);
-    audio_config_.codec = transport::kOpus;
+    audio_config_.codec = CODEC_AUDIO_OPUS;
     audio_config_.use_external_encoder = false;
     audio_config_.frequency = kDefaultAudioSamplingRate;
     audio_config_.channels = 2;
     audio_config_.bitrate = kDefaultAudioEncoderBitrate;
-    audio_config_.rtp_config.payload_type = 127;
+    audio_config_.rtp_payload_type = 127;
 
     net::IPEndPoint dummy_endpoint;
 
-    transport_sender_.reset(new transport::CastTransportSenderImpl(
+    transport_sender_.reset(new CastTransportSenderImpl(
         NULL,
         testing_clock_,
         dummy_endpoint,
         base::Bind(&UpdateCastTransportStatus),
-        transport::BulkRawEventsCallback(),
+        BulkRawEventsCallback(),
         base::TimeDelta(),
         task_runner_,
         &transport_));
@@ -90,13 +90,13 @@ class AudioSenderTest : public ::testing::Test {
 
   virtual ~AudioSenderTest() {}
 
-  static void UpdateCastTransportStatus(transport::CastTransportStatus status) {
-    EXPECT_EQ(transport::TRANSPORT_AUDIO_INITIALIZED, status);
+  static void UpdateCastTransportStatus(CastTransportStatus status) {
+    EXPECT_EQ(TRANSPORT_AUDIO_INITIALIZED, status);
   }
 
   base::SimpleTestTickClock* testing_clock_;  // Owned by CastEnvironment.
   TestPacketSender transport_;
-  scoped_ptr<transport::CastTransportSenderImpl> transport_sender_;
+  scoped_ptr<CastTransportSenderImpl> transport_sender_;
   scoped_refptr<test::FakeSingleThreadTaskRunner> task_runner_;
   scoped_ptr<AudioSender> audio_sender_;
   scoped_refptr<CastEnvironment> cast_environment_;
diff --git a/media/cast/congestion_control/congestion_control.cc b/media/cast/sender/congestion_control.cc
index d24e0ac3d0..70fcfe9dc9 100644
--- a/media/cast/congestion_control/congestion_control.cc
+++ b/media/cast/sender/congestion_control.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.
 
@@ -13,7 +13,7 @@
 // If we estimate that our virtual buffer is mostly empty, we try to use
 // more bandwidth than our recent usage, otherwise we use less.
 
-#include "media/cast/congestion_control/congestion_control.h"
+#include "media/cast/sender/congestion_control.h"
 
 #include "base/logging.h"
 #include "media/cast/cast_config.h"
@@ -59,8 +59,7 @@ CongestionControl::CongestionControl(base::TickClock* clock,
 CongestionControl::~CongestionControl() {}
 
 void CongestionControl::UpdateRtt(base::TimeDelta rtt) {
-  rtt_ = base::TimeDelta::FromSecondsD(
-      (rtt_.InSecondsF() * 7 + rtt.InSecondsF()) / 8);
+  rtt_ = (7 * rtt_ + rtt) / 8;
 }
 
 // Calculate how much "dead air" there is between two frames.
diff --git a/media/cast/congestion_control/congestion_control.h b/media/cast/sender/congestion_control.h
index 54622ab114..8537037c71 100644
--- a/media/cast/congestion_control/congestion_control.h
+++ b/media/cast/sender/congestion_control.h
@@ -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.
 
@@ -54,7 +54,7 @@ class CongestionControl {
   // Get the FrameStats for a given |frame_id|.
   // Note: Older FrameStats will be removed automatically.
   FrameStats* GetFrameStats(uint32 frame_id);
-  // Calculata safe bitrate. This is based on how much we've been
+  // Calculate a safe bitrate. This is based on how much we've been
   // sending in the past.
   double CalculateSafeBitrate();
 
@@ -63,7 +63,7 @@ class CongestionControl {
   base::TimeTicks EstimatedAckTime(uint32 frame_id, double bitrate);
   // Calculate when we start sending the data for a given frame.
   // This is done by calculating when we were done sending the previous
-  // frame, but obvoiusly can't be less than |sent_time| (if known).
+  // frame, but obviously can't be less than |sent_time| (if known).
   base::TimeTicks EstimatedSendingTime(uint32 frame_id, double bitrate);
 
   base::TickClock* const clock_;  // Not owned by this class.
diff --git a/media/cast/congestion_control/congestion_control_unittest.cc b/media/cast/sender/congestion_control_unittest.cc
index 5745eab21d..afdce0031f 100644
--- a/media/cast/congestion_control/congestion_control_unittest.cc
+++ b/media/cast/sender/congestion_control_unittest.cc
@@ -1,12 +1,13 @@
-// 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 <stdint.h>
 
+#include "base/bind.h"
 #include "base/test/simple_test_tick_clock.h"
 #include "media/cast/cast_defines.h"
-#include "media/cast/congestion_control/congestion_control.h"
+#include "media/cast/sender/congestion_control.h"
 #include "media/cast/test/fake_single_thread_task_runner.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
diff --git a/media/cast/video_sender/external_video_encoder.cc b/media/cast/sender/external_video_encoder.cc
index ca30bcd47a..e3abecd407 100644
--- a/media/cast/video_sender/external_video_encoder.cc
+++ b/media/cast/sender/external_video_encoder.cc
@@ -2,7 +2,7 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-#include "media/cast/video_sender/external_video_encoder.h"
+#include "media/cast/sender/external_video_encoder.h"
 
 #include "base/bind.h"
 #include "base/logging.h"
@@ -13,7 +13,7 @@
 #include "media/base/video_util.h"
 #include "media/cast/cast_defines.h"
 #include "media/cast/logging/logging_defines.h"
-#include "media/cast/transport/cast_transport_config.h"
+#include "media/cast/net/cast_transport_config.h"
 #include "media/video/video_encode_accelerator.h"
 
 namespace media {
@@ -99,20 +99,19 @@ class LocalVideoEncodeAcceleratorClient
 
     VideoCodecProfile output_profile = media::VIDEO_CODEC_PROFILE_UNKNOWN;
     switch (video_config.codec) {
-      case transport::kVp8:
+      case CODEC_VIDEO_VP8:
         output_profile = media::VP8PROFILE_MAIN;
         break;
-      case transport::kH264:
+      case CODEC_VIDEO_H264:
         output_profile = media::H264PROFILE_MAIN;
         break;
-      case transport::kFakeSoftwareVideo:
+      case CODEC_VIDEO_FAKE:
         NOTREACHED() << "Fake software video encoder cannot be external";
         break;
-      case transport::kUnknownVideoCodec:
-        NOTREACHED() << "Video codec not specified";
+      default:
+        NOTREACHED() << "Video codec not specified or not supported";
         break;
     }
-    codec_ = video_config.codec;
     max_frame_rate_ = video_config.max_frame_rate;
 
     if (!video_encode_accelerator_->Initialize(
@@ -221,10 +220,10 @@ class LocalVideoEncodeAcceleratorClient
       stream_header_.append(static_cast<const char*>(output_buffer->memory()),
                             payload_size);
     } else if (!encoded_frame_data_storage_.empty()) {
-      scoped_ptr<transport::EncodedFrame> encoded_frame(
-          new transport::EncodedFrame());
-      encoded_frame->dependency = key_frame ? transport::EncodedFrame::KEY :
-          transport::EncodedFrame::DEPENDENT;
+      scoped_ptr<EncodedFrame> encoded_frame(
+          new EncodedFrame());
+      encoded_frame->dependency = key_frame ? EncodedFrame::KEY :
+          EncodedFrame::DEPENDENT;
       encoded_frame->frame_id = ++last_encoded_frame_id_;
       if (key_frame)
         encoded_frame->referenced_frame_id = encoded_frame->frame_id;
@@ -313,7 +312,6 @@ class LocalVideoEncodeAcceleratorClient
   const CreateVideoEncodeMemoryCallback create_video_encode_memory_cb_;
   const base::WeakPtr<ExternalVideoEncoder> weak_owner_;
   int max_frame_rate_;
-  transport::VideoCodec codec_;
   uint32 last_encoded_frame_id_;
   bool key_frame_encountered_;
   std::string stream_header_;
diff --git a/media/cast/video_sender/external_video_encoder.h b/media/cast/sender/external_video_encoder.h
index 29fe0c5fcd..84de7f08f4 100644
--- a/media/cast/video_sender/external_video_encoder.h
+++ b/media/cast/sender/external_video_encoder.h
@@ -2,14 +2,14 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-#ifndef MEDIA_CAST_VIDEO_SENDER_EXTERNAL_VIDEO_ENCODER_H_
-#define MEDIA_CAST_VIDEO_SENDER_EXTERNAL_VIDEO_ENCODER_H_
+#ifndef MEDIA_CAST_SENDER_EXTERNAL_VIDEO_ENCODER_H_
+#define MEDIA_CAST_SENDER_EXTERNAL_VIDEO_ENCODER_H_
 
 #include "base/memory/scoped_ptr.h"
 #include "base/memory/weak_ptr.h"
 #include "media/cast/cast_config.h"
 #include "media/cast/cast_environment.h"
-#include "media/cast/video_sender/video_encoder.h"
+#include "media/cast/sender/video_encoder.h"
 #include "media/video/video_encode_accelerator.h"
 
 namespace media {
@@ -83,4 +83,4 @@ class ExternalVideoEncoder : public VideoEncoder {
 }  // namespace cast
 }  // namespace media
 
-#endif  // MEDIA_CAST_VIDEO_SENDER_EXTERNAL_VIDEO_ENCODER_H_
+#endif  // MEDIA_CAST_SENDER_EXTERNAL_VIDEO_ENCODER_H_
diff --git a/media/cast/video_sender/external_video_encoder_unittest.cc b/media/cast/sender/external_video_encoder_unittest.cc
index 853258ce30..0b446f866e 100644
--- a/media/cast/video_sender/external_video_encoder_unittest.cc
+++ b/media/cast/sender/external_video_encoder_unittest.cc
@@ -10,10 +10,10 @@
 #include "media/base/video_frame.h"
 #include "media/cast/cast_defines.h"
 #include "media/cast/cast_environment.h"
+#include "media/cast/sender/external_video_encoder.h"
 #include "media/cast/test/fake_single_thread_task_runner.h"
 #include "media/cast/test/fake_video_encode_accelerator.h"
 #include "media/cast/test/utility/video_utility.h"
-#include "media/cast/video_sender/external_video_encoder.h"
 #include "testing/gmock/include/gmock/gmock.h"
 
 namespace media {
@@ -54,11 +54,11 @@ class TestVideoEncoderCallback
   }
 
   void DeliverEncodedVideoFrame(
-      scoped_ptr<transport::EncodedFrame> encoded_frame) {
+      scoped_ptr<EncodedFrame> encoded_frame) {
     if (expected_frame_id_ == expected_last_referenced_frame_id_) {
-      EXPECT_EQ(transport::EncodedFrame::KEY, encoded_frame->dependency);
+      EXPECT_EQ(EncodedFrame::KEY, encoded_frame->dependency);
     } else {
-      EXPECT_EQ(transport::EncodedFrame::DEPENDENT,
+      EXPECT_EQ(EncodedFrame::DEPENDENT,
                 encoded_frame->dependency);
     }
     EXPECT_EQ(expected_frame_id_, encoded_frame->frame_id);
@@ -86,9 +86,9 @@ class ExternalVideoEncoderTest : public ::testing::Test {
  protected:
   ExternalVideoEncoderTest()
       : test_video_encoder_callback_(new TestVideoEncoderCallback()) {
-    video_config_.rtp_config.ssrc = 1;
+    video_config_.ssrc = 1;
     video_config_.incoming_feedback_ssrc = 2;
-    video_config_.rtp_config.payload_type = 127;
+    video_config_.rtp_payload_type = 127;
     video_config_.use_external_encoder = true;
     video_config_.width = 320;
     video_config_.height = 240;
@@ -99,7 +99,7 @@ class ExternalVideoEncoderTest : public ::testing::Test {
     video_config_.min_qp = 0;
     video_config_.max_frame_rate = 30;
     video_config_.max_number_of_video_buffers_used = 3;
-    video_config_.codec = transport::kVp8;
+    video_config_.codec = CODEC_VIDEO_VP8;
     gfx::Size size(video_config_.width, video_config_.height);
     video_frame_ = media::VideoFrame::CreateFrame(
         VideoFrame::I420, size, gfx::Rect(size), size, base::TimeDelta());
diff --git a/media/cast/video_sender/fake_software_video_encoder.cc b/media/cast/sender/fake_software_video_encoder.cc
index 7c5c952641..bd96f78c89 100644
--- a/media/cast/video_sender/fake_software_video_encoder.cc
+++ b/media/cast/sender/fake_software_video_encoder.cc
@@ -2,11 +2,11 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-#include "media/cast/video_sender/fake_software_video_encoder.h"
+#include "media/cast/sender/fake_software_video_encoder.h"
 
 #include "base/json/json_writer.h"
 #include "base/values.h"
-#include "media/cast/transport/cast_transport_config.h"
+#include "media/cast/net/cast_transport_config.h"
 
 #ifndef OFFICIAL_BUILD
 
@@ -28,26 +28,26 @@ void FakeSoftwareVideoEncoder::Initialize() {}
 
 bool FakeSoftwareVideoEncoder::Encode(
     const scoped_refptr<media::VideoFrame>& video_frame,
-    transport::EncodedFrame* encoded_image) {
+    EncodedFrame* encoded_image) {
   encoded_image->frame_id = frame_id_++;
   if (next_frame_is_key_) {
-    encoded_image->dependency = transport::EncodedFrame::KEY;
+    encoded_image->dependency = EncodedFrame::KEY;
     encoded_image->referenced_frame_id = encoded_image->frame_id;
     next_frame_is_key_ = false;
   } else {
-    encoded_image->dependency = transport::EncodedFrame::DEPENDENT;
+    encoded_image->dependency = EncodedFrame::DEPENDENT;
     encoded_image->referenced_frame_id = encoded_image->frame_id - 1;
   }
 
   base::DictionaryValue values;
   values.SetBoolean("key",
-                    encoded_image->dependency == transport::EncodedFrame::KEY);
+                    encoded_image->dependency == EncodedFrame::KEY);
   values.SetInteger("ref", encoded_image->referenced_frame_id);
   values.SetInteger("id", encoded_image->frame_id);
   values.SetInteger("size", frame_size_);
   base::JSONWriter::Write(&values, &encoded_image->data);
   encoded_image->data.resize(
-      std::max<size_t>(encoded_image->data.size(), frame_size_));
+      std::max<size_t>(encoded_image->data.size(), frame_size_), ' ');
   return true;
 }
 
diff --git a/media/cast/video_sender/fake_software_video_encoder.h b/media/cast/sender/fake_software_video_encoder.h
index 0eb88ddfe1..5491ae0cc8 100644
--- a/media/cast/video_sender/fake_software_video_encoder.h
+++ b/media/cast/sender/fake_software_video_encoder.h
@@ -2,11 +2,11 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-#ifndef MEDIA_CAST_VIDEO_SENDER_FAKE_SOFTWARE_VIDEO_ENCODER_H_
-#define MEDIA_CAST_VIDEO_SENDER_FAKE_SOFTWARE_VIDEO_ENCODER_H_
+#ifndef MEDIA_CAST_SENDER_FAKE_SOFTWARE_VIDEO_ENCODER_H_
+#define MEDIA_CAST_SENDER_FAKE_SOFTWARE_VIDEO_ENCODER_H_
 
 #include "media/cast/cast_config.h"
-#include "media/cast/video_sender/software_video_encoder.h"
+#include "media/cast/sender/software_video_encoder.h"
 
 namespace media {
 namespace cast {
@@ -19,7 +19,7 @@ class FakeSoftwareVideoEncoder : public SoftwareVideoEncoder {
   // SoftwareVideoEncoder implementations.
   virtual void Initialize() OVERRIDE;
   virtual bool Encode(const scoped_refptr<media::VideoFrame>& video_frame,
-                      transport::EncodedFrame* encoded_image) OVERRIDE;
+                      EncodedFrame* encoded_image) OVERRIDE;
   virtual void UpdateRates(uint32 new_bitrate) OVERRIDE;
   virtual void GenerateKeyFrame() OVERRIDE;
   virtual void LatestFrameIdToReference(uint32 frame_id) OVERRIDE;
@@ -35,4 +35,4 @@ class FakeSoftwareVideoEncoder : public SoftwareVideoEncoder {
 }  // namespace cast
 }  // namespace media
 
-#endif  // MEDIA_CAST_VIDEO_SENDER_FAKE_SOFTWARE_VIDEO_ENCODER_H_
+#endif  // MEDIA_CAST_SENDER_FAKE_SOFTWARE_VIDEO_ENCODER_H_
diff --git a/media/cast/rtp_timestamp_helper.cc b/media/cast/sender/rtp_timestamp_helper.cc
index 3349e7b33f..ea0c35c66f 100644
--- a/media/cast/rtp_timestamp_helper.cc
+++ b/media/cast/sender/rtp_timestamp_helper.cc
@@ -2,7 +2,7 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-#include "media/cast/rtp_timestamp_helper.h"
+#include "media/cast/sender/rtp_timestamp_helper.h"
 
 namespace media {
 namespace cast {
diff --git a/media/cast/rtp_timestamp_helper.h b/media/cast/sender/rtp_timestamp_helper.h
index b9c650c506..8f56681dac 100644
--- a/media/cast/rtp_timestamp_helper.h
+++ b/media/cast/sender/rtp_timestamp_helper.h
@@ -2,8 +2,8 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-#ifndef MEDIA_CAST_RTP_TIMESTAMP_HELPER_H_
-#define MEDIA_CAST_RTP_TIMESTAMP_HELPER_H_
+#ifndef MEDIA_CAST_SENDER_RTP_TIMESTAMP_HELPER_H_
+#define MEDIA_CAST_SENDER_RTP_TIMESTAMP_HELPER_H_
 
 #include "base/basictypes.h"
 #include "base/time/time.h"
@@ -38,4 +38,4 @@ class RtpTimestampHelper {
 }  // namespace cast
 }  // namespace media
 
-#endif  // MEDIA_CAST_CAST_DEFINES_H_
+#endif  // MEDIA_CAST_SENDER_RTP_TIMESTAMP_HELPER_H_
diff --git a/media/cast/video_sender/software_video_encoder.h b/media/cast/sender/software_video_encoder.h
index f1bf6f6331..16c8cd3473 100644
--- a/media/cast/video_sender/software_video_encoder.h
+++ b/media/cast/sender/software_video_encoder.h
@@ -2,8 +2,8 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-#ifndef MEDIA_CAST_VIDEO_SENDER_SOFTWARE_VIDEO_ENCODER_H_
-#define MEDIA_CAST_VIDEO_SENDER_SOFTWARE_VIDEO_ENCODER_H_
+#ifndef MEDIA_CAST_SENDER_SOFTWARE_VIDEO_ENCODER_H_
+#define MEDIA_CAST_SENDER_SOFTWARE_VIDEO_ENCODER_H_
 
 #include "base/basictypes.h"
 #include "base/memory/ref_counted.h"
@@ -14,9 +14,7 @@ class VideoFrame;
 
 namespace media {
 namespace cast {
-namespace transport {
 struct EncodedFrame;
-}  // namespace transport
 
 class SoftwareVideoEncoder {
  public:
@@ -28,7 +26,7 @@ class SoftwareVideoEncoder {
 
   // Encode a raw image (as a part of a video stream).
   virtual bool Encode(const scoped_refptr<media::VideoFrame>& video_frame,
-                      transport::EncodedFrame* encoded_image) = 0;
+                      EncodedFrame* encoded_image) = 0;
 
   // Update the encoder with a new target bit rate.
   virtual void UpdateRates(uint32 new_bitrate) = 0;
@@ -43,4 +41,4 @@ class SoftwareVideoEncoder {
 }  // namespace cast
 }  // namespace media
 
-#endif  // MEDIA_CAST_VIDEO_SENDER_SOFTWARE_VIDEO_ENCODER_H_
+#endif  // MEDIA_CAST_SENDER_SOFTWARE_VIDEO_ENCODER_H_
diff --git a/media/cast/video_sender/video_encoder.h b/media/cast/sender/video_encoder.h
index c7b1049ce6..d788c7b2aa 100644
--- a/media/cast/video_sender/video_encoder.h
+++ b/media/cast/sender/video_encoder.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_SENDER_VIDEO_ENCODER_H_
-#define MEDIA_CAST_VIDEO_SENDER_VIDEO_ENCODER_H_
+#ifndef MEDIA_CAST_SENDER_VIDEO_ENCODER_H_
+#define MEDIA_CAST_SENDER_VIDEO_ENCODER_H_
 
 #include "base/callback.h"
 #include "base/memory/ref_counted.h"
@@ -20,7 +20,7 @@ namespace cast {
 // All these functions are called from the main cast thread.
 class VideoEncoder {
  public:
-  typedef base::Callback<void(scoped_ptr<transport::EncodedFrame>)>
+  typedef base::Callback<void(scoped_ptr<EncodedFrame>)>
       FrameEncodedCallback;
 
   virtual ~VideoEncoder() {}
@@ -48,4 +48,4 @@ class VideoEncoder {
 }  // namespace cast
 }  // namespace media
 
-#endif  // MEDIA_CAST_VIDEO_SENDER_VIDEO_ENCODER_H_
+#endif  // MEDIA_CAST_SENDER_VIDEO_ENCODER_H_
diff --git a/media/cast/video_sender/video_encoder_impl.cc b/media/cast/sender/video_encoder_impl.cc
index b90ef0f07e..d21649784e 100644
--- a/media/cast/video_sender/video_encoder_impl.cc
+++ b/media/cast/sender/video_encoder_impl.cc
@@ -2,7 +2,7 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-#include "media/cast/video_sender/video_encoder_impl.h"
+#include "media/cast/sender/video_encoder_impl.h"
 
 #include "base/bind.h"
 #include "base/bind_helpers.h"
@@ -11,8 +11,8 @@
 #include "base/message_loop/message_loop.h"
 #include "media/base/video_frame.h"
 #include "media/cast/cast_defines.h"
-#include "media/cast/video_sender/codecs/vp8/vp8_encoder.h"
-#include "media/cast/video_sender/fake_software_video_encoder.h"
+#include "media/cast/sender/fake_software_video_encoder.h"
+#include "media/cast/sender/vp8_encoder.h"
 
 namespace media {
 namespace cast {
@@ -43,8 +43,8 @@ void EncodeVideoFrameOnEncoderThread(
       dynamic_config.latest_frame_id_to_reference);
   encoder->UpdateRates(dynamic_config.bit_rate);
 
-  scoped_ptr<transport::EncodedFrame> encoded_frame(
-      new transport::EncodedFrame());
+  scoped_ptr<EncodedFrame> encoded_frame(
+      new EncodedFrame());
   if (!encoder->Encode(video_frame, encoded_frame.get())) {
     VLOG(1) << "Encoding failed";
     return;
@@ -53,7 +53,7 @@ void EncodeVideoFrameOnEncoderThread(
     VLOG(1) << "Encoding resulted in an empty frame";
     return;
   }
-  encoded_frame->rtp_timestamp = transport::GetVideoRtpTimestamp(capture_time);
+  encoded_frame->rtp_timestamp = GetVideoRtpTimestamp(capture_time);
   encoded_frame->reference_time = capture_time;
 
   environment->PostTask(
@@ -68,9 +68,8 @@ VideoEncoderImpl::VideoEncoderImpl(
     scoped_refptr<CastEnvironment> cast_environment,
     const VideoSenderConfig& video_config,
     int max_unacked_frames)
-    : video_config_(video_config),
-      cast_environment_(cast_environment) {
-  if (video_config.codec == transport::kVp8) {
+    : cast_environment_(cast_environment) {
+  if (video_config.codec == CODEC_VIDEO_VP8) {
     encoder_.reset(new Vp8Encoder(video_config, max_unacked_frames));
     cast_environment_->PostTask(CastEnvironment::VIDEO,
                                 FROM_HERE,
@@ -78,7 +77,7 @@ VideoEncoderImpl::VideoEncoderImpl(
                                            cast_environment,
                                            encoder_.get()));
 #ifndef OFFICIAL_BUILD
-  } else if (video_config.codec == transport::kFakeSoftwareVideo) {
+  } else if (video_config.codec == CODEC_VIDEO_FAKE) {
     encoder_.reset(new FakeSoftwareVideoEncoder(video_config));
 #endif
   } else {
diff --git a/media/cast/video_sender/video_encoder_impl.h b/media/cast/sender/video_encoder_impl.h
index b34b440c93..54a380265c 100644
--- a/media/cast/video_sender/video_encoder_impl.h
+++ b/media/cast/sender/video_encoder_impl.h
@@ -2,14 +2,14 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-#ifndef MEDIA_CAST_VIDEO_SENDER_VIDEO_ENCODER_IMPL_H_
-#define MEDIA_CAST_VIDEO_SENDER_VIDEO_ENCODER_IMPL_H_
+#ifndef MEDIA_CAST_SENDER_VIDEO_ENCODER_IMPL_H_
+#define MEDIA_CAST_SENDER_VIDEO_ENCODER_IMPL_H_
 
 #include "base/memory/scoped_ptr.h"
 #include "media/cast/cast_config.h"
 #include "media/cast/cast_environment.h"
-#include "media/cast/video_sender/software_video_encoder.h"
-#include "media/cast/video_sender/video_encoder.h"
+#include "media/cast/sender/software_video_encoder.h"
+#include "media/cast/sender/video_encoder.h"
 
 namespace media {
 class VideoFrame;
@@ -26,7 +26,7 @@ class VideoEncoderImpl : public VideoEncoder {
     int bit_rate;
   };
 
-  typedef base::Callback<void(scoped_ptr<transport::EncodedFrame>)>
+  typedef base::Callback<void(scoped_ptr<EncodedFrame>)>
       FrameEncodedCallback;
 
   VideoEncoderImpl(scoped_refptr<CastEnvironment> cast_environment,
@@ -53,7 +53,6 @@ class VideoEncoderImpl : public VideoEncoder {
   virtual void LatestFrameIdToReference(uint32 frame_id) OVERRIDE;
 
  private:
-  const VideoSenderConfig video_config_;
   scoped_refptr<CastEnvironment> cast_environment_;
   CodecDynamicConfig dynamic_config_;
 
@@ -69,4 +68,4 @@ class VideoEncoderImpl : public VideoEncoder {
 }  // namespace cast
 }  // namespace media
 
-#endif  // MEDIA_CAST_VIDEO_SENDER_VIDEO_ENCODER_IMPL_H_
+#endif  // MEDIA_CAST_SENDER_VIDEO_ENCODER_IMPL_H_
diff --git a/media/cast/video_sender/video_encoder_impl_unittest.cc b/media/cast/sender/video_encoder_impl_unittest.cc
index a60812304f..190ca2aacc 100644
--- a/media/cast/video_sender/video_encoder_impl_unittest.cc
+++ b/media/cast/sender/video_encoder_impl_unittest.cc
@@ -10,9 +10,9 @@
 #include "media/base/video_frame.h"
 #include "media/cast/cast_defines.h"
 #include "media/cast/cast_environment.h"
+#include "media/cast/sender/video_encoder_impl.h"
 #include "media/cast/test/fake_single_thread_task_runner.h"
 #include "media/cast/test/utility/video_utility.h"
-#include "media/cast/video_sender/video_encoder_impl.h"
 #include "testing/gmock/include/gmock/gmock.h"
 
 namespace media {
@@ -35,11 +35,11 @@ class TestVideoEncoderCallback
   }
 
   void DeliverEncodedVideoFrame(
-      scoped_ptr<transport::EncodedFrame> encoded_frame) {
+      scoped_ptr<EncodedFrame> encoded_frame) {
     if (expected_frame_id_ == expected_last_referenced_frame_id_) {
-      EXPECT_EQ(transport::EncodedFrame::KEY, encoded_frame->dependency);
+      EXPECT_EQ(EncodedFrame::KEY, encoded_frame->dependency);
     } else {
-      EXPECT_EQ(transport::EncodedFrame::DEPENDENT,
+      EXPECT_EQ(EncodedFrame::DEPENDENT,
                 encoded_frame->dependency);
     }
     EXPECT_EQ(expected_frame_id_, encoded_frame->frame_id);
@@ -66,9 +66,9 @@ class VideoEncoderImplTest : public ::testing::Test {
  protected:
   VideoEncoderImplTest()
       : test_video_encoder_callback_(new TestVideoEncoderCallback()) {
-    video_config_.rtp_config.ssrc = 1;
+    video_config_.ssrc = 1;
     video_config_.incoming_feedback_ssrc = 2;
-    video_config_.rtp_config.payload_type = 127;
+    video_config_.rtp_payload_type = 127;
     video_config_.use_external_encoder = false;
     video_config_.width = 320;
     video_config_.height = 240;
@@ -79,7 +79,7 @@ class VideoEncoderImplTest : public ::testing::Test {
     video_config_.min_qp = 0;
     video_config_.max_frame_rate = 30;
     video_config_.max_number_of_video_buffers_used = 3;
-    video_config_.codec = transport::kVp8;
+    video_config_.codec = CODEC_VIDEO_VP8;
     gfx::Size size(video_config_.width, video_config_.height);
     video_frame_ = media::VideoFrame::CreateFrame(
         VideoFrame::I420, size, gfx::Rect(size), size, base::TimeDelta());
diff --git a/media/cast/video_sender/video_sender.cc b/media/cast/sender/video_sender.cc
index cf050b7f10..0d826903fd 100644
--- a/media/cast/video_sender/video_sender.cc
+++ b/media/cast/sender/video_sender.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_sender/video_sender.h"
+#include "media/cast/sender/video_sender.h"
 
 #include <algorithm>
 #include <cstring>
@@ -12,10 +12,10 @@
 #include "base/logging.h"
 #include "base/message_loop/message_loop.h"
 #include "media/cast/cast_defines.h"
-#include "media/cast/rtcp/rtcp_defines.h"
-#include "media/cast/transport/cast_transport_config.h"
-#include "media/cast/video_sender/external_video_encoder.h"
-#include "media/cast/video_sender/video_encoder_impl.h"
+#include "media/cast/net/cast_transport_config.h"
+#include "media/cast/net/rtcp/rtcp_defines.h"
+#include "media/cast/sender/external_video_encoder.h"
+#include "media/cast/sender/video_encoder_impl.h"
 
 namespace media {
 namespace cast {
@@ -28,10 +28,9 @@ VideoSender::VideoSender(
     const VideoSenderConfig& video_config,
     const CreateVideoEncodeAcceleratorCallback& create_vea_cb,
     const CreateVideoEncodeMemoryCallback& create_video_encode_mem_cb,
-    transport::CastTransportSender* const transport_sender)
+    CastTransportSender* const transport_sender)
     : cast_environment_(cast_environment),
-      target_playout_delay_(base::TimeDelta::FromMilliseconds(
-          video_config.rtp_config.max_delay_ms)),
+      target_playout_delay_(video_config.target_playout_delay),
       transport_sender_(transport_sender),
       max_unacked_frames_(
           std::min(kMaxUnackedFrames,
@@ -45,7 +44,7 @@ VideoSender::VideoSender(
             NULL,
             video_config.rtcp_mode,
             base::TimeDelta::FromMilliseconds(video_config.rtcp_interval),
-            video_config.rtp_config.ssrc,
+            video_config.ssrc,
             video_config.incoming_feedback_ssrc,
             video_config.rtcp_c_name,
             VIDEO_EVENT),
@@ -75,10 +74,12 @@ VideoSender::VideoSender(
   }
   cast_initialization_status_ = STATUS_VIDEO_INITIALIZED;
 
-  media::cast::transport::CastTransportVideoConfig transport_config;
-  transport_config.codec = video_config.codec;
-  transport_config.rtp.config = video_config.rtp_config;
-  transport_config.rtp.max_outstanding_frames = max_unacked_frames_;
+  media::cast::CastTransportRtpConfig transport_config;
+  transport_config.ssrc = video_config.ssrc;
+  transport_config.rtp_payload_type = video_config.rtp_payload_type;
+  transport_config.stored_frames = max_unacked_frames_;
+  transport_config.aes_key = video_config.aes_key;
+  transport_config.aes_iv_mask = video_config.aes_iv_mask;
   transport_sender_->InitializeVideo(transport_config);
 
   rtcp_.SetCastReceiverEventHistorySize(kReceiverRtcpEventHistorySize);
@@ -140,7 +141,7 @@ void VideoSender::InsertRawVideoFrame(
 
 void VideoSender::SendEncodedVideoFrame(
     int requested_bitrate_before_encode,
-    scoped_ptr<transport::EncodedFrame> encoded_frame) {
+    scoped_ptr<EncodedFrame> encoded_frame) {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
 
   DCHECK_GT(frames_in_encoder_, 0);
@@ -159,13 +160,13 @@ void VideoSender::SendEncodedVideoFrame(
     ScheduleNextResendCheck();
   }
 
-  VLOG_IF(1, encoded_frame->dependency == transport::EncodedFrame::KEY)
+  VLOG_IF(1, encoded_frame->dependency == EncodedFrame::KEY)
       << "Send encoded key frame; frame_id: " << frame_id;
 
   cast_environment_->Logging()->InsertEncodedFrameEvent(
       last_send_time_, FRAME_ENCODED, VIDEO_EVENT, encoded_frame->rtp_timestamp,
       frame_id, static_cast<int>(encoded_frame->data.size()),
-      encoded_frame->dependency == transport::EncodedFrame::KEY,
+      encoded_frame->dependency == EncodedFrame::KEY,
       requested_bitrate_before_encode);
   // Only use lowest 8 bits as key.
   frame_id_to_rtp_timestamp_[frame_id & 0xff] = encoded_frame->rtp_timestamp;
diff --git a/media/cast/video_sender/video_sender.h b/media/cast/sender/video_sender.h
index cf8d27511c..6587572806 100644
--- a/media/cast/video_sender/video_sender.h
+++ b/media/cast/sender/video_sender.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_SENDER_VIDEO_SENDER_H_
-#define MEDIA_CAST_VIDEO_SENDER_VIDEO_SENDER_H_
+#ifndef MEDIA_CAST_SENDER_VIDEO_SENDER_H_
+#define MEDIA_CAST_SENDER_VIDEO_SENDER_H_
 
 #include "base/callback.h"
 #include "base/memory/ref_counted.h"
@@ -14,10 +14,10 @@
 #include "base/time/time.h"
 #include "media/cast/cast_config.h"
 #include "media/cast/cast_environment.h"
-#include "media/cast/congestion_control/congestion_control.h"
 #include "media/cast/logging/logging_defines.h"
-#include "media/cast/rtcp/rtcp.h"
-#include "media/cast/rtp_timestamp_helper.h"
+#include "media/cast/net/rtcp/rtcp.h"
+#include "media/cast/sender/congestion_control.h"
+#include "media/cast/sender/rtp_timestamp_helper.h"
 
 namespace media {
 
@@ -25,13 +25,10 @@ class VideoFrame;
 
 namespace cast {
 
+class CastTransportSender;
 class LocalVideoEncoderCallback;
 class VideoEncoder;
 
-namespace transport {
-class CastTransportSender;
-}
-
 // Not thread safe. Only called from the main cast thread.
 // This class owns all objects related to sending video, objects that create RTP
 // packets, congestion control, video encoder, parsing and sending of
@@ -46,7 +43,7 @@ class VideoSender : public RtcpSenderFeedback,
               const VideoSenderConfig& video_config,
               const CreateVideoEncodeAcceleratorCallback& create_vea_cb,
               const CreateVideoEncodeMemoryCallback& create_video_encode_mem_cb,
-              transport::CastTransportSender* const transport_sender);
+              CastTransportSender* const transport_sender);
 
   virtual ~VideoSender();
 
@@ -93,7 +90,7 @@ class VideoSender : public RtcpSenderFeedback,
 
   // Called by the |video_encoder_| with the next EncodeFrame to send.
   void SendEncodedVideoFrame(int requested_bitrate_before_encode,
-                             scoped_ptr<transport::EncodedFrame> encoded_frame);
+                             scoped_ptr<EncodedFrame> encoded_frame);
 
   const scoped_refptr<CastEnvironment> cast_environment_;
 
@@ -110,7 +107,7 @@ class VideoSender : public RtcpSenderFeedback,
   // process to the browser process over IPC, with the browser process being
   // responsible for "packetizing" the frames and pushing packets into the
   // network layer.
-  transport::CastTransportSender* const transport_sender_;
+  CastTransportSender* const transport_sender_;
 
   // Maximum number of outstanding frames before the encoding and sending of
   // new frames shall halt.
@@ -179,4 +176,4 @@ class VideoSender : public RtcpSenderFeedback,
 }  // namespace cast
 }  // namespace media
 
-#endif  // MEDIA_CAST_VIDEO_SENDER_VIDEO_SENDER_H_
+#endif  // MEDIA_CAST_SENDER_VIDEO_SENDER_H_
diff --git a/media/cast/video_sender/video_sender_unittest.cc b/media/cast/sender/video_sender_unittest.cc
index 49fae46c73..bccc248680 100644
--- a/media/cast/video_sender/video_sender_unittest.cc
+++ b/media/cast/sender/video_sender_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.
 
@@ -12,14 +12,14 @@
 #include "media/base/video_frame.h"
 #include "media/cast/cast_environment.h"
 #include "media/cast/logging/simple_event_subscriber.h"
+#include "media/cast/net/cast_transport_config.h"
+#include "media/cast/net/cast_transport_sender_impl.h"
+#include "media/cast/net/pacing/paced_sender.h"
+#include "media/cast/sender/video_sender.h"
 #include "media/cast/test/fake_single_thread_task_runner.h"
 #include "media/cast/test/fake_video_encode_accelerator.h"
 #include "media/cast/test/utility/default_config.h"
 #include "media/cast/test/utility/video_utility.h"
-#include "media/cast/transport/cast_transport_config.h"
-#include "media/cast/transport/cast_transport_sender_impl.h"
-#include "media/cast/transport/pacing/paced_sender.h"
-#include "media/cast/video_sender/video_sender.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
@@ -51,7 +51,7 @@ void CreateSharedMemory(
   callback.Run(shm.Pass());
 }
 
-class TestPacketSender : public transport::PacketSender {
+class TestPacketSender : public PacketSender {
  public:
   TestPacketSender()
       : number_of_rtp_packets_(0),
@@ -59,7 +59,7 @@ class TestPacketSender : public transport::PacketSender {
         paused_(false) {}
 
   // A singular packet implies a RTCP packet.
-  virtual bool SendPacket(transport::PacketRef packet,
+  virtual bool SendPacket(PacketRef packet,
                           const base::Closure& cb) OVERRIDE {
     if (paused_) {
       stored_packet_ = packet;
@@ -97,7 +97,7 @@ class TestPacketSender : public transport::PacketSender {
   int number_of_rtcp_packets_;
   bool paused_;
   base::Closure callback_;
-  transport::PacketRef stored_packet_;
+  PacketRef stored_packet_;
 
   DISALLOW_COPY_AND_ASSIGN(TestPacketSender);
 };
@@ -109,7 +109,7 @@ class PeerVideoSender : public VideoSender {
       const VideoSenderConfig& video_config,
       const CreateVideoEncodeAcceleratorCallback& create_vea_cb,
       const CreateVideoEncodeMemoryCallback& create_video_encode_mem_cb,
-      transport::CastTransportSender* const transport_sender)
+      CastTransportSender* const transport_sender)
       : VideoSender(cast_environment,
                     video_config,
                     create_vea_cb,
@@ -132,12 +132,12 @@ class VideoSenderTest : public ::testing::Test {
                             task_runner_);
     last_pixel_value_ = kPixelValue;
     net::IPEndPoint dummy_endpoint;
-    transport_sender_.reset(new transport::CastTransportSenderImpl(
+    transport_sender_.reset(new CastTransportSenderImpl(
         NULL,
         testing_clock_,
         dummy_endpoint,
         base::Bind(&UpdateCastTransportStatus),
-        transport::BulkRawEventsCallback(),
+        BulkRawEventsCallback(),
         base::TimeDelta(),
         task_runner_,
         &transport_));
@@ -150,16 +150,16 @@ class VideoSenderTest : public ::testing::Test {
     task_runner_->RunTasks();
   }
 
-  static void UpdateCastTransportStatus(transport::CastTransportStatus status) {
-    EXPECT_EQ(transport::TRANSPORT_VIDEO_INITIALIZED, status);
+  static void UpdateCastTransportStatus(CastTransportStatus status) {
+    EXPECT_EQ(TRANSPORT_VIDEO_INITIALIZED, status);
   }
 
   void InitEncoder(bool external) {
     VideoSenderConfig video_config;
-    video_config.rtp_config.ssrc = 1;
+    video_config.ssrc = 1;
     video_config.incoming_feedback_ssrc = 2;
     video_config.rtcp_c_name = "video_test@10.1.1.1";
-    video_config.rtp_config.payload_type = 127;
+    video_config.rtp_payload_type = 127;
     video_config.use_external_encoder = external;
     video_config.width = kWidth;
     video_config.height = kHeight;
@@ -170,7 +170,7 @@ class VideoSenderTest : public ::testing::Test {
     video_config.min_qp = 0;
     video_config.max_frame_rate = 30;
     video_config.max_number_of_video_buffers_used = 1;
-    video_config.codec = transport::kVp8;
+    video_config.codec = CODEC_VIDEO_VP8;
 
     if (external) {
       scoped_ptr<VideoEncodeAccelerator> fake_vea(
@@ -218,7 +218,7 @@ class VideoSenderTest : public ::testing::Test {
 
   base::SimpleTestTickClock* testing_clock_;  // Owned by CastEnvironment.
   TestPacketSender transport_;
-  scoped_ptr<transport::CastTransportSenderImpl> transport_sender_;
+  scoped_ptr<CastTransportSenderImpl> transport_sender_;
   scoped_refptr<test::FakeSingleThreadTaskRunner> task_runner_;
   scoped_ptr<PeerVideoSender> video_sender_;
   scoped_refptr<CastEnvironment> cast_environment_;
diff --git a/media/cast/video_sender/codecs/vp8/vp8_encoder.cc b/media/cast/sender/vp8_encoder.cc
index c7374babd1..b43b5c881e 100644
--- a/media/cast/video_sender/codecs/vp8/vp8_encoder.cc
+++ b/media/cast/sender/vp8_encoder.cc
@@ -1,17 +1,15 @@
-// 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.
-//
-// TODO (pwestin): add a link to the design document describing the generic
-// protocol and the VP8 specific details.
-#include "media/cast/video_sender/codecs/vp8/vp8_encoder.h"
+
+#include "media/cast/sender/vp8_encoder.h"
 
 #include <vector>
 
 #include "base/logging.h"
 #include "media/base/video_frame.h"
 #include "media/cast/cast_defines.h"
-#include "media/cast/transport/cast_transport_config.h"
+#include "media/cast/net/cast_transport_config.h"
 #include "third_party/libvpx/source/libvpx/vpx/vp8cx.h"
 
 namespace media {
@@ -135,7 +133,7 @@ void Vp8Encoder::InitEncode(int number_of_encode_threads) {
 }
 
 bool Vp8Encoder::Encode(const scoped_refptr<media::VideoFrame>& video_frame,
-                        transport::EncodedFrame* encoded_image) {
+                        EncodedFrame* encoded_image) {
   DCHECK(thread_checker_.CalledOnValidThread());
   // Image in vpx_image_t format.
   // Input image is const. VP8's raw image is not defined as const.
@@ -216,10 +214,10 @@ bool Vp8Encoder::Encode(const scoped_refptr<media::VideoFrame>& video_frame,
   // Populate the encoded frame.
   encoded_image->frame_id = ++last_encoded_frame_id_;
   if (is_key_frame) {
-    encoded_image->dependency = transport::EncodedFrame::KEY;
+    encoded_image->dependency = EncodedFrame::KEY;
     encoded_image->referenced_frame_id = encoded_image->frame_id;
   } else {
-    encoded_image->dependency = transport::EncodedFrame::DEPENDENT;
+    encoded_image->dependency = EncodedFrame::DEPENDENT;
     encoded_image->referenced_frame_id = latest_frame_id_to_reference;
   }
 
diff --git a/media/cast/video_sender/codecs/vp8/vp8_encoder.h b/media/cast/sender/vp8_encoder.h
index 2421cf1511..0437dbc2f5 100644
--- a/media/cast/video_sender/codecs/vp8/vp8_encoder.h
+++ b/media/cast/sender/vp8_encoder.h
@@ -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.
 
-#ifndef MEDIA_CAST_VIDEO_SENDER_CODECS_VP8_VP8_ENCODER_H_
-#define MEDIA_CAST_VIDEO_SENDER_CODECS_VP8_VP8_ENCODER_H_
+#ifndef MEDIA_CAST_SENDER_CODECS_VP8_VP8_ENCODER_H_
+#define MEDIA_CAST_SENDER_CODECS_VP8_VP8_ENCODER_H_
 
 #include "base/basictypes.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/threading/thread_checker.h"
 #include "base/time/time.h"
 #include "media/cast/cast_config.h"
-#include "media/cast/video_sender/software_video_encoder.h"
+#include "media/cast/sender/software_video_encoder.h"
 #include "third_party/libvpx/source/libvpx/vpx/vpx_encoder.h"
 
 namespace media {
@@ -37,7 +37,7 @@ class Vp8Encoder : public SoftwareVideoEncoder {
 
   // Encode a raw image (as a part of a video stream).
   virtual bool Encode(const scoped_refptr<media::VideoFrame>& video_frame,
-                      transport::EncodedFrame* encoded_image) OVERRIDE;
+                      EncodedFrame* encoded_image) OVERRIDE;
 
   // Update the encoder with a new target bit rate.
   virtual void UpdateRates(uint32 new_bitrate) OVERRIDE;
@@ -100,4 +100,4 @@ class Vp8Encoder : public SoftwareVideoEncoder {
 }  // namespace cast
 }  // namespace media
 
-#endif  // MEDIA_CAST_VIDEO_SENDER_CODECS_VP8_VP8_ENCODER_H_
+#endif  // MEDIA_CAST_SENDER_CODECS_VP8_VP8_ENCODER_H_
diff --git a/media/cast/test/cast_benchmarks.cc b/media/cast/test/cast_benchmarks.cc
index 66257626bd..a2483facff 100644
--- a/media/cast/test/cast_benchmarks.cc
+++ b/media/cast/test/cast_benchmarks.cc
@@ -45,7 +45,12 @@
 #include "media/cast/cast_receiver.h"
 #include "media/cast/cast_sender.h"
 #include "media/cast/logging/simple_event_subscriber.h"
+#include "media/cast/net/cast_transport_config.h"
+#include "media/cast/net/cast_transport_defines.h"
+#include "media/cast/net/cast_transport_sender.h"
+#include "media/cast/net/cast_transport_sender_impl.h"
 #include "media/cast/test/fake_single_thread_task_runner.h"
+#include "media/cast/test/loopback_transport.h"
 #include "media/cast/test/skewed_single_thread_task_runner.h"
 #include "media/cast/test/skewed_tick_clock.h"
 #include "media/cast/test/utility/audio_utility.h"
@@ -53,10 +58,6 @@
 #include "media/cast/test/utility/test_util.h"
 #include "media/cast/test/utility/udp_proxy.h"
 #include "media/cast/test/utility/video_utility.h"
-#include "media/cast/transport/cast_transport_config.h"
-#include "media/cast/transport/cast_transport_defines.h"
-#include "media/cast/transport/cast_transport_sender.h"
-#include "media/cast/transport/cast_transport_sender_impl.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
 namespace media {
@@ -68,15 +69,15 @@ static const int64 kStartMillisecond = INT64_C(1245);
 static const int kAudioChannels = 2;
 static const int kVideoHdWidth = 1280;
 static const int kVideoHdHeight = 720;
-static const int kTargetDelay = 300;
+static const int kTargetPlayoutDelayMs = 300;
 
 // The tests are commonly implemented with |kFrameTimerMs| RunTask function;
 // a normal video is 30 fps hence the 33 ms between frames.
 static const int kFrameTimerMs = 33;
 
-void UpdateCastTransportStatus(transport::CastTransportStatus status) {
-  bool result = (status == transport::TRANSPORT_AUDIO_INITIALIZED ||
-                 status == transport::TRANSPORT_VIDEO_INITIALIZED);
+void UpdateCastTransportStatus(CastTransportStatus status) {
+  bool result = (status == TRANSPORT_AUDIO_INITIALIZED ||
+                 status == TRANSPORT_VIDEO_INITIALIZED);
   EXPECT_TRUE(result);
 }
 
@@ -93,68 +94,9 @@ void IgnoreRawEvents(const std::vector<PacketEvent>& packet_events) {
 
 }  // namespace
 
-// Shim that turns forwards packets from a test::PacketPipe to a
-// PacketReceiverCallback.
-class LoopBackPacketPipe : public test::PacketPipe {
- public:
-  LoopBackPacketPipe(const transport::PacketReceiverCallback& packet_receiver)
-      : packet_receiver_(packet_receiver) {}
-
-  virtual ~LoopBackPacketPipe() {}
-
-  // PacketPipe implementations.
-  virtual void Send(scoped_ptr<transport::Packet> packet) OVERRIDE {
-    packet_receiver_.Run(packet.Pass());
-  }
-
- private:
-  transport::PacketReceiverCallback packet_receiver_;
-};
-
-// Class that sends the packet direct from sender into the receiver with the
-// ability to drop packets between the two.
-// TODO(hubbe): Break this out and share code with end2end_unittest.cc
-class LoopBackTransport : public transport::PacketSender {
- public:
-  explicit LoopBackTransport(scoped_refptr<CastEnvironment> cast_environment)
-      : cast_environment_(cast_environment) {}
-
-  void SetPacketReceiver(
-      const transport::PacketReceiverCallback& packet_receiver,
-      const scoped_refptr<base::SingleThreadTaskRunner>& task_runner,
-      base::TickClock* clock) {
-    scoped_ptr<test::PacketPipe> loopback_pipe(
-        new LoopBackPacketPipe(packet_receiver));
-    if (packet_pipe_) {
-      packet_pipe_->AppendToPipe(loopback_pipe.Pass());
-    } else {
-      packet_pipe_ = loopback_pipe.Pass();
-    }
-    packet_pipe_->InitOnIOThread(task_runner, clock);
-  }
-
-  virtual bool SendPacket(transport::PacketRef packet,
-                          const base::Closure& cb) OVERRIDE {
-    DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
-    scoped_ptr<Packet> packet_copy(new Packet(packet->data));
-    packet_pipe_->Send(packet_copy.Pass());
-    return true;
-  }
-
-  void SetPacketPipe(scoped_ptr<test::PacketPipe> pipe) {
-    // Append the loopback pipe to the end.
-    pipe->AppendToPipe(packet_pipe_.Pass());
-    packet_pipe_ = pipe.Pass();
-  }
-
- private:
-  scoped_refptr<CastEnvironment> cast_environment_;
-  scoped_ptr<test::PacketPipe> packet_pipe_;
-};
-
 // Wraps a CastTransportSender and records some statistics about
 // the data that goes through it.
-class CastTransportSenderWrapper : public transport::CastTransportSender {
+class CastTransportSenderWrapper : public CastTransportSender {
  public:
   // Takes ownership of |transport|.
   void Init(CastTransportSender* transport,
@@ -166,28 +108,28 @@ class CastTransportSenderWrapper : public transport::CastTransportSender {
   }
 
   virtual void InitializeAudio(
-      const transport::CastTransportAudioConfig& config) OVERRIDE {
+      const CastTransportRtpConfig& config) OVERRIDE {
     transport_->InitializeAudio(config);
   }
 
   virtual void InitializeVideo(
-      const transport::CastTransportVideoConfig& config) OVERRIDE {
+      const CastTransportRtpConfig& config) OVERRIDE {
     transport_->InitializeVideo(config);
   }
 
   virtual void SetPacketReceiver(
-      const transport::PacketReceiverCallback& packet_receiver) OVERRIDE {
+      const PacketReceiverCallback& packet_receiver) OVERRIDE {
     transport_->SetPacketReceiver(packet_receiver);
   }
 
   virtual void InsertCodedAudioFrame(
-      const transport::EncodedFrame& audio_frame) OVERRIDE {
+      const EncodedFrame& audio_frame) OVERRIDE {
     *encoded_audio_bytes_ += audio_frame.data.size();
     transport_->InsertCodedAudioFrame(audio_frame);
   }
 
   virtual void InsertCodedVideoFrame(
-      const transport::EncodedFrame& video_frame) OVERRIDE {
+      const EncodedFrame& video_frame) OVERRIDE {
     *encoded_video_bytes_ += video_frame.data.size();
     transport_->InsertCodedVideoFrame(video_frame);
   }
@@ -196,7 +138,7 @@ class CastTransportSenderWrapper : public transport::CastTransportSender {
                                      uint32 ntp_seconds,
                                      uint32 ntp_fraction,
                                      uint32 rtp_timestamp,
-                                     const transport::RtcpDlrrReportBlock& dlrr,
+                                     const RtcpDlrrReportBlock& dlrr,
                                      uint32 sending_ssrc,
                                      const std::string& c_name) OVERRIDE {
     transport_->SendRtcpFromRtpSender(packet_type_flags,
@@ -219,7 +161,7 @@ class CastTransportSenderWrapper : public transport::CastTransportSender {
   }
 
  private:
-  scoped_ptr<transport::CastTransportSender> transport_;
+  scoped_ptr<CastTransportSender> transport_;
   uint64* encoded_video_bytes_;
   uint64* encoded_audio_bytes_;
 };
@@ -278,34 +220,37 @@ class RunOneBenchmark {
         base::TimeDelta::FromMilliseconds(kStartMillisecond));
   }
 
-  void Configure(transport::VideoCodec video_codec,
-                 transport::AudioCodec audio_codec,
+  void Configure(Codec video_codec,
+                 Codec audio_codec,
                  int audio_sampling_frequency,
                  int max_number_of_video_buffers_used) {
-    audio_sender_config_.rtp_config.ssrc = 1;
+    audio_sender_config_.ssrc = 1;
     audio_sender_config_.incoming_feedback_ssrc = 2;
-    audio_sender_config_.rtp_config.payload_type = 96;
+    audio_sender_config_.target_playout_delay =
+        base::TimeDelta::FromMilliseconds(kTargetPlayoutDelayMs);
+    audio_sender_config_.rtp_payload_type = 96;
     audio_sender_config_.use_external_encoder = false;
     audio_sender_config_.frequency = audio_sampling_frequency;
     audio_sender_config_.channels = kAudioChannels;
     audio_sender_config_.bitrate = kDefaultAudioEncoderBitrate;
     audio_sender_config_.codec = audio_codec;
-    audio_sender_config_.rtp_config.max_delay_ms = kTargetDelay;
 
     audio_receiver_config_.feedback_ssrc =
         audio_sender_config_.incoming_feedback_ssrc;
-    audio_receiver_config_.incoming_ssrc = audio_sender_config_.rtp_config.ssrc;
+    audio_receiver_config_.incoming_ssrc = audio_sender_config_.ssrc;
     audio_receiver_config_.rtp_payload_type =
-        audio_sender_config_.rtp_config.payload_type;
+        audio_sender_config_.rtp_payload_type;
     audio_receiver_config_.frequency = audio_sender_config_.frequency;
     audio_receiver_config_.channels = kAudioChannels;
     audio_receiver_config_.max_frame_rate = 100;
-    audio_receiver_config_.codec.audio = audio_sender_config_.codec;
-    audio_receiver_config_.rtp_max_delay_ms = kTargetDelay;
+    audio_receiver_config_.codec = audio_sender_config_.codec;
+    audio_receiver_config_.rtp_max_delay_ms = kTargetPlayoutDelayMs;
 
-    video_sender_config_.rtp_config.ssrc = 3;
+    video_sender_config_.ssrc = 3;
     video_sender_config_.incoming_feedback_ssrc = 4;
-    video_sender_config_.rtp_config.payload_type = 97;
+    video_sender_config_.target_playout_delay =
+        base::TimeDelta::FromMilliseconds(kTargetPlayoutDelayMs);
+    video_sender_config_.rtp_payload_type = 97;
     video_sender_config_.use_external_encoder = false;
     video_sender_config_.width = kVideoHdWidth;
     video_sender_config_.height = kVideoHdHeight;
@@ -324,18 +269,17 @@ class RunOneBenchmark {
     video_sender_config_.max_number_of_video_buffers_used =
         max_number_of_video_buffers_used;
     video_sender_config_.codec = video_codec;
-    video_sender_config_.rtp_config.max_delay_ms = kTargetDelay;
 
     video_receiver_config_.feedback_ssrc =
         video_sender_config_.incoming_feedback_ssrc;
-    video_receiver_config_.incoming_ssrc = video_sender_config_.rtp_config.ssrc;
+    video_receiver_config_.incoming_ssrc = video_sender_config_.ssrc;
     video_receiver_config_.rtp_payload_type =
-        video_sender_config_.rtp_config.payload_type;
-    video_receiver_config_.codec.video = video_sender_config_.codec;
+        video_sender_config_.rtp_payload_type;
+    video_receiver_config_.codec = video_sender_config_.codec;
     video_receiver_config_.frequency = kVideoFrequency;
     video_receiver_config_.channels = 1;
     video_receiver_config_.max_frame_rate = 100;
-    video_receiver_config_.rtp_max_delay_ms = kTargetDelay;
+    video_receiver_config_.rtp_max_delay_ms = kTargetPlayoutDelayMs;
   }
 
   void SetSenderClockSkew(double skew, base::TimeDelta offset) {
@@ -348,13 +292,13 @@ class RunOneBenchmark {
     task_runner_receiver_->SetSkew(1.0 / skew);
   }
 
-  void Create() {
+  void Create(const MeasuringPoint& p) {
     cast_receiver_ = CastReceiver::Create(cast_environment_receiver_,
                                           audio_receiver_config_,
                                           video_receiver_config_,
                                           &receiver_to_sender_);
     net::IPEndPoint dummy_endpoint;
-    transport_sender_.Init(new transport::CastTransportSenderImpl(
+    transport_sender_.Init(new CastTransportSenderImpl(
                                NULL,
                                testing_clock_sender_,
                                dummy_endpoint,
@@ -377,10 +321,12 @@ class RunOneBenchmark {
                                   CreateDefaultVideoEncodeAcceleratorCallback(),
                                   CreateDefaultVideoEncodeMemoryCallback());
 
-    receiver_to_sender_.SetPacketReceiver(
-        cast_sender_->packet_receiver(), task_runner_, &testing_clock_);
-    sender_to_receiver_.SetPacketReceiver(
-        cast_receiver_->packet_receiver(), task_runner_, &testing_clock_);
+    receiver_to_sender_.Initialize(
+        CreateSimplePipe(p).Pass(), cast_sender_->packet_receiver(),
+        task_runner_, &testing_clock_);
+    sender_to_receiver_.Initialize(
+        CreateSimplePipe(p).Pass(), cast_receiver_->packet_receiver(),
+        task_runner_, &testing_clock_);
   }
 
   virtual ~RunOneBenchmark() {
@@ -436,10 +382,9 @@ class RunOneBenchmark {
 
   void Run(const MeasuringPoint& p) {
     available_bitrate_ = p.bitrate;
-    Configure(transport::kFakeSoftwareVideo, transport::kPcm16, 32000, 1);
-    receiver_to_sender_.SetPacketPipe(CreateSimplePipe(p).Pass());
-    sender_to_receiver_.SetPacketPipe(CreateSimplePipe(p).Pass());
-    Create();
+    Configure(
+        CODEC_VIDEO_FAKE, CODEC_AUDIO_PCM16, 32000, 1);
+    Create(p);
     StartBasicPlayer();
 
     for (int frame = 0; frame < 1000; frame++) {
diff --git a/media/cast/test/end2end_unittest.cc b/media/cast/test/end2end_unittest.cc
index 4a0d820214..a1d4a3ab34 100644
--- a/media/cast/test/end2end_unittest.cc
+++ b/media/cast/test/end2end_unittest.cc
@@ -29,6 +29,10 @@
 #include "media/cast/cast_receiver.h"
 #include "media/cast/cast_sender.h"
 #include "media/cast/logging/simple_event_subscriber.h"
+#include "media/cast/net/cast_transport_config.h"
+#include "media/cast/net/cast_transport_defines.h"
+#include "media/cast/net/cast_transport_sender.h"
+#include "media/cast/net/cast_transport_sender_impl.h"
 #include "media/cast/test/fake_single_thread_task_runner.h"
 #include "media/cast/test/skewed_single_thread_task_runner.h"
 #include "media/cast/test/skewed_tick_clock.h"
@@ -36,10 +40,6 @@
 #include "media/cast/test/utility/default_config.h"
 #include "media/cast/test/utility/udp_proxy.h"
 #include "media/cast/test/utility/video_utility.h"
-#include "media/cast/transport/cast_transport_config.h"
-#include "media/cast/transport/cast_transport_defines.h"
-#include "media/cast/transport/cast_transport_sender.h"
-#include "media/cast/transport/cast_transport_sender_impl.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
 namespace media {
@@ -97,9 +97,9 @@ std::string ConvertFromBase16String(const std::string base_16) {
   return compressed;
 }
 
-void UpdateCastTransportStatus(transport::CastTransportStatus status) {
-  bool result = (status == transport::TRANSPORT_AUDIO_INITIALIZED ||
-                 status == transport::TRANSPORT_VIDEO_INITIALIZED);
+void UpdateCastTransportStatus(CastTransportStatus status) {
+  bool result = (status == TRANSPORT_AUDIO_INITIALIZED ||
+                 status == TRANSPORT_VIDEO_INITIALIZED);
   EXPECT_TRUE(result);
 }
 
@@ -168,23 +168,23 @@ std::map<uint16, LoggingEventCounts> GetEventCountForPacketEvents(
 // PacketReceiverCallback.
 class LoopBackPacketPipe : public test::PacketPipe {
  public:
-  LoopBackPacketPipe(const transport::PacketReceiverCallback& packet_receiver)
+  LoopBackPacketPipe(const PacketReceiverCallback& packet_receiver)
       : packet_receiver_(packet_receiver) {}
 
   virtual ~LoopBackPacketPipe() {}
 
   // PacketPipe implementations.
-  virtual void Send(scoped_ptr<transport::Packet> packet) OVERRIDE {
+  virtual void Send(scoped_ptr<Packet> packet) OVERRIDE {
     packet_receiver_.Run(packet.Pass());
   }
 
  private:
-  transport::PacketReceiverCallback packet_receiver_;
+  PacketReceiverCallback packet_receiver_;
 };
 
 // Class that sends the packet direct from sender into the receiver with the
 // ability to drop packets between the two.
-class LoopBackTransport : public transport::PacketSender {
+class LoopBackTransport : public PacketSender {
  public:
   explicit LoopBackTransport(scoped_refptr<CastEnvironment> cast_environment)
       : send_packets_(true),
@@ -192,7 +192,7 @@ class LoopBackTransport : public transport::PacketSender {
         cast_environment_(cast_environment) {}
 
   void SetPacketReceiver(
-      const transport::PacketReceiverCallback& packet_receiver,
+      const PacketReceiverCallback& packet_receiver,
       const scoped_refptr<base::SingleThreadTaskRunner>& task_runner,
       base::TickClock* clock) {
     scoped_ptr<test::PacketPipe> loopback_pipe(
@@ -205,7 +205,7 @@ class LoopBackTransport : public transport::PacketSender {
     packet_pipe_->InitOnIOThread(task_runner, clock);
   }
 
-  virtual bool SendPacket(transport::PacketRef packet,
+  virtual bool SendPacket(PacketRef packet,
                           const base::Closure& cb) OVERRIDE {
     DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
     if (!send_packets_)
@@ -307,7 +307,7 @@ class TestReceiverAudioCallback
     EXPECT_TRUE(is_continuous);
   }
 
-  void CheckCodedAudioFrame(scoped_ptr<transport::EncodedFrame> audio_frame) {
+  void CheckCodedAudioFrame(scoped_ptr<EncodedFrame> audio_frame) {
     ASSERT_TRUE(!!audio_frame);
     ASSERT_FALSE(expected_frames_.empty());
     const ExpectedAudioFrame& expected_audio_frame =
@@ -458,14 +458,15 @@ class End2EndTest : public ::testing::Test {
         &event_subscriber_sender_);
   }
 
-  void Configure(transport::VideoCodec video_codec,
-                 transport::AudioCodec audio_codec,
+  void Configure(Codec video_codec,
+                 Codec audio_codec,
                  int audio_sampling_frequency,
                  int max_number_of_video_buffers_used) {
-    audio_sender_config_.rtp_config.ssrc = 1;
-    audio_sender_config_.rtp_config.max_delay_ms = kTargetPlayoutDelayMs;
+    audio_sender_config_.ssrc = 1;
     audio_sender_config_.incoming_feedback_ssrc = 2;
-    audio_sender_config_.rtp_config.payload_type = 96;
+    audio_sender_config_.target_playout_delay =
+        base::TimeDelta::FromMilliseconds(kTargetPlayoutDelayMs);
+    audio_sender_config_.rtp_payload_type = 96;
     audio_sender_config_.use_external_encoder = false;
     audio_sender_config_.frequency = audio_sampling_frequency;
     audio_sender_config_.channels = kAudioChannels;
@@ -474,22 +475,23 @@ class End2EndTest : public ::testing::Test {
 
     audio_receiver_config_.feedback_ssrc =
         audio_sender_config_.incoming_feedback_ssrc;
-    audio_receiver_config_.incoming_ssrc = audio_sender_config_.rtp_config.ssrc;
+    audio_receiver_config_.incoming_ssrc = audio_sender_config_.ssrc;
     audio_receiver_config_.rtp_max_delay_ms = kTargetPlayoutDelayMs;
     audio_receiver_config_.rtp_payload_type =
-        audio_sender_config_.rtp_config.payload_type;
+        audio_sender_config_.rtp_payload_type;
     audio_receiver_config_.frequency = audio_sender_config_.frequency;
     audio_receiver_config_.channels = kAudioChannels;
     audio_receiver_config_.max_frame_rate = 100;
-    audio_receiver_config_.codec.audio = audio_sender_config_.codec;
+    audio_receiver_config_.codec = audio_sender_config_.codec;
 
     test_receiver_audio_callback_->SetExpectedSamplingFrequency(
         audio_receiver_config_.frequency);
 
-    video_sender_config_.rtp_config.ssrc = 3;
-    video_sender_config_.rtp_config.max_delay_ms = kTargetPlayoutDelayMs;
+    video_sender_config_.ssrc = 3;
     video_sender_config_.incoming_feedback_ssrc = 4;
-    video_sender_config_.rtp_config.payload_type = 97;
+    video_sender_config_.target_playout_delay =
+        base::TimeDelta::FromMilliseconds(kTargetPlayoutDelayMs);
+    video_sender_config_.rtp_payload_type = 97;
     video_sender_config_.use_external_encoder = false;
     video_sender_config_.width = kVideoHdWidth;
     video_sender_config_.height = kVideoHdHeight;
@@ -505,14 +507,14 @@ class End2EndTest : public ::testing::Test {
 
     video_receiver_config_.feedback_ssrc =
         video_sender_config_.incoming_feedback_ssrc;
-    video_receiver_config_.incoming_ssrc = video_sender_config_.rtp_config.ssrc;
+    video_receiver_config_.incoming_ssrc = video_sender_config_.ssrc;
     video_receiver_config_.rtp_max_delay_ms = kTargetPlayoutDelayMs;
     video_receiver_config_.rtp_payload_type =
-        video_sender_config_.rtp_config.payload_type;
+        video_sender_config_.rtp_payload_type;
     video_receiver_config_.frequency = kVideoFrequency;
     video_receiver_config_.channels = 1;
     video_receiver_config_.max_frame_rate = video_sender_config_.max_frame_rate;
-    video_receiver_config_.codec.video = video_sender_config_.codec;
+    video_receiver_config_.codec = video_sender_config_.codec;
   }
 
   void SetReceiverSkew(double skew, base::TimeDelta offset) {
@@ -580,7 +582,7 @@ class End2EndTest : public ::testing::Test {
                                           &receiver_to_sender_);
 
     net::IPEndPoint dummy_endpoint;
-    transport_sender_.reset(new transport::CastTransportSenderImpl(
+    transport_sender_.reset(new CastTransportSenderImpl(
         NULL,
         testing_clock_sender_,
         dummy_endpoint,
@@ -760,7 +762,7 @@ class End2EndTest : public ::testing::Test {
 
   LoopBackTransport receiver_to_sender_;
   LoopBackTransport sender_to_receiver_;
-  scoped_ptr<transport::CastTransportSenderImpl> transport_sender_;
+  scoped_ptr<CastTransportSenderImpl> transport_sender_;
 
   scoped_ptr<CastReceiver> cast_receiver_;
   scoped_ptr<CastSender> cast_sender_;
@@ -782,7 +784,7 @@ class End2EndTest : public ::testing::Test {
 };
 
 TEST_F(End2EndTest, LoopNoLossPcm16) {
-  Configure(transport::kVp8, transport::kPcm16, 32000, 1);
+  Configure(CODEC_VIDEO_VP8, CODEC_AUDIO_PCM16, 32000, 1);
   // Reduce video resolution to allow processing multiple frames within a
   // reasonable time frame.
   video_sender_config_.width = kVideoQcifWidth;
@@ -836,7 +838,7 @@ TEST_F(End2EndTest, LoopNoLossPcm16) {
 // This tests our external decoder interface for Audio.
 // Audio test without packet loss using raw PCM 16 audio "codec";
 TEST_F(End2EndTest, LoopNoLossPcm16ExternalDecoder) {
-  Configure(transport::kVp8, transport::kPcm16, 32000, 1);
+  Configure(CODEC_VIDEO_VP8, CODEC_AUDIO_PCM16, 32000, 1);
   Create();
 
   const int kNumIterations = 10;
@@ -854,7 +856,8 @@ TEST_F(End2EndTest, LoopNoLossPcm16ExternalDecoder) {
 
 // This tests our Opus audio codec without video.
 TEST_F(End2EndTest, LoopNoLossOpus) {
-  Configure(transport::kVp8, transport::kOpus, kDefaultAudioSamplingRate, 1);
+  Configure(CODEC_VIDEO_VP8, CODEC_AUDIO_OPUS,
+            kDefaultAudioSamplingRate, 1);
   Create();
 
   const int kNumIterations = 300;
@@ -880,7 +883,8 @@ TEST_F(End2EndTest, LoopNoLossOpus) {
 // in audio_receiver.cc for likely cause(s) of this bug.
 // http://crbug.com/356942
 TEST_F(End2EndTest, DISABLED_StartSenderBeforeReceiver) {
-  Configure(transport::kVp8, transport::kPcm16, kDefaultAudioSamplingRate, 1);
+  Configure(CODEC_VIDEO_VP8, CODEC_AUDIO_PCM16,
+            kDefaultAudioSamplingRate, 1);
   Create();
 
   int video_start = kVideoStart;
@@ -968,8 +972,10 @@ TEST_F(End2EndTest, DISABLED_StartSenderBeforeReceiver) {
 // This tests a network glitch lasting for 10 video frames.
 // Flaky. See crbug.com/351596.
 TEST_F(End2EndTest, DISABLED_GlitchWith3Buffers) {
-  Configure(transport::kVp8, transport::kOpus, kDefaultAudioSamplingRate, 3);
-  video_sender_config_.rtp_config.max_delay_ms = 67;
+  Configure(CODEC_VIDEO_VP8, CODEC_AUDIO_OPUS,
+            kDefaultAudioSamplingRate, 3);
+  video_sender_config_.target_playout_delay =
+      base::TimeDelta::FromMilliseconds(67);
   video_receiver_config_.rtp_max_delay_ms = 67;
   Create();
 
@@ -1031,8 +1037,10 @@ TEST_F(End2EndTest, DISABLED_GlitchWith3Buffers) {
 
 // Disabled due to flakiness and crashiness.  http://crbug.com/360951
 TEST_F(End2EndTest, DISABLED_DropEveryOtherFrame3Buffers) {
-  Configure(transport::kVp8, transport::kOpus, kDefaultAudioSamplingRate, 3);
-  video_sender_config_.rtp_config.max_delay_ms = 67;
+  Configure(CODEC_VIDEO_VP8, CODEC_AUDIO_OPUS,
+            kDefaultAudioSamplingRate, 3);
+  video_sender_config_.target_playout_delay =
+      base::TimeDelta::FromMilliseconds(67);
   video_receiver_config_.rtp_max_delay_ms = 67;
   Create();
   sender_to_receiver_.DropAllPacketsBelongingToOddFrames();
@@ -1069,17 +1077,17 @@ TEST_F(End2EndTest, DISABLED_DropEveryOtherFrame3Buffers) {
 }
 
 TEST_F(End2EndTest, CryptoVideo) {
-  Configure(transport::kVp8, transport::kPcm16, 32000, 1);
+  Configure(CODEC_VIDEO_VP8, CODEC_AUDIO_PCM16, 32000, 1);
 
-  video_sender_config_.rtp_config.aes_iv_mask =
+  video_sender_config_.aes_iv_mask =
       ConvertFromBase16String("1234567890abcdeffedcba0987654321");
-  video_sender_config_.rtp_config.aes_key =
+  video_sender_config_.aes_key =
       ConvertFromBase16String("deadbeefcafeb0b0b0b0cafedeadbeef");
 
   video_receiver_config_.aes_iv_mask =
-      video_sender_config_.rtp_config.aes_iv_mask;
+      video_sender_config_.aes_iv_mask;
   video_receiver_config_.aes_key =
-      video_sender_config_.rtp_config.aes_key;
+      video_sender_config_.aes_key;
 
   Create();
 
@@ -1107,17 +1115,17 @@ TEST_F(End2EndTest, CryptoVideo) {
 }
 
 TEST_F(End2EndTest, CryptoAudio) {
-  Configure(transport::kVp8, transport::kPcm16, 32000, 1);
+  Configure(CODEC_VIDEO_VP8, CODEC_AUDIO_PCM16, 32000, 1);
 
-  audio_sender_config_.rtp_config.aes_iv_mask =
+  audio_sender_config_.aes_iv_mask =
       ConvertFromBase16String("abcdeffedcba12345678900987654321");
-    audio_sender_config_.rtp_config.aes_key =
+  audio_sender_config_.aes_key =
       ConvertFromBase16String("deadbeefcafecafedeadbeefb0b0b0b0");
 
   audio_receiver_config_.aes_iv_mask =
-      audio_sender_config_.rtp_config.aes_iv_mask;
+      audio_sender_config_.aes_iv_mask;
   audio_receiver_config_.aes_key =
-      audio_sender_config_.rtp_config.aes_key;
+      audio_sender_config_.aes_key;
 
   Create();
 
@@ -1136,7 +1144,7 @@ TEST_F(End2EndTest, CryptoAudio) {
 // Video test without packet loss - tests the logging aspects of the end2end,
 // but is basically equivalent to LoopNoLossPcm16.
 TEST_F(End2EndTest, VideoLogging) {
-  Configure(transport::kVp8, transport::kPcm16, 32000, 1);
+  Configure(CODEC_VIDEO_VP8, CODEC_AUDIO_PCM16, 32000, 1);
   Create();
 
   int video_start = kVideoStart;
@@ -1260,7 +1268,7 @@ TEST_F(End2EndTest, VideoLogging) {
 // Audio test without packet loss - tests the logging aspects of the end2end,
 // but is basically equivalent to LoopNoLossPcm16.
 TEST_F(End2EndTest, AudioLogging) {
-  Configure(transport::kVp8, transport::kPcm16, 32000, 1);
+  Configure(CODEC_VIDEO_VP8, CODEC_AUDIO_PCM16, 32000, 1);
   Create();
 
   int audio_diff = kFrameTimerMs;
@@ -1341,7 +1349,8 @@ TEST_F(End2EndTest, AudioLogging) {
 }
 
 TEST_F(End2EndTest, BasicFakeSoftwareVideo) {
-  Configure(transport::kFakeSoftwareVideo, transport::kPcm16, 32000, 1);
+  Configure(CODEC_VIDEO_FAKE, CODEC_AUDIO_PCM16, 32000,
+            1);
   Create();
   StartBasicPlayer();
   SetReceiverSkew(1.0, base::TimeDelta::FromMilliseconds(1));
@@ -1362,7 +1371,8 @@ TEST_F(End2EndTest, BasicFakeSoftwareVideo) {
 }
 
 TEST_F(End2EndTest, ReceiverClockFast) {
-  Configure(transport::kFakeSoftwareVideo, transport::kPcm16, 32000, 1);
+  Configure(CODEC_VIDEO_FAKE, CODEC_AUDIO_PCM16, 32000,
+            1);
   Create();
   StartBasicPlayer();
   SetReceiverSkew(2.0, base::TimeDelta::FromMicroseconds(1234567));
@@ -1377,7 +1387,8 @@ TEST_F(End2EndTest, ReceiverClockFast) {
 }
 
 TEST_F(End2EndTest, ReceiverClockSlow) {
-  Configure(transport::kFakeSoftwareVideo, transport::kPcm16, 32000, 1);
+  Configure(CODEC_VIDEO_FAKE, CODEC_AUDIO_PCM16, 32000,
+            1);
   Create();
   StartBasicPlayer();
   SetReceiverSkew(0.5, base::TimeDelta::FromMicroseconds(-765432));
@@ -1392,7 +1403,8 @@ TEST_F(End2EndTest, ReceiverClockSlow) {
 }
 
 TEST_F(End2EndTest, SmoothPlayoutWithFivePercentClockRateSkew) {
-  Configure(transport::kFakeSoftwareVideo, transport::kPcm16, 32000, 1);
+  Configure(CODEC_VIDEO_FAKE, CODEC_AUDIO_PCM16, 32000,
+            1);
   Create();
   StartBasicPlayer();
   SetReceiverSkew(1.05, base::TimeDelta::FromMilliseconds(-42));
@@ -1413,7 +1425,8 @@ TEST_F(End2EndTest, SmoothPlayoutWithFivePercentClockRateSkew) {
 }
 
 TEST_F(End2EndTest, EvilNetwork) {
-  Configure(transport::kFakeSoftwareVideo, transport::kPcm16, 32000, 1);
+  Configure(CODEC_VIDEO_FAKE, CODEC_AUDIO_PCM16, 32000,
+            1);
   receiver_to_sender_.SetPacketPipe(test::EvilNetwork().Pass());
   sender_to_receiver_.SetPacketPipe(test::EvilNetwork().Pass());
   Create();
diff --git a/media/cast/test/fake_media_source.cc b/media/cast/test/fake_media_source.cc
new file mode 100644
index 0000000000..07baebe6e3
--- /dev/null
+++ b/media/cast/test/fake_media_source.cc
@@ -0,0 +1,594 @@
+// 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/test/fake_media_source.h"
+
+#include "base/files/memory_mapped_file.h"
+#include "base/files/scoped_file.h"
+#include "base/logging.h"
+#include "base/strings/string_number_conversions.h"
+#include "media/audio/audio_parameters.h"
+#include "media/base/audio_buffer.h"
+#include "media/base/audio_bus.h"
+#include "media/base/audio_fifo.h"
+#include "media/base/audio_timestamp_helper.h"
+#include "media/base/media.h"
+#include "media/base/multi_channel_resampler.h"
+#include "media/base/video_frame.h"
+#include "media/base/video_util.h"
+#include "media/cast/cast_sender.h"
+#include "media/cast/test/utility/audio_utility.h"
+#include "media/cast/test/utility/video_utility.h"
+#include "media/ffmpeg/ffmpeg_common.h"
+#include "media/ffmpeg/ffmpeg_deleters.h"
+#include "media/filters/audio_renderer_algorithm.h"
+#include "media/filters/ffmpeg_demuxer.h"
+#include "media/filters/ffmpeg_glue.h"
+#include "media/filters/in_memory_url_protocol.h"
+#include "ui/gfx/size.h"
+
+namespace {
+
+static const int kAudioChannels = 2;
+static const int kAudioSamplingFrequency = 48000;
+static const int kSoundFrequency = 1234;  // Frequency of sinusoid wave.
+static const float kSoundVolume = 0.5f;
+static const int kAudioFrameMs = 10;  // Each audio frame is exactly 10ms.
+static const int kAudioPacketsPerSecond = 1000 / kAudioFrameMs;
+
+void AVFreeFrame(AVFrame* frame) {
+  av_frame_free(&frame);
+}
+
+}  // namespace
+
+namespace media {
+namespace cast {
+
+FakeMediaSource::FakeMediaSource(
+    scoped_refptr<base::SingleThreadTaskRunner> task_runner,
+    base::TickClock* clock,
+    const VideoSenderConfig& video_config)
+    : task_runner_(task_runner),
+      video_config_(video_config),
+      synthetic_count_(0),
+      clock_(clock),
+      audio_frame_count_(0),
+      video_frame_count_(0),
+      weak_factory_(this),
+      av_format_context_(NULL),
+      audio_stream_index_(-1),
+      playback_rate_(1.0),
+      video_stream_index_(-1),
+      video_frame_rate_numerator_(video_config.max_frame_rate),
+      video_frame_rate_denominator_(1),
+      video_first_pts_(0),
+      video_first_pts_set_(false) {
+  audio_bus_factory_.reset(new TestAudioBusFactory(kAudioChannels,
+                                                   kAudioSamplingFrequency,
+                                                   kSoundFrequency,
+                                                   kSoundVolume));
+}
+
+FakeMediaSource::~FakeMediaSource() {
+}
+
+void FakeMediaSource::SetSourceFile(const base::FilePath& video_file,
+                                    int override_fps) {
+  DCHECK(!video_file.empty());
+
+  if (override_fps) {
+    video_config_.max_frame_rate = override_fps;
+    video_frame_rate_numerator_ = override_fps;
+  }
+
+  LOG(INFO) << "Source: " << video_file.value();
+  if (!file_data_.Initialize(video_file)) {
+    LOG(ERROR) << "Cannot load file.";
+    return;
+  }
+  protocol_.reset(
+      new InMemoryUrlProtocol(file_data_.data(), file_data_.length(), false));
+  glue_.reset(new FFmpegGlue(protocol_.get()));
+
+  if (!glue_->OpenContext()) {
+    LOG(ERROR) << "Cannot open file.";
+    return;
+  }
+
+  // AVFormatContext is owned by the glue.
+  av_format_context_ = glue_->format_context();
+  if (avformat_find_stream_info(av_format_context_, NULL) < 0) {
+    LOG(ERROR) << "Cannot find stream information.";
+    return;
+  }
+
+  // Prepare FFmpeg decoders.
+  for (unsigned int i = 0; i < av_format_context_->nb_streams; ++i) {
+    AVStream* av_stream = av_format_context_->streams[i];
+    AVCodecContext* av_codec_context = av_stream->codec;
+    AVCodec* av_codec = avcodec_find_decoder(av_codec_context->codec_id);
+
+    if (!av_codec) {
+      LOG(ERROR) << "Cannot find decoder for the codec: "
+                 << av_codec_context->codec_id;
+      continue;
+    }
+
+    // Number of threads for decoding.
+    av_codec_context->thread_count = 2;
+    av_codec_context->error_concealment = FF_EC_GUESS_MVS | FF_EC_DEBLOCK;
+    av_codec_context->request_sample_fmt = AV_SAMPLE_FMT_S16;
+
+    if (avcodec_open2(av_codec_context, av_codec, NULL) < 0) {
+      LOG(ERROR) << "Cannot open AVCodecContext for the codec: "
+                 << av_codec_context->codec_id;
+      return;
+    }
+
+    if (av_codec->type == AVMEDIA_TYPE_AUDIO) {
+      if (av_codec_context->sample_fmt == AV_SAMPLE_FMT_S16P) {
+        LOG(ERROR) << "Audio format not supported.";
+        continue;
+      }
+      ChannelLayout layout = ChannelLayoutToChromeChannelLayout(
+          av_codec_context->channel_layout,
+          av_codec_context->channels);
+      if (layout == CHANNEL_LAYOUT_UNSUPPORTED) {
+        LOG(ERROR) << "Unsupported audio channels layout.";
+        continue;
+      }
+      if (audio_stream_index_ != -1) {
+        LOG(WARNING) << "Found multiple audio streams.";
+      }
+      audio_stream_index_ = static_cast<int>(i);
+      audio_params_.Reset(
+          AudioParameters::AUDIO_PCM_LINEAR,
+          layout,
+          av_codec_context->channels,
+          av_codec_context->channels,
+          av_codec_context->sample_rate,
+          8 * av_get_bytes_per_sample(av_codec_context->sample_fmt),
+          av_codec_context->sample_rate / kAudioPacketsPerSecond);
+      LOG(INFO) << "Source file has audio.";
+    } else if (av_codec->type == AVMEDIA_TYPE_VIDEO) {
+      VideoFrame::Format format =
+          PixelFormatToVideoFormat(av_codec_context->pix_fmt);
+      if (format != VideoFrame::YV12) {
+        LOG(ERROR) << "Cannot handle non YV12 video format: " << format;
+        continue;
+      }
+      if (video_stream_index_ != -1) {
+        LOG(WARNING) << "Found multiple video streams.";
+      }
+      video_stream_index_ = static_cast<int>(i);
+      if (!override_fps) {
+        video_frame_rate_numerator_ = av_stream->r_frame_rate.num;
+        video_frame_rate_denominator_ = av_stream->r_frame_rate.den;
+        // Max frame rate is rounded up.
+        video_config_.max_frame_rate =
+            video_frame_rate_denominator_ +
+            video_frame_rate_numerator_ - 1;
+        video_config_.max_frame_rate /= video_frame_rate_denominator_;
+      } else {
+        // If video is played at a manual speed audio needs to match.
+        playback_rate_ = 1.0 * override_fps *
+            av_stream->r_frame_rate.den /  av_stream->r_frame_rate.num;
+      }
+      LOG(INFO) << "Source file has video.";
+    } else {
+      LOG(ERROR) << "Unknown stream type; ignore.";
+    }
+  }
+
+  Rewind();
+}
+
+void FakeMediaSource::Start(scoped_refptr<AudioFrameInput> audio_frame_input,
+                            scoped_refptr<VideoFrameInput> video_frame_input) {
+  audio_frame_input_ = audio_frame_input;
+  video_frame_input_ = video_frame_input;
+
+  LOG(INFO) << "Max Frame rate: " << video_config_.max_frame_rate;
+  LOG(INFO) << "Real Frame rate: "
+            << video_frame_rate_numerator_ << "/"
+            << video_frame_rate_denominator_ << " fps.";
+  LOG(INFO) << "Audio playback rate: " << playback_rate_;
+
+  if (!is_transcoding_audio() && !is_transcoding_video()) {
+    // Send fake patterns.
+    task_runner_->PostTask(
+        FROM_HERE,
+        base::Bind(
+            &FakeMediaSource::SendNextFakeFrame,
+            base::Unretained(this)));
+    return;
+  }
+
+  // Send transcoding streams.
+  audio_algo_.Initialize(playback_rate_, audio_params_);
+  audio_algo_.FlushBuffers();
+  audio_fifo_input_bus_ =
+      AudioBus::Create(
+          audio_params_.channels(), audio_params_.frames_per_buffer());
+  // Audio FIFO can carry all data fron AudioRendererAlgorithm.
+  audio_fifo_.reset(
+      new AudioFifo(audio_params_.channels(),
+                    audio_algo_.QueueCapacity()));
+  audio_resampler_.reset(new media::MultiChannelResampler(
+      audio_params_.channels(),
+      static_cast<double>(audio_params_.sample_rate()) /
+      kAudioSamplingFrequency,
+      audio_params_.frames_per_buffer(),
+      base::Bind(&FakeMediaSource::ProvideData, base::Unretained(this))));
+  task_runner_->PostTask(
+      FROM_HERE,
+      base::Bind(
+          &FakeMediaSource::SendNextFrame,
+          base::Unretained(this)));
+}
+
+void FakeMediaSource::SendNextFakeFrame() {
+  gfx::Size size(video_config_.width, video_config_.height);
+  scoped_refptr<VideoFrame> video_frame =
+      VideoFrame::CreateBlackFrame(size);
+  PopulateVideoFrame(video_frame, synthetic_count_);
+  ++synthetic_count_;
+
+  base::TimeTicks now = clock_->NowTicks();
+  if (start_time_.is_null())
+    start_time_ = now;
+
+  base::TimeDelta video_time = VideoFrameTime(++video_frame_count_);
+  video_frame->set_timestamp(video_time);
+  video_frame_input_->InsertRawVideoFrame(video_frame,
+                                          start_time_ + video_time);
+
+  // Send just enough audio data to match next video frame's time.
+  base::TimeDelta audio_time = AudioFrameTime(audio_frame_count_);
+  while (audio_time < video_time) {
+    if (is_transcoding_audio()) {
+      Decode(true);
+      CHECK(!audio_bus_queue_.empty()) << "No audio decoded.";
+      scoped_ptr<AudioBus> bus(audio_bus_queue_.front());
+      audio_bus_queue_.pop();
+      audio_frame_input_->InsertAudio(
+          bus.Pass(), start_time_ + audio_time);
+    } else {
+      audio_frame_input_->InsertAudio(
+          audio_bus_factory_->NextAudioBus(
+              base::TimeDelta::FromMilliseconds(kAudioFrameMs)),
+          start_time_ + audio_time);
+    }
+    audio_time = AudioFrameTime(++audio_frame_count_);
+  }
+
+  // This is the time since the stream started.
+  const base::TimeDelta elapsed_time = now - start_time_;
+
+  // Handle the case when frame generation cannot keep up.
+  // Move the time ahead to match the next frame.
+  while (video_time < elapsed_time) {
+    LOG(WARNING) << "Skipping one frame.";
+    video_time = VideoFrameTime(++video_frame_count_);
+  }
+
+  task_runner_->PostDelayedTask(
+      FROM_HERE,
+      base::Bind(&FakeMediaSource::SendNextFakeFrame,
+                 weak_factory_.GetWeakPtr()),
+      video_time - elapsed_time);
+}
+
+bool FakeMediaSource::SendNextTranscodedVideo(base::TimeDelta elapsed_time) {
+  if (!is_transcoding_video())
+    return false;
+
+  Decode(false);
+  if (video_frame_queue_.empty())
+    return false;
+
+  scoped_refptr<VideoFrame> decoded_frame =
+      video_frame_queue_.front();
+  if (elapsed_time < decoded_frame->timestamp())
+    return false;
+
+  gfx::Size size(video_config_.width, video_config_.height);
+  scoped_refptr<VideoFrame> video_frame =
+      VideoFrame::CreateBlackFrame(size);
+  video_frame_queue_.pop();
+  media::CopyPlane(VideoFrame::kYPlane,
+                   decoded_frame->data(VideoFrame::kYPlane),
+                   decoded_frame->stride(VideoFrame::kYPlane),
+                   decoded_frame->rows(VideoFrame::kYPlane),
+                   video_frame);
+  media::CopyPlane(VideoFrame::kUPlane,
+                   decoded_frame->data(VideoFrame::kUPlane),
+                   decoded_frame->stride(VideoFrame::kUPlane),
+                   decoded_frame->rows(VideoFrame::kUPlane),
+                   video_frame);
+  media::CopyPlane(VideoFrame::kVPlane,
+                   decoded_frame->data(VideoFrame::kVPlane),
+                   decoded_frame->stride(VideoFrame::kVPlane),
+                   decoded_frame->rows(VideoFrame::kVPlane),
+                   video_frame);
+
+  base::TimeDelta video_time;
+  // Use the timestamp from the file if we're transcoding.
+  video_time = ScaleTimestamp(decoded_frame->timestamp());
+  video_frame_input_->InsertRawVideoFrame(
+      video_frame, start_time_ + video_time);
+
+  // Make sure queue is not empty.
+  Decode(false);
+  return true;
+}
+
+bool FakeMediaSource::SendNextTranscodedAudio(base::TimeDelta elapsed_time) {
+  if (!is_transcoding_audio())
+    return false;
+
+  Decode(true);
+  if (audio_bus_queue_.empty())
+    return false;
+
+  base::TimeDelta audio_time = audio_sent_ts_->GetTimestamp();
+  if (elapsed_time < audio_time)
+    return false;
+  scoped_ptr<AudioBus> bus(audio_bus_queue_.front());
+  audio_bus_queue_.pop();
+  audio_sent_ts_->AddFrames(bus->frames());
+  audio_frame_input_->InsertAudio(
+      bus.Pass(), start_time_ + audio_time);
+
+  // Make sure queue is not empty.
+  Decode(true);
+  return true;
+}
+
+void FakeMediaSource::SendNextFrame() {
+  if (start_time_.is_null())
+    start_time_ = clock_->NowTicks();
+  if (start_time_.is_null())
+    start_time_ = clock_->NowTicks();
+
+  // Send as much as possible. Audio is sent according to
+  // system time.
+  while (SendNextTranscodedAudio(clock_->NowTicks() - start_time_));
+
+  // Video is sync'ed to audio.
+  while (SendNextTranscodedVideo(audio_sent_ts_->GetTimestamp()));
+
+  if (audio_bus_queue_.empty() && video_frame_queue_.empty()) {
+    // Both queues are empty can only mean that we have reached
+    // the end of the stream.
+    LOG(INFO) << "Rewind.";
+    Rewind();
+    start_time_ = base::TimeTicks();
+    audio_sent_ts_.reset();
+    video_first_pts_set_ = false;
+  }
+
+  // Send next send.
+  task_runner_->PostDelayedTask(
+      FROM_HERE,
+      base::Bind(
+          &FakeMediaSource::SendNextFrame,
+          base::Unretained(this)),
+      base::TimeDelta::FromMilliseconds(kAudioFrameMs));
+}
+
+base::TimeDelta FakeMediaSource::VideoFrameTime(int frame_number) {
+  return frame_number * base::TimeDelta::FromSeconds(1) *
+      video_frame_rate_denominator_ / video_frame_rate_numerator_;
+}
+
+base::TimeDelta FakeMediaSource::ScaleTimestamp(base::TimeDelta timestamp) {
+  return base::TimeDelta::FromMicroseconds(
+      timestamp.InMicroseconds() / playback_rate_);
+}
+
+base::TimeDelta FakeMediaSource::AudioFrameTime(int frame_number) {
+  return frame_number * base::TimeDelta::FromMilliseconds(kAudioFrameMs);
+}
+
+void FakeMediaSource::Rewind() {
+  CHECK(av_seek_frame(av_format_context_, -1, 0, AVSEEK_FLAG_BACKWARD) >= 0)
+      << "Failed to rewind to the beginning.";
+}
+
+ScopedAVPacket FakeMediaSource::DemuxOnePacket(bool* audio) {
+  ScopedAVPacket packet(new AVPacket());
+  if (av_read_frame(av_format_context_, packet.get()) < 0) {
+    LOG(ERROR) << "Failed to read one AVPacket.";
+    packet.reset();
+    return packet.Pass();
+  }
+
+  int stream_index = static_cast<int>(packet->stream_index);
+  if (stream_index == audio_stream_index_) {
+    *audio = true;
+  } else if (stream_index == video_stream_index_) {
+    *audio = false;
+  } else {
+    // Ignore unknown packet.
+    LOG(INFO) << "Unknown packet.";
+    packet.reset();
+  }
+  return packet.Pass();
+}
+
+void FakeMediaSource::DecodeAudio(ScopedAVPacket packet) {
+  // Audio.
+  AVFrame* avframe = av_frame_alloc();
+
+  // Make a shallow copy of packet so we can slide packet.data as frames are
+  // decoded from the packet; otherwise av_free_packet() will corrupt memory.
+  AVPacket packet_temp = *packet.get();
+
+  do {
+    int frame_decoded = 0;
+    int result = avcodec_decode_audio4(
+        av_audio_context(), avframe, &frame_decoded, &packet_temp);
+    CHECK(result >= 0) << "Failed to decode audio.";
+    packet_temp.size -= result;
+    packet_temp.data += result;
+    if (!frame_decoded)
+      continue;
+
+    int frames_read = avframe->nb_samples;
+    if (frames_read < 0)
+      break;
+
+    if (!audio_sent_ts_) {
+      // Initialize the base time to the first packet in the file.
+      // This is set to the frequency we send to the receiver.
+      // Not the frequency of the source file. This is because we
+      // increment the frame count by samples we sent.
+      audio_sent_ts_.reset(
+          new AudioTimestampHelper(kAudioSamplingFrequency));
+      // For some files this is an invalid value.
+      base::TimeDelta base_ts;
+      audio_sent_ts_->SetBaseTimestamp(base_ts);
+    }
+
+    scoped_refptr<AudioBuffer> buffer =
+        AudioBuffer::CopyFrom(
+            AVSampleFormatToSampleFormat(
+                av_audio_context()->sample_fmt),
+            ChannelLayoutToChromeChannelLayout(
+                av_audio_context()->channel_layout,
+                av_audio_context()->channels),
+            av_audio_context()->channels,
+            av_audio_context()->sample_rate,
+            frames_read,
+            &avframe->data[0],
+            // Note: Not all files have correct values for pkt_pts.
+            base::TimeDelta::FromMilliseconds(avframe->pkt_pts));
+    audio_algo_.EnqueueBuffer(buffer);
+    av_frame_unref(avframe);
+  } while (packet_temp.size > 0);
+  av_frame_free(&avframe);
+
+  const int frames_needed_to_scale =
+      playback_rate_ * av_audio_context()->sample_rate /
+      kAudioPacketsPerSecond;
+  while (frames_needed_to_scale <= audio_algo_.frames_buffered()) {
+    if (!audio_algo_.FillBuffer(audio_fifo_input_bus_.get(),
+                                audio_fifo_input_bus_->frames())) {
+      // Nothing can be scaled. Decode some more.
+      return;
+    }
+
+    // Prevent overflow of audio data in the FIFO.
+    if (audio_fifo_input_bus_->frames() + audio_fifo_->frames()
+        <= audio_fifo_->max_frames()) {
+      audio_fifo_->Push(audio_fifo_input_bus_.get());
+    } else {
+      LOG(WARNING) << "Audio FIFO full; dropping samples.";
+    }
+
+    // Make sure there's enough data to resample audio.
+    if (audio_fifo_->frames() <
+        2 * audio_params_.sample_rate() / kAudioPacketsPerSecond) {
+      continue;
+    }
+
+    scoped_ptr<media::AudioBus> resampled_bus(
+        media::AudioBus::Create(
+            audio_params_.channels(),
+            kAudioSamplingFrequency / kAudioPacketsPerSecond));
+    audio_resampler_->Resample(resampled_bus->frames(),
+                               resampled_bus.get());
+    audio_bus_queue_.push(resampled_bus.release());
+  }
+}
+
+void FakeMediaSource::DecodeVideo(ScopedAVPacket packet) {
+  // Video.
+  int got_picture;
+  AVFrame* avframe = av_frame_alloc();
+  // Tell the decoder to reorder for us.
+  avframe->reordered_opaque =
+      av_video_context()->reordered_opaque = packet->pts;
+  CHECK(avcodec_decode_video2(
+      av_video_context(), avframe, &got_picture, packet.get()) >= 0)
+      << "Video decode error.";
+  if (!got_picture) {
+    av_frame_free(&avframe);
+    return;
+  }
+  gfx::Size size(av_video_context()->width, av_video_context()->height);
+  if (!video_first_pts_set_ ||
+      avframe->reordered_opaque < video_first_pts_) {
+    video_first_pts_set_ = true;
+    video_first_pts_ = avframe->reordered_opaque;
+  }
+  int64 pts = avframe->reordered_opaque - video_first_pts_;
+  video_frame_queue_.push(
+      VideoFrame::WrapExternalYuvData(
+          media::VideoFrame::YV12,
+          size,
+          gfx::Rect(size),
+          size,
+          avframe->linesize[0],
+          avframe->linesize[1],
+          avframe->linesize[2],
+          avframe->data[0],
+          avframe->data[1],
+          avframe->data[2],
+          base::TimeDelta::FromMilliseconds(pts),
+          base::Bind(&AVFreeFrame, avframe)));
+}
+
+void FakeMediaSource::Decode(bool decode_audio) {
+  // Read the stream until one video frame can be decoded.
+  while (true) {
+    if (decode_audio && !audio_bus_queue_.empty())
+      return;
+    if (!decode_audio && !video_frame_queue_.empty())
+      return;
+
+    bool audio_packet = false;
+    ScopedAVPacket packet = DemuxOnePacket(&audio_packet);
+    if (!packet) {
+      LOG(INFO) << "End of stream.";
+      return;
+    }
+
+    if (audio_packet)
+      DecodeAudio(packet.Pass());
+    else
+      DecodeVideo(packet.Pass());
+  }
+}
+
+void FakeMediaSource::ProvideData(int frame_delay,
+                                  media::AudioBus* output_bus) {
+  if (audio_fifo_->frames() >= output_bus->frames()) {
+    audio_fifo_->Consume(output_bus, 0, output_bus->frames());
+  } else {
+    LOG(WARNING) << "Not enough audio data for resampling.";
+    output_bus->Zero();
+  }
+}
+
+AVStream* FakeMediaSource::av_audio_stream() {
+  return av_format_context_->streams[audio_stream_index_];
+}
+
+AVStream* FakeMediaSource::av_video_stream() {
+  return av_format_context_->streams[video_stream_index_];
+}
+
+AVCodecContext* FakeMediaSource::av_audio_context() {
+  return av_audio_stream()->codec;
+}
+
+AVCodecContext* FakeMediaSource::av_video_context() {
+  return av_video_stream()->codec;
+}
+
+}  // namespace cast
+}  // namespace media
diff --git a/media/cast/test/fake_media_source.h b/media/cast/test/fake_media_source.h
new file mode 100644
index 0000000000..b18d44ea2a
--- /dev/null
+++ b/media/cast/test/fake_media_source.h
@@ -0,0 +1,152 @@
+// 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.
+
+// A fake media source that generates video and audio frames to a cast
+// sender.
+// This class can transcode a WebM file using FFmpeg. It can also
+// generate an animation and audio of fixed frequency.
+
+#ifndef MEDIA_CAST_TEST_FAKE_MEDIA_SOURCE_H_
+#define MEDIA_CAST_TEST_FAKE_MEDIA_SOURCE_H_
+
+#include <queue>
+
+#include "base/files/file_path.h"
+#include "base/files/memory_mapped_file.h"
+#include "base/memory/ref_counted.h"
+#include "base/memory/weak_ptr.h"
+#include "base/single_thread_task_runner.h"
+#include "base/time/tick_clock.h"
+#include "media/audio/audio_parameters.h"
+#include "media/cast/cast_config.h"
+#include "media/filters/audio_renderer_algorithm.h"
+#include "media/filters/ffmpeg_demuxer.h"
+
+struct AVCodecContext;
+struct AVFormatContext;
+
+namespace media {
+
+class AudioBus;
+class AudioFifo;
+class AudioTimestampHelper;
+class FFmpegGlue;
+class InMemoryUrlProtocol;
+class MultiChannelResampler;
+
+namespace cast {
+
+class AudioFrameInput;
+class VideoFrameInput;
+class TestAudioBusFactory;
+
+class FakeMediaSource {
+ public:
+  // |task_runner| is to schedule decoding tasks.
+  // |clock| is used by this source but is not owned.
+  // |video_config| is the desired video config.
+  FakeMediaSource(scoped_refptr<base::SingleThreadTaskRunner> task_runner,
+                  base::TickClock* clock,
+                  const VideoSenderConfig& video_config);
+  ~FakeMediaSource();
+
+  // Transcode this file as the source of video and audio frames.
+  // If |override_fps| is non zero then the file is played at the desired rate.
+  void SetSourceFile(const base::FilePath& video_file, int override_fps);
+
+  void Start(scoped_refptr<AudioFrameInput> audio_frame_input,
+             scoped_refptr<VideoFrameInput> video_frame_input);
+
+  const VideoSenderConfig& get_video_config() const { return video_config_; }
+
+ private:
+  bool is_transcoding_audio() const { return audio_stream_index_ >= 0; }
+  bool is_transcoding_video() const { return video_stream_index_ >= 0; }
+
+  void SendNextFrame();
+  void SendNextFakeFrame();
+
+  // Return true if a frame was sent.
+  bool SendNextTranscodedVideo(base::TimeDelta elapsed_time);
+
+  // Return true if a frame was sent.
+  bool SendNextTranscodedAudio(base::TimeDelta elapsed_time);
+
+  // Helper methods to compute timestamps for the frame number specified.
+  base::TimeDelta VideoFrameTime(int frame_number);
+
+  base::TimeDelta ScaleTimestamp(base::TimeDelta timestamp);
+
+  base::TimeDelta AudioFrameTime(int frame_number);
+
+  // Go to the beginning of the stream.
+  void Rewind();
+
+  // Call FFmpeg to fetch one packet.
+  ScopedAVPacket DemuxOnePacket(bool* audio);
+
+  void DecodeAudio(ScopedAVPacket packet);
+  void DecodeVideo(ScopedAVPacket packet);
+  void Decode(bool decode_audio);
+
+  void ProvideData(int frame_delay, media::AudioBus* output_bus);
+
+  AVStream* av_audio_stream();
+  AVStream* av_video_stream();
+  AVCodecContext* av_audio_context();
+  AVCodecContext* av_video_context();
+
+  scoped_refptr<base::SingleThreadTaskRunner> task_runner_;
+  VideoSenderConfig video_config_;
+  scoped_refptr<AudioFrameInput> audio_frame_input_;
+  scoped_refptr<VideoFrameInput> video_frame_input_;
+  uint8 synthetic_count_;
+  base::TickClock* const clock_;  // Not owned by this class.
+
+  // Time when the stream starts.
+  base::TimeTicks start_time_;
+
+  // The following three members are used only for fake frames.
+  int audio_frame_count_;  // Each audio frame is exactly 10ms.
+  int video_frame_count_;
+  scoped_ptr<TestAudioBusFactory> audio_bus_factory_;
+
+  // NOTE: Weak pointers must be invalidated before all other member variables.
+  base::WeakPtrFactory<FakeMediaSource> weak_factory_;
+
+  base::MemoryMappedFile file_data_;
+  scoped_ptr<InMemoryUrlProtocol> protocol_;
+  scoped_ptr<FFmpegGlue> glue_;
+  AVFormatContext* av_format_context_;
+
+  int audio_stream_index_;
+  AudioParameters audio_params_;
+  double playback_rate_;
+
+  int video_stream_index_;
+  int video_frame_rate_numerator_;
+  int video_frame_rate_denominator_;
+
+  // These are used for audio resampling.
+  scoped_ptr<media::MultiChannelResampler> audio_resampler_;
+  scoped_ptr<media::AudioFifo> audio_fifo_;
+  scoped_ptr<media::AudioBus> audio_fifo_input_bus_;
+  media::AudioRendererAlgorithm audio_algo_;
+
+  // Track the timestamp of audio sent to the receiver.
+  scoped_ptr<media::AudioTimestampHelper> audio_sent_ts_;
+
+  std::queue<scoped_refptr<VideoFrame> > video_frame_queue_;
+  int64 video_first_pts_;
+  bool video_first_pts_set_;
+
+  std::queue<AudioBus*> audio_bus_queue_;
+
+  DISALLOW_COPY_AND_ASSIGN(FakeMediaSource);
+};
+
+}  // namespace cast
+}  // namespace media
+
+#endif // MEDIA_CAST_TEST_FAKE_MEDIA_SOURCE_H_
diff --git a/media/cast/test/loopback_transport.cc b/media/cast/test/loopback_transport.cc
new file mode 100644
index 0000000000..3b72a7ef5a
--- /dev/null
+++ b/media/cast/test/loopback_transport.cc
@@ -0,0 +1,68 @@
+// 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/test/loopback_transport.h"
+
+#include "base/single_thread_task_runner.h"
+#include "base/time/tick_clock.h"
+#include "media/cast/test/utility/udp_proxy.h"
+
+namespace media {
+namespace cast {
+namespace {
+
+// Shim that turns forwards packets from a test::PacketPipe to a
+// PacketReceiverCallback.
+class LoopBackPacketPipe : public test::PacketPipe {
+ public:
+  LoopBackPacketPipe(
+      const PacketReceiverCallback& packet_receiver)
+      : packet_receiver_(packet_receiver) {}
+
+  virtual ~LoopBackPacketPipe() {}
+
+  // PacketPipe implementations.
+  virtual void Send(scoped_ptr<Packet> packet) OVERRIDE {
+    packet_receiver_.Run(packet.Pass());
+  }
+
+ private:
+  PacketReceiverCallback packet_receiver_;
+
+  DISALLOW_COPY_AND_ASSIGN(LoopBackPacketPipe);
+};
+
+}  // namespace
+
+LoopBackTransport::LoopBackTransport(
+    scoped_refptr<CastEnvironment> cast_environment)
+    : cast_environment_(cast_environment) {
+}
+
+LoopBackTransport::~LoopBackTransport() {
+}
+
+bool LoopBackTransport::SendPacket(PacketRef packet,
+                                   const base::Closure& cb) {
+  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
+  scoped_ptr<Packet> packet_copy(new Packet(packet->data));
+  packet_pipe_->Send(packet_copy.Pass());
+  return true;
+}
+
+void LoopBackTransport::Initialize(
+    scoped_ptr<test::PacketPipe> pipe,
+    const PacketReceiverCallback& packet_receiver,
+    const scoped_refptr<base::SingleThreadTaskRunner>& task_runner,
+    base::TickClock* clock) {
+  scoped_ptr<test::PacketPipe> loopback_pipe(
+      new LoopBackPacketPipe(packet_receiver));
+  // Append the loopback pipe to the end.
+  pipe->AppendToPipe(loopback_pipe.Pass());
+  packet_pipe_ = pipe.Pass();
+  packet_pipe_->InitOnIOThread(task_runner, clock);
+}
+
+}  // namespace cast
+}  // namespace media
diff --git a/media/cast/test/loopback_transport.h b/media/cast/test/loopback_transport.h
new file mode 100644
index 0000000000..8942caa4b9
--- /dev/null
+++ b/media/cast/test/loopback_transport.h
@@ -0,0 +1,55 @@
+// 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_TEST_LOOPBACK_TRANSPORT_H_
+#define MEDIA_CAST_TEST_LOOPBACK_TRANSPORT_H_
+
+#include "base/memory/ref_counted.h"
+#include "base/memory/scoped_ptr.h"
+#include "media/cast/cast_environment.h"
+#include "media/cast/net/cast_transport_config.h"
+
+namespace base {
+class SingleThreadTaskRunner;
+class TickClock;
+}  // namespace base
+
+namespace media {
+namespace cast {
+
+namespace test {
+class PacketPipe;
+}  // namespace test
+
+// Class that sends the packet to a receiver through a stack of PacketPipes.
+class LoopBackTransport : public PacketSender {
+ public:
+  explicit LoopBackTransport(
+      scoped_refptr<CastEnvironment> cast_environment);
+  virtual ~LoopBackTransport();
+
+  virtual bool SendPacket(PacketRef packet,
+                          const base::Closure& cb) OVERRIDE;
+
+  // Initiailize this loopback transport.
+  // Establish a flow of packets from |pipe| to |packet_receiver|.
+  // The data flow looks like:
+  // SendPacket() -> |pipe| -> Fake loopback pipe -> |packet_receiver|.
+  void Initialize(
+      scoped_ptr<test::PacketPipe> pipe,
+      const PacketReceiverCallback& packet_receiver,
+      const scoped_refptr<base::SingleThreadTaskRunner>& task_runner,
+      base::TickClock* clock);
+
+ private:
+  const scoped_refptr<CastEnvironment> cast_environment_;
+  scoped_ptr<test::PacketPipe> packet_pipe_;
+
+  DISALLOW_COPY_AND_ASSIGN(LoopBackTransport);
+};
+
+}  // namespace cast
+}  // namespace media
+
+#endif  // MEDIA_CAST_TEST_LOOPBACK_TRANSPORT_H_
diff --git a/media/cast/test/proto/BUILD.gn b/media/cast/test/proto/BUILD.gn
new file mode 100644
index 0000000000..7ac71d2e7f
--- /dev/null
+++ b/media/cast/test/proto/BUILD.gn
@@ -0,0 +1,14 @@
+# 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.
+
+import("//third_party/protobuf/proto_library.gni")
+
+# GYP version: media/cast/cast.gyp:cast_logging_proto
+proto_library("cast_network_simulation_proto") {
+  visibility = ":proto"
+  sources = [
+    "network_simulation_model.proto",
+  ]
+}
+
diff --git a/media/cast/test/proto/network_simulation_model.proto b/media/cast/test/proto/network_simulation_model.proto
new file mode 100644
index 0000000000..902712f7fa
--- /dev/null
+++ b/media/cast/test/proto/network_simulation_model.proto
@@ -0,0 +1,27 @@
+// 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.
+
+// Contains parameters for a network simulation model.
+
+syntax = "proto2";
+
+option optimize_for = LITE_RUNTIME;
+
+package media.cast.proto;
+
+message NetworkSimulationModel {
+  optional NetworkSimulationModelType type = 1;
+  optional IPPModel ipp = 2;
+}
+
+enum NetworkSimulationModelType {
+  INTERRUPTED_POISSON_PROCESS = 1;
+}
+
+message IPPModel {
+  optional double coef_burstiness = 1;
+  optional double coef_variance = 2;
+  repeated double average_rate = 3;
+}
+
diff --git a/media/cast/test/receiver.cc b/media/cast/test/receiver.cc
index 9861d3834e..e87055420f 100644
--- a/media/cast/test/receiver.cc
+++ b/media/cast/test/receiver.cc
@@ -33,13 +33,13 @@
 #include "media/cast/cast_environment.h"
 #include "media/cast/cast_receiver.h"
 #include "media/cast/logging/logging_defines.h"
+#include "media/cast/net/udp_transport.h"
 #include "media/cast/test/utility/audio_utility.h"
 #include "media/cast/test/utility/barcode.h"
 #include "media/cast/test/utility/default_config.h"
 #include "media/cast/test/utility/in_process_receiver.h"
 #include "media/cast/test/utility/input_builder.h"
 #include "media/cast/test/utility/standalone_cast_environment.h"
-#include "media/cast/transport/transport/udp_transport.h"
 #include "net/base/net_util.h"
 
 #if defined(OS_LINUX)
diff --git a/media/cast/test/sender.cc b/media/cast/test/sender.cc
index e457e2c9fe..6f9074f17e 100644
--- a/media/cast/test/sender.cc
+++ b/media/cast/test/sender.cc
@@ -10,10 +10,7 @@
 #include "base/at_exit.h"
 #include "base/base_paths.h"
 #include "base/command_line.h"
-#include "base/file_util.h"
 #include "base/files/file_path.h"
-#include "base/files/memory_mapped_file.h"
-#include "base/files/scoped_file.h"
 #include "base/json/json_writer.h"
 #include "base/logging.h"
 #include "base/memory/scoped_ptr.h"
@@ -22,15 +19,8 @@
 #include "base/threading/thread.h"
 #include "base/time/default_tick_clock.h"
 #include "base/values.h"
-#include "media/audio/audio_parameters.h"
-#include "media/base/audio_buffer.h"
-#include "media/base/audio_bus.h"
-#include "media/base/audio_fifo.h"
-#include "media/base/audio_timestamp_helper.h"
 #include "media/base/media.h"
-#include "media/base/multi_channel_resampler.h"
 #include "media/base/video_frame.h"
-#include "media/base/video_util.h"
 #include "media/cast/cast_config.h"
 #include "media/cast/cast_environment.h"
 #include "media/cast/cast_sender.h"
@@ -40,28 +30,16 @@
 #include "media/cast/logging/proto/raw_events.pb.h"
 #include "media/cast/logging/receiver_time_offset_estimator_impl.h"
 #include "media/cast/logging/stats_event_subscriber.h"
-#include "media/cast/test/utility/audio_utility.h"
+#include "media/cast/net/cast_transport_defines.h"
+#include "media/cast/net/cast_transport_sender.h"
+#include "media/cast/net/udp_transport.h"
+#include "media/cast/test/fake_media_source.h"
 #include "media/cast/test/utility/default_config.h"
 #include "media/cast/test/utility/input_builder.h"
-#include "media/cast/test/utility/video_utility.h"
-#include "media/cast/transport/cast_transport_defines.h"
-#include "media/cast/transport/cast_transport_sender.h"
-#include "media/cast/transport/transport/udp_transport.h"
-#include "media/ffmpeg/ffmpeg_common.h"
-#include "media/ffmpeg/ffmpeg_deleters.h"
-#include "media/filters/audio_renderer_algorithm.h"
-#include "media/filters/ffmpeg_demuxer.h"
-#include "media/filters/ffmpeg_glue.h"
-#include "media/filters/in_memory_url_protocol.h"
-#include "ui/gfx/size.h"
 
 namespace {
 static const int kAudioChannels = 2;
 static const int kAudioSamplingFrequency = 48000;
-static const int kSoundFrequency = 1234;  // Frequency of sinusoid wave.
-static const float kSoundVolume = 0.5f;
-static const int kAudioFrameMs = 10;  // Each audio frame is exactly 10ms.
-static const int kAudioPacketsPerSecond = 1000 / kAudioFrameMs;
 
 // The max allowed size of serialized log.
 const int kMaxSerializedLogBytes = 10 * 1000 * 1000;
@@ -79,36 +57,32 @@ const int kMaxSerializedLogBytes = 10 * 1000 * 1000;
 //
 // --fps=xx
 //   Override framerate of the video stream.
-
 const char kSwitchAddress[] = "address";
 const char kSwitchPort[] = "port";
 const char kSwitchSourceFile[] = "source-file";
 const char kSwitchFps[] = "fps";
 
-}  // namespace
-
-namespace media {
-namespace cast {
-
-AudioSenderConfig GetAudioSenderConfig() {
-  AudioSenderConfig audio_config;
+media::cast::AudioSenderConfig GetAudioSenderConfig() {
+  media::cast::AudioSenderConfig audio_config;
 
   audio_config.rtcp_c_name = "audio_sender@a.b.c.d";
 
   audio_config.use_external_encoder = false;
   audio_config.frequency = kAudioSamplingFrequency;
   audio_config.channels = kAudioChannels;
-  audio_config.bitrate = 64000;
-  audio_config.codec = transport::kOpus;
-  audio_config.rtp_config.ssrc = 1;
+  audio_config.bitrate = 0;  // Use Opus auto-VBR mode.
+  audio_config.codec = media::cast::CODEC_AUDIO_OPUS;
+  audio_config.ssrc = 1;
   audio_config.incoming_feedback_ssrc = 2;
-  audio_config.rtp_config.payload_type = 127;
-  audio_config.rtp_config.max_delay_ms = 300;
+  audio_config.rtp_payload_type = 127;
+  // TODO(miu): The default in cast_defines.h is 100.  Should this be 100, and
+  // should receiver.cc's config also be 100?
+  audio_config.target_playout_delay = base::TimeDelta::FromMilliseconds(300);
   return audio_config;
 }
 
-VideoSenderConfig GetVideoSenderConfig() {
-  VideoSenderConfig video_config;
+media::cast::VideoSenderConfig GetVideoSenderConfig() {
+  media::cast::VideoSenderConfig video_config;
 
   video_config.rtcp_c_name = "video_sender@a.b.c.d";
   video_config.use_external_encoder = false;
@@ -124,7 +98,7 @@ VideoSenderConfig GetVideoSenderConfig() {
   video_config.start_bitrate = video_config.min_bitrate;
 
   // Codec.
-  video_config.codec = transport::kVp8;
+  video_config.codec = media::cast::CODEC_VIDEO_VP8;
   video_config.max_number_of_video_buffers_used = 1;
   video_config.number_of_encode_threads = 2;
 
@@ -133,620 +107,17 @@ VideoSenderConfig GetVideoSenderConfig() {
   video_config.max_qp = 40;
 
   // SSRCs and payload type. Don't change them.
-  video_config.rtp_config.ssrc = 11;
+  video_config.ssrc = 11;
   video_config.incoming_feedback_ssrc = 12;
-  video_config.rtp_config.payload_type = 96;
-  video_config.rtp_config.max_delay_ms = 300;
+  video_config.rtp_payload_type = 96;
+  // TODO(miu): The default in cast_defines.h is 100.  Should this be 100, and
+  // should receiver.cc's config also be 100?
+  video_config.target_playout_delay = base::TimeDelta::FromMilliseconds(300);
   return video_config;
 }
 
-void AVFreeFrame(AVFrame* frame) { av_frame_free(&frame); }
-
-class SendProcess {
- public:
-  SendProcess(scoped_refptr<base::SingleThreadTaskRunner> thread_proxy,
-              base::TickClock* clock,
-              const VideoSenderConfig& video_config)
-      : test_app_thread_proxy_(thread_proxy),
-        video_config_(video_config),
-        synthetic_count_(0),
-        clock_(clock),
-        audio_frame_count_(0),
-        video_frame_count_(0),
-        weak_factory_(this),
-        av_format_context_(NULL),
-        audio_stream_index_(-1),
-        playback_rate_(1.0),
-        video_stream_index_(-1),
-        video_frame_rate_numerator_(video_config.max_frame_rate),
-        video_frame_rate_denominator_(1),
-        video_first_pts_(0),
-        video_first_pts_set_(false) {
-    audio_bus_factory_.reset(new TestAudioBusFactory(kAudioChannels,
-                                                     kAudioSamplingFrequency,
-                                                     kSoundFrequency,
-                                                     kSoundVolume));
-    const CommandLine* cmd = CommandLine::ForCurrentProcess();
-    int override_fps = 0;
-    if (base::StringToInt(cmd->GetSwitchValueASCII(kSwitchFps),
-                          &override_fps)) {
-      video_config_.max_frame_rate = override_fps;
-      video_frame_rate_numerator_ = override_fps;
-    }
-
-    // Load source file and prepare FFmpeg demuxer.
-    base::FilePath source_path = cmd->GetSwitchValuePath(kSwitchSourceFile);
-    if (source_path.empty())
-      return;
-
-    LOG(INFO) << "Source: " << source_path.value();
-    if (!file_data_.Initialize(source_path)) {
-      LOG(ERROR) << "Cannot load file.";
-      return;
-    }
-    protocol_.reset(
-        new InMemoryUrlProtocol(file_data_.data(), file_data_.length(), false));
-    glue_.reset(new FFmpegGlue(protocol_.get()));
-
-    if (!glue_->OpenContext()) {
-      LOG(ERROR) << "Cannot open file.";
-      return;
-    }
-
-    // AVFormatContext is owned by the glue.
-    av_format_context_ = glue_->format_context();
-    if (avformat_find_stream_info(av_format_context_, NULL) < 0) {
-      LOG(ERROR) << "Cannot find stream information.";
-      return;
-    }
-
-    // Prepare FFmpeg decoders.
-    for (unsigned int i = 0; i < av_format_context_->nb_streams; ++i) {
-      AVStream* av_stream = av_format_context_->streams[i];
-      AVCodecContext* av_codec_context = av_stream->codec;
-      AVCodec* av_codec = avcodec_find_decoder(av_codec_context->codec_id);
-
-      if (!av_codec) {
-        LOG(ERROR) << "Cannot find decoder for the codec: "
-                   << av_codec_context->codec_id;
-        continue;
-      }
-
-      // Number of threads for decoding.
-      av_codec_context->thread_count = 2;
-      av_codec_context->error_concealment = FF_EC_GUESS_MVS | FF_EC_DEBLOCK;
-      av_codec_context->request_sample_fmt = AV_SAMPLE_FMT_S16;
-
-      if (avcodec_open2(av_codec_context, av_codec, NULL) < 0) {
-        LOG(ERROR) << "Cannot open AVCodecContext for the codec: "
-                   << av_codec_context->codec_id;
-        return;
-      }
-
-      if (av_codec->type == AVMEDIA_TYPE_AUDIO) {
-        if (av_codec_context->sample_fmt == AV_SAMPLE_FMT_S16P) {
-          LOG(ERROR) << "Audio format not supported.";
-          continue;
-        }
-        ChannelLayout layout = ChannelLayoutToChromeChannelLayout(
-            av_codec_context->channel_layout,
-            av_codec_context->channels);
-        if (layout == CHANNEL_LAYOUT_UNSUPPORTED) {
-          LOG(ERROR) << "Unsupported audio channels layout.";
-          continue;
-        }
-        if (audio_stream_index_ != -1) {
-          LOG(WARNING) << "Found multiple audio streams.";
-        }
-        audio_stream_index_ = static_cast<int>(i);
-        audio_params_.Reset(
-            AudioParameters::AUDIO_PCM_LINEAR,
-            layout,
-            av_codec_context->channels,
-            av_codec_context->channels,
-            av_codec_context->sample_rate,
-            8 * av_get_bytes_per_sample(av_codec_context->sample_fmt),
-            av_codec_context->sample_rate / kAudioPacketsPerSecond);
-        LOG(INFO) << "Source file has audio.";
-      } else if (av_codec->type == AVMEDIA_TYPE_VIDEO) {
-        VideoFrame::Format format =
-            PixelFormatToVideoFormat(av_codec_context->pix_fmt);
-        if (format != VideoFrame::YV12) {
-          LOG(ERROR) << "Cannot handle non YV12 video format: " << format;
-          continue;
-        }
-        if (video_stream_index_ != -1) {
-          LOG(WARNING) << "Found multiple video streams.";
-        }
-        video_stream_index_ = static_cast<int>(i);
-        if (!override_fps) {
-          video_frame_rate_numerator_ = av_stream->r_frame_rate.num;
-          video_frame_rate_denominator_ = av_stream->r_frame_rate.den;
-          // Max frame rate is rounded up.
-          video_config_.max_frame_rate =
-              video_frame_rate_denominator_ +
-              video_frame_rate_numerator_ - 1;
-          video_config_.max_frame_rate /= video_frame_rate_denominator_;
-        } else {
-          // If video is played at a manual speed audio needs to match.
-          playback_rate_ = 1.0 * override_fps *
-               av_stream->r_frame_rate.den /  av_stream->r_frame_rate.num;
-        }
-        LOG(INFO) << "Source file has video.";
-      } else {
-        LOG(ERROR) << "Unknown stream type; ignore.";
-      }
-    }
-
-    Rewind();
-  }
-
-  ~SendProcess() {
-  }
-
-  void Start(scoped_refptr<AudioFrameInput> audio_frame_input,
-             scoped_refptr<VideoFrameInput> video_frame_input) {
-    audio_frame_input_ = audio_frame_input;
-    video_frame_input_ = video_frame_input;
-
-    LOG(INFO) << "Max Frame rate: " << video_config_.max_frame_rate;
-    LOG(INFO) << "Real Frame rate: "
-              << video_frame_rate_numerator_ << "/"
-              << video_frame_rate_denominator_ << " fps.";
-    LOG(INFO) << "Audio playback rate: " << playback_rate_;
-
-    if (!is_transcoding_audio() && !is_transcoding_video()) {
-      // Send fake patterns.
-      test_app_thread_proxy_->PostTask(
-          FROM_HERE,
-          base::Bind(
-              &SendProcess::SendNextFakeFrame,
-              base::Unretained(this)));
-      return;
-    }
-
-    // Send transcoding streams.
-    audio_algo_.Initialize(playback_rate_, audio_params_);
-    audio_algo_.FlushBuffers();
-    audio_fifo_input_bus_ =
-        AudioBus::Create(
-            audio_params_.channels(), audio_params_.frames_per_buffer());
-    // Audio FIFO can carry all data fron AudioRendererAlgorithm.
-    audio_fifo_.reset(
-        new AudioFifo(audio_params_.channels(),
-                      audio_algo_.QueueCapacity()));
-    audio_resampler_.reset(new media::MultiChannelResampler(
-        audio_params_.channels(),
-        static_cast<double>(audio_params_.sample_rate()) /
-        kAudioSamplingFrequency,
-        audio_params_.frames_per_buffer(),
-        base::Bind(&SendProcess::ProvideData, base::Unretained(this))));
-    test_app_thread_proxy_->PostTask(
-        FROM_HERE,
-        base::Bind(
-            &SendProcess::SendNextFrame,
-            base::Unretained(this)));
-  }
-
-  void SendNextFakeFrame() {
-    gfx::Size size(video_config_.width, video_config_.height);
-    scoped_refptr<VideoFrame> video_frame =
-        VideoFrame::CreateBlackFrame(size);
-    PopulateVideoFrame(video_frame, synthetic_count_);
-    ++synthetic_count_;
-
-    base::TimeTicks now = clock_->NowTicks();
-    if (start_time_.is_null())
-      start_time_ = now;
-
-    base::TimeDelta video_time = VideoFrameTime(video_frame_count_);
-    video_frame->set_timestamp(video_time);
-    video_frame_input_->InsertRawVideoFrame(video_frame,
-                                            start_time_ + video_time);
-
-    // Send just enough audio data to match next video frame's time.
-    base::TimeDelta audio_time = AudioFrameTime(audio_frame_count_);
-    while (audio_time < video_time) {
-      if (is_transcoding_audio()) {
-        Decode(true);
-        CHECK(!audio_bus_queue_.empty()) << "No audio decoded.";
-        scoped_ptr<AudioBus> bus(audio_bus_queue_.front());
-        audio_bus_queue_.pop();
-        audio_frame_input_->InsertAudio(
-            bus.Pass(), start_time_ + audio_time);
-      } else {
-        audio_frame_input_->InsertAudio(
-            audio_bus_factory_->NextAudioBus(
-                base::TimeDelta::FromMilliseconds(kAudioFrameMs)),
-            start_time_ + audio_time);
-      }
-      audio_time = AudioFrameTime(++audio_frame_count_);
-    }
-
-    // This is the time since the stream started.
-    const base::TimeDelta elapsed_time = now - start_time_;
-
-    // Handle the case when frame generation cannot keep up.
-    // Move the time ahead to match the next frame.
-    while (video_time < elapsed_time) {
-      LOG(WARNING) << "Skipping one frame.";
-      video_time = VideoFrameTime(++video_frame_count_);
-    }
-
-    test_app_thread_proxy_->PostDelayedTask(
-        FROM_HERE,
-        base::Bind(&SendProcess::SendNextFakeFrame,
-                   weak_factory_.GetWeakPtr()),
-        video_time - elapsed_time);
-  }
-
-  // Return true if a frame was sent.
-  bool SendNextTranscodedVideo(base::TimeDelta elapsed_time) {
-    if (!is_transcoding_video())
-      return false;
-
-    Decode(false);
-    if (video_frame_queue_.empty())
-      return false;
-
-    scoped_refptr<VideoFrame> decoded_frame =
-        video_frame_queue_.front();
-    if (elapsed_time < decoded_frame->timestamp())
-      return false;
-
-    gfx::Size size(video_config_.width, video_config_.height);
-    scoped_refptr<VideoFrame> video_frame =
-        VideoFrame::CreateBlackFrame(size);
-    video_frame_queue_.pop();
-    media::CopyPlane(VideoFrame::kYPlane,
-                     decoded_frame->data(VideoFrame::kYPlane),
-                     decoded_frame->stride(VideoFrame::kYPlane),
-                     decoded_frame->rows(VideoFrame::kYPlane),
-                     video_frame);
-    media::CopyPlane(VideoFrame::kUPlane,
-                     decoded_frame->data(VideoFrame::kUPlane),
-                     decoded_frame->stride(VideoFrame::kUPlane),
-                     decoded_frame->rows(VideoFrame::kUPlane),
-                     video_frame);
-    media::CopyPlane(VideoFrame::kVPlane,
-                     decoded_frame->data(VideoFrame::kVPlane),
-                     decoded_frame->stride(VideoFrame::kVPlane),
-                     decoded_frame->rows(VideoFrame::kVPlane),
-                     video_frame);
-
-    base::TimeDelta video_time;
-    // Use the timestamp from the file if we're transcoding.
-    video_time = ScaleTimestamp(decoded_frame->timestamp());
-    video_frame_input_->InsertRawVideoFrame(
-        video_frame, start_time_ + video_time);
-
-    // Make sure queue is not empty.
-    Decode(false);
-    return true;
-  }
-
-  // Return true if a frame was sent.
-  bool SendNextTranscodedAudio(base::TimeDelta elapsed_time) {
-    if (!is_transcoding_audio())
-      return false;
-
-    Decode(true);
-    if (audio_bus_queue_.empty())
-      return false;
-
-    base::TimeDelta audio_time = audio_sent_ts_->GetTimestamp();
-    if (elapsed_time < audio_time)
-      return false;
-    scoped_ptr<AudioBus> bus(audio_bus_queue_.front());
-    audio_bus_queue_.pop();
-    audio_sent_ts_->AddFrames(bus->frames());
-    audio_frame_input_->InsertAudio(
-        bus.Pass(), start_time_ + audio_time);
-
-    // Make sure queue is not empty.
-    Decode(true);
-    return true;
-  }
-
-  void SendNextFrame() {
-    if (start_time_.is_null())
-      start_time_ = clock_->NowTicks();
-    if (start_time_.is_null())
-      start_time_ = clock_->NowTicks();
-
-    // Send as much as possible. Audio is sent according to
-    // system time.
-    while (SendNextTranscodedAudio(clock_->NowTicks() - start_time_));
-
-    // Video is sync'ed to audio.
-    while (SendNextTranscodedVideo(audio_sent_ts_->GetTimestamp()));
-
-    if (audio_bus_queue_.empty() && video_frame_queue_.empty()) {
-      // Both queues are empty can only mean that we have reached
-      // the end of the stream.
-      LOG(INFO) << "Rewind.";
-      Rewind();
-      start_time_ = base::TimeTicks();
-      audio_sent_ts_.reset();
-      video_first_pts_set_ = false;
-    }
-
-    // Send next send.
-    test_app_thread_proxy_->PostDelayedTask(
-        FROM_HERE,
-        base::Bind(
-            &SendProcess::SendNextFrame,
-            base::Unretained(this)),
-        base::TimeDelta::FromMilliseconds(kAudioFrameMs));
-  }
-
-  const VideoSenderConfig& get_video_config() const { return video_config_; }
-
- private:
-  bool is_transcoding_audio() { return audio_stream_index_ >= 0; }
-  bool is_transcoding_video() { return video_stream_index_ >= 0; }
-
-  // Helper methods to compute timestamps for the frame number specified.
-  base::TimeDelta VideoFrameTime(int frame_number) {
-    return frame_number * base::TimeDelta::FromSeconds(1) *
-        video_frame_rate_denominator_ / video_frame_rate_numerator_;
-  }
-
-  base::TimeDelta ScaleTimestamp(base::TimeDelta timestamp) {
-    return base::TimeDelta::FromMicroseconds(
-        timestamp.InMicroseconds() / playback_rate_);
-  }
-
-  base::TimeDelta AudioFrameTime(int frame_number) {
-    return frame_number * base::TimeDelta::FromMilliseconds(kAudioFrameMs);
-  }
-
-  // Go to the beginning of the stream.
-  void Rewind() {
-    CHECK(av_seek_frame(av_format_context_, -1, 0, AVSEEK_FLAG_BACKWARD) >= 0)
-        << "Failed to rewind to the beginning.";
-  }
-
-  // Call FFmpeg to fetch one packet.
-  ScopedAVPacket DemuxOnePacket(bool* audio) {
-    ScopedAVPacket packet(new AVPacket());
-    if (av_read_frame(av_format_context_, packet.get()) < 0) {
-      LOG(ERROR) << "Failed to read one AVPacket.";
-      packet.reset();
-      return packet.Pass();
-    }
-
-    int stream_index = static_cast<int>(packet->stream_index);
-    if (stream_index == audio_stream_index_) {
-      *audio = true;
-    } else if (stream_index == video_stream_index_) {
-      *audio = false;
-    } else {
-      // Ignore unknown packet.
-      LOG(INFO) << "Unknown packet.";
-      packet.reset();
-    }
-    return packet.Pass();
-  }
-
-  void DecodeAudio(ScopedAVPacket packet) {
-    // Audio.
-    AVFrame* avframe = av_frame_alloc();
-
-    // Make a shallow copy of packet so we can slide packet.data as frames are
-    // decoded from the packet; otherwise av_free_packet() will corrupt memory.
-    AVPacket packet_temp = *packet.get();
-
-    do {
-      int frame_decoded = 0;
-      int result = avcodec_decode_audio4(
-          av_audio_context(), avframe, &frame_decoded, &packet_temp);
-      CHECK(result >= 0) << "Failed to decode audio.";
-      packet_temp.size -= result;
-      packet_temp.data += result;
-      if (!frame_decoded)
-        continue;
-
-      int frames_read = avframe->nb_samples;
-      if (frames_read < 0)
-        break;
-
-      if (!audio_sent_ts_) {
-        // Initialize the base time to the first packet in the file.
-        // This is set to the frequency we send to the receiver.
-        // Not the frequency of the source file. This is because we
-        // increment the frame count by samples we sent.
-        audio_sent_ts_.reset(
-            new AudioTimestampHelper(kAudioSamplingFrequency));
-        // For some files this is an invalid value.
-        base::TimeDelta base_ts;
-        audio_sent_ts_->SetBaseTimestamp(base_ts);
-      }
-
-      scoped_refptr<AudioBuffer> buffer =
-          AudioBuffer::CopyFrom(
-              AVSampleFormatToSampleFormat(
-                  av_audio_context()->sample_fmt),
-              ChannelLayoutToChromeChannelLayout(
-                  av_audio_context()->channel_layout,
-                  av_audio_context()->channels),
-              av_audio_context()->channels,
-              av_audio_context()->sample_rate,
-              frames_read,
-              &avframe->data[0],
-              // Note: Not all files have correct values for pkt_pts.
-              base::TimeDelta::FromMilliseconds(avframe->pkt_pts));
-      audio_algo_.EnqueueBuffer(buffer);
-      av_frame_unref(avframe);
-    } while (packet_temp.size > 0);
-    av_frame_free(&avframe);
-
-    const int frames_needed_to_scale =
-        playback_rate_ * av_audio_context()->sample_rate /
-        kAudioPacketsPerSecond;
-    while (frames_needed_to_scale <= audio_algo_.frames_buffered()) {
-      if (!audio_algo_.FillBuffer(audio_fifo_input_bus_.get(),
-                                  audio_fifo_input_bus_->frames())) {
-        // Nothing can be scaled. Decode some more.
-        return;
-      }
-
-      // Prevent overflow of audio data in the FIFO.
-      if (audio_fifo_input_bus_->frames() + audio_fifo_->frames()
-          <= audio_fifo_->max_frames()) {
-        audio_fifo_->Push(audio_fifo_input_bus_.get());
-      } else {
-        LOG(WARNING) << "Audio FIFO full; dropping samples.";
-      }
-
-      // Make sure there's enough data to resample audio.
-      if (audio_fifo_->frames() <
-          2 * audio_params_.sample_rate() / kAudioPacketsPerSecond) {
-        continue;
-      }
-
-      scoped_ptr<media::AudioBus> resampled_bus(
-          media::AudioBus::Create(
-              audio_params_.channels(),
-              kAudioSamplingFrequency / kAudioPacketsPerSecond));
-      audio_resampler_->Resample(resampled_bus->frames(),
-                                 resampled_bus.get());
-      audio_bus_queue_.push(resampled_bus.release());
-    }
-  }
-
-  void DecodeVideo(ScopedAVPacket packet) {
-    // Video.
-    int got_picture;
-    AVFrame* avframe = av_frame_alloc();
-    // Tell the decoder to reorder for us.
-    avframe->reordered_opaque =
-        av_video_context()->reordered_opaque = packet->pts;
-    CHECK(avcodec_decode_video2(
-        av_video_context(), avframe, &got_picture, packet.get()) >= 0)
-        << "Video decode error.";
-    if (!got_picture) {
-      av_frame_free(&avframe);
-      return;
-    }
-    gfx::Size size(av_video_context()->width, av_video_context()->height);
-    if (!video_first_pts_set_ ||
-        avframe->reordered_opaque < video_first_pts_) {
-      video_first_pts_set_ = true;
-      video_first_pts_ = avframe->reordered_opaque;
-    }
-    int64 pts = avframe->reordered_opaque - video_first_pts_;
-    video_frame_queue_.push(
-        VideoFrame::WrapExternalYuvData(
-            media::VideoFrame::YV12,
-            size,
-            gfx::Rect(size),
-            size,
-            avframe->linesize[0],
-            avframe->linesize[1],
-            avframe->linesize[2],
-            avframe->data[0],
-            avframe->data[1],
-            avframe->data[2],
-            base::TimeDelta::FromMilliseconds(pts),
-            base::Bind(&AVFreeFrame, avframe)));
-  }
-
-  void Decode(bool decode_audio) {
-    // Read the stream until one video frame can be decoded.
-    while (true) {
-      if (decode_audio && !audio_bus_queue_.empty())
-        return;
-      if (!decode_audio && !video_frame_queue_.empty())
-        return;
-
-      bool audio_packet = false;
-      ScopedAVPacket packet = DemuxOnePacket(&audio_packet);
-      if (!packet) {
-        LOG(INFO) << "End of stream.";
-        return;
-      }
-
-      if (audio_packet)
-        DecodeAudio(packet.Pass());
-      else
-        DecodeVideo(packet.Pass());
-    }
-  }
-
-  void ProvideData(int frame_delay, media::AudioBus* output_bus) {
-    if (audio_fifo_->frames() >= output_bus->frames()) {
-      audio_fifo_->Consume(output_bus, 0, output_bus->frames());
-    } else {
-      LOG(WARNING) << "Not enough audio data for resampling.";
-      output_bus->Zero();
-    }
-  }
-
-  AVStream* av_audio_stream() {
-    return av_format_context_->streams[audio_stream_index_];
-  }
-  AVStream* av_video_stream() {
-    return av_format_context_->streams[video_stream_index_];
-  }
-  AVCodecContext* av_audio_context() { return av_audio_stream()->codec; }
-  AVCodecContext* av_video_context() { return av_video_stream()->codec; }
-
-  scoped_refptr<base::SingleThreadTaskRunner> test_app_thread_proxy_;
-  VideoSenderConfig video_config_;
-  scoped_refptr<AudioFrameInput> audio_frame_input_;
-  scoped_refptr<VideoFrameInput> video_frame_input_;
-  uint8 synthetic_count_;
-  base::TickClock* const clock_;  // Not owned by this class.
-
-  // Time when the stream starts.
-  base::TimeTicks start_time_;
-
-  // The following three members are used only for fake frames.
-  int audio_frame_count_;  // Each audio frame is exactly 10ms.
-  int video_frame_count_;
-  scoped_ptr<TestAudioBusFactory> audio_bus_factory_;
-
-  // NOTE: Weak pointers must be invalidated before all other member variables.
-  base::WeakPtrFactory<SendProcess> weak_factory_;
-
-  base::MemoryMappedFile file_data_;
-  scoped_ptr<InMemoryUrlProtocol> protocol_;
-  scoped_ptr<FFmpegGlue> glue_;
-  AVFormatContext* av_format_context_;
-
-  int audio_stream_index_;
-  AudioParameters audio_params_;
-  double playback_rate_;
-
-  int video_stream_index_;
-  int video_frame_rate_numerator_;
-  int video_frame_rate_denominator_;
-
-  // These are used for audio resampling.
-  scoped_ptr<media::MultiChannelResampler> audio_resampler_;
-  scoped_ptr<media::AudioFifo> audio_fifo_;
-  scoped_ptr<media::AudioBus> audio_fifo_input_bus_;
-  media::AudioRendererAlgorithm audio_algo_;
-
-  // Track the timestamp of audio sent to the receiver.
-  scoped_ptr<media::AudioTimestampHelper> audio_sent_ts_;
-
-  std::queue<scoped_refptr<VideoFrame> > video_frame_queue_;
-  int64 video_first_pts_;
-  bool video_first_pts_set_;
-
-  std::queue<AudioBus*> audio_bus_queue_;
-
-  DISALLOW_COPY_AND_ASSIGN(SendProcess);
-};
-
-}  // namespace cast
-}  // namespace media
-
-namespace {
 void UpdateCastTransportStatus(
-    media::cast::transport::CastTransportStatus status) {
+    media::cast::CastTransportStatus status) {
   VLOG(1) << "Transport status: " << status;
 }
 
@@ -902,10 +273,8 @@ int main(int argc, char** argv) {
   LOG(INFO) << "Sending to " << remote_ip_address << ":" << remote_port
             << ".";
 
-  media::cast::AudioSenderConfig audio_config =
-      media::cast::GetAudioSenderConfig();
-  media::cast::VideoSenderConfig video_config =
-      media::cast::GetVideoSenderConfig();
+  media::cast::AudioSenderConfig audio_config = GetAudioSenderConfig();
+  media::cast::VideoSenderConfig video_config = GetVideoSenderConfig();
 
   // Running transport on the main thread.
   // Setting up transport config.
@@ -922,14 +291,25 @@ int main(int argc, char** argv) {
           video_thread.message_loop_proxy()));
 
   // SendProcess initialization.
-  scoped_ptr<media::cast::SendProcess> send_process(
-      new media::cast::SendProcess(test_thread.message_loop_proxy(),
-                                   cast_environment->Clock(),
-                                   video_config));
+  scoped_ptr<media::cast::FakeMediaSource> fake_media_source(
+      new media::cast::FakeMediaSource(test_thread.message_loop_proxy(),
+                                       cast_environment->Clock(),
+                                       video_config));
+
+  int override_fps = 0;
+  if (!base::StringToInt(cmd->GetSwitchValueASCII(kSwitchFps),
+                         &override_fps)){
+    override_fps = 0;
+  }
+  base::FilePath source_path = cmd->GetSwitchValuePath(kSwitchSourceFile);
+  if (!source_path.empty()) {
+    LOG(INFO) << "Source: " << source_path.value();
+    fake_media_source->SetSourceFile(source_path, override_fps);
+  }
 
   // CastTransportSender initialization.
-  scoped_ptr<media::cast::transport::CastTransportSender> transport_sender =
-      media::cast::transport::CastTransportSender::Create(
+  scoped_ptr<media::cast::CastTransportSender> transport_sender =
+      media::cast::CastTransportSender::Create(
           NULL,  // net log.
           cast_environment->Clock(),
           remote_endpoint,
@@ -942,7 +322,7 @@ int main(int argc, char** argv) {
   scoped_ptr<media::cast::CastSender> cast_sender =
       media::cast::CastSender::Create(cast_environment, transport_sender.get());
   cast_sender->InitializeVideo(
-      send_process->get_video_config(),
+      fake_media_source->get_video_config(),
       base::Bind(&InitializationResult),
       media::cast::CreateDefaultVideoEncodeAcceleratorCallback(),
       media::cast::CreateDefaultVideoEncodeMemoryCallback());
@@ -1014,8 +394,8 @@ int main(int argc, char** argv) {
                  base::Passed(&offset_estimator)),
       base::TimeDelta::FromSeconds(logging_duration_seconds));
 
-  send_process->Start(cast_sender->audio_frame_input(),
-                      cast_sender->video_frame_input());
+  fake_media_source->Start(cast_sender->audio_frame_input(),
+                           cast_sender->video_frame_input());
 
   io_message_loop.Run();
   return 0;
diff --git a/media/cast/test/simulator.cc b/media/cast/test/simulator.cc
new file mode 100644
index 0000000000..43b3a02dc6
--- /dev/null
+++ b/media/cast/test/simulator.cc
@@ -0,0 +1,445 @@
+// 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.
+
+// Simulate end to end streaming.
+//
+// Input:
+// --source=
+//   WebM used as the source of video and audio frames.
+// --output=
+//   File path to writing out the raw event log of the simulation session.
+// --sim-id=
+//   Unique simulation ID.
+//
+// Output:
+// - Raw event log of the simulation session tagged with the unique test ID,
+//   written out to the specified file path.
+
+#include "base/at_exit.h"
+#include "base/base_paths.h"
+#include "base/command_line.h"
+#include "base/file_util.h"
+#include "base/files/file_path.h"
+#include "base/files/memory_mapped_file.h"
+#include "base/files/scoped_file.h"
+#include "base/json/json_writer.h"
+#include "base/logging.h"
+#include "base/path_service.h"
+#include "base/test/simple_test_tick_clock.h"
+#include "base/thread_task_runner_handle.h"
+#include "base/time/tick_clock.h"
+#include "base/values.h"
+#include "media/base/audio_bus.h"
+#include "media/base/media.h"
+#include "media/base/video_frame.h"
+#include "media/cast/cast_config.h"
+#include "media/cast/cast_environment.h"
+#include "media/cast/cast_receiver.h"
+#include "media/cast/cast_sender.h"
+#include "media/cast/logging/encoding_event_subscriber.h"
+#include "media/cast/logging/log_serializer.h"
+#include "media/cast/logging/logging_defines.h"
+#include "media/cast/logging/proto/raw_events.pb.h"
+#include "media/cast/logging/raw_event_subscriber_bundle.h"
+#include "media/cast/logging/simple_event_subscriber.h"
+#include "media/cast/net/cast_transport_config.h"
+#include "media/cast/net/cast_transport_defines.h"
+#include "media/cast/net/cast_transport_sender.h"
+#include "media/cast/net/cast_transport_sender_impl.h"
+#include "media/cast/test/fake_media_source.h"
+#include "media/cast/test/fake_single_thread_task_runner.h"
+#include "media/cast/test/loopback_transport.h"
+#include "media/cast/test/proto/network_simulation_model.pb.h"
+#include "media/cast/test/skewed_tick_clock.h"
+#include "media/cast/test/utility/audio_utility.h"
+#include "media/cast/test/utility/default_config.h"
+#include "media/cast/test/utility/test_util.h"
+#include "media/cast/test/utility/udp_proxy.h"
+#include "media/cast/test/utility/video_utility.h"
+
+using media::cast::proto::IPPModel;
+using media::cast::proto::NetworkSimulationModel;
+using media::cast::proto::NetworkSimulationModelType;
+
+namespace media {
+namespace cast {
+namespace {
+const int kTargetDelay = 300;
+const char kSourcePath[] = "source";
+const char kModelPath[] = "model";
+const char kOutputPath[] = "output";
+const char kSimulationId[] = "sim-id";
+const char kLibDir[] = "lib-dir";
+
+void UpdateCastTransportStatus(CastTransportStatus status) {
+  LOG(INFO) << "Cast transport status: " << status;
+}
+
+void AudioInitializationStatus(CastInitializationStatus status) {
+  LOG(INFO) << "Audio status: " << status;
+}
+
+void VideoInitializationStatus(CastInitializationStatus status) {
+  LOG(INFO) << "Video status: " << status;
+}
+
+void LogTransportEvents(const scoped_refptr<CastEnvironment>& env,
+                        const std::vector<PacketEvent>& packet_events) {
+  for (std::vector<media::cast::PacketEvent>::const_iterator it =
+           packet_events.begin();
+       it != packet_events.end();
+       ++it) {
+    env->Logging()->InsertPacketEvent(it->timestamp,
+                                      it->type,
+                                      it->media_type,
+                                      it->rtp_timestamp,
+                                      it->frame_id,
+                                      it->packet_id,
+                                      it->max_packet_id,
+                                      it->size);
+  }
+}
+
+void GotVideoFrame(
+    int* counter,
+    CastReceiver* cast_receiver,
+    const scoped_refptr<media::VideoFrame>& video_frame,
+    const base::TimeTicks& render_time,
+    bool continuous) {
+  ++*counter;
+  cast_receiver->RequestDecodedVideoFrame(
+      base::Bind(&GotVideoFrame, counter, cast_receiver));
+}
+
+void GotAudioFrame(
+    int* counter,
+    CastReceiver* cast_receiver,
+    scoped_ptr<AudioBus> audio_bus,
+    const base::TimeTicks& playout_time,
+    bool is_continuous) {
+  ++*counter;
+  cast_receiver->RequestDecodedAudioFrame(
+      base::Bind(&GotAudioFrame, counter, cast_receiver));
+}
+
+void AppendLog(EncodingEventSubscriber* subscriber,
+               const std::string& extra_data,
+               const base::FilePath& output_path) {
+  media::cast::proto::LogMetadata metadata;
+  metadata.set_extra_data(extra_data);
+
+  media::cast::FrameEventList frame_events;
+  media::cast::PacketEventList packet_events;
+  subscriber->GetEventsAndReset(
+      &metadata, &frame_events, &packet_events);
+  media::cast::proto::GeneralDescription* gen_desc =
+      metadata.mutable_general_description();
+  gen_desc->set_product("Cast Simulator");
+  gen_desc->set_product_version("0.1");
+
+  scoped_ptr<char[]> serialized_log(new char[media::cast::kMaxSerializedBytes]);
+  int output_bytes;
+  bool success = media::cast::SerializeEvents(metadata,
+                                              frame_events,
+                                              packet_events,
+                                              true,
+                                              media::cast::kMaxSerializedBytes,
+                                              serialized_log.get(),
+                                              &output_bytes);
+
+  if (!success) {
+    LOG(ERROR) << "Failed to serialize log.";
+    return;
+  }
+
+  if (AppendToFile(output_path, serialized_log.get(), output_bytes) == -1) {
+    LOG(ERROR) << "Failed to append to log.";
+  }
+}
+
+// Run simulation once.
+//
+// |output_path| is the path to write serialized log.
+// |extra_data| is extra tagging information to write to log.
+void RunSimulation(const base::FilePath& source_path,
+                   const base::FilePath& output_path,
+                   const std::string& extra_data,
+                   const NetworkSimulationModel& model) {
+  // Fake clock. Make sure start time is non zero.
+  base::SimpleTestTickClock testing_clock;
+  testing_clock.Advance(base::TimeDelta::FromSeconds(1));
+
+  // Task runner.
+  scoped_refptr<test::FakeSingleThreadTaskRunner> task_runner =
+      new test::FakeSingleThreadTaskRunner(&testing_clock);
+  base::ThreadTaskRunnerHandle task_runner_handle(task_runner);
+
+  // CastEnvironments.
+  scoped_refptr<CastEnvironment> sender_env =
+      new CastEnvironment(
+          scoped_ptr<base::TickClock>(
+              new test::SkewedTickClock(&testing_clock)).Pass(),
+          task_runner,
+          task_runner,
+          task_runner);
+  scoped_refptr<CastEnvironment> receiver_env =
+      new CastEnvironment(
+          scoped_ptr<base::TickClock>(
+              new test::SkewedTickClock(&testing_clock)).Pass(),
+          task_runner,
+          task_runner,
+          task_runner);
+
+  // Event subscriber. Store at most 1 hour of events.
+  EncodingEventSubscriber audio_event_subscriber(AUDIO_EVENT,
+                                                 100 * 60 * 60);
+  EncodingEventSubscriber video_event_subscriber(VIDEO_EVENT,
+                                                 30 * 60 * 60);
+  sender_env->Logging()->AddRawEventSubscriber(&audio_event_subscriber);
+  sender_env->Logging()->AddRawEventSubscriber(&video_event_subscriber);
+
+  // Audio sender config.
+  AudioSenderConfig audio_sender_config = GetDefaultAudioSenderConfig();
+  audio_sender_config.target_playout_delay =
+      base::TimeDelta::FromMilliseconds(kTargetDelay);
+
+  // Audio receiver config.
+  FrameReceiverConfig audio_receiver_config =
+      GetDefaultAudioReceiverConfig();
+  audio_receiver_config.rtp_max_delay_ms =
+      audio_sender_config.target_playout_delay.InMilliseconds();
+
+  // Video sender config.
+  VideoSenderConfig video_sender_config = GetDefaultVideoSenderConfig();
+  video_sender_config.max_bitrate = 4000000;
+  video_sender_config.min_bitrate = 2000000;
+  video_sender_config.start_bitrate = 4000000;
+  video_sender_config.target_playout_delay =
+      base::TimeDelta::FromMilliseconds(kTargetDelay);
+
+  // Video receiver config.
+  FrameReceiverConfig video_receiver_config =
+      GetDefaultVideoReceiverConfig();
+  video_receiver_config.rtp_max_delay_ms =
+      video_sender_config.target_playout_delay.InMilliseconds();
+
+  // Loopback transport.
+  LoopBackTransport receiver_to_sender(receiver_env);
+  LoopBackTransport sender_to_receiver(sender_env);
+
+  // Cast receiver.
+  scoped_ptr<CastReceiver> cast_receiver(
+      CastReceiver::Create(receiver_env,
+                           audio_receiver_config,
+                           video_receiver_config,
+                           &receiver_to_sender));
+
+  // Cast sender and transport sender.
+  scoped_ptr<CastTransportSender> transport_sender(
+      new CastTransportSenderImpl(
+          NULL,
+          &testing_clock,
+          net::IPEndPoint(),
+          base::Bind(&UpdateCastTransportStatus),
+          base::Bind(&LogTransportEvents, sender_env),
+          base::TimeDelta::FromSeconds(1),
+          task_runner,
+          &sender_to_receiver));
+  scoped_ptr<CastSender> cast_sender(
+      CastSender::Create(sender_env, transport_sender.get()));
+
+  // Build packet pipe.
+  if (model.type() != media::cast::proto::INTERRUPTED_POISSON_PROCESS) {
+    LOG(ERROR) << "Unknown model type " << model.type() << ".";
+    return;
+  }
+
+  const IPPModel& ipp_model = model.ipp();
+
+  std::vector<double> average_rates(ipp_model.average_rate_size());
+  std::copy(ipp_model.average_rate().begin(), ipp_model.average_rate().end(),
+      average_rates.begin());
+  test::InterruptedPoissonProcess ipp(average_rates,
+      ipp_model.coef_burstiness(), ipp_model.coef_variance(), 0);
+
+  // Connect sender to receiver. This initializes the pipe.
+  receiver_to_sender.Initialize(
+      ipp.NewBuffer(128 * 1024), cast_sender->packet_receiver(), task_runner,
+      &testing_clock);
+  sender_to_receiver.Initialize(
+      ipp.NewBuffer(128 * 1024), cast_receiver->packet_receiver(), task_runner,
+      &testing_clock);
+
+  // Start receiver.
+  int audio_frame_count = 0;
+  int video_frame_count = 0;
+  cast_receiver->RequestDecodedVideoFrame(
+      base::Bind(&GotVideoFrame, &video_frame_count, cast_receiver.get()));
+  cast_receiver->RequestDecodedAudioFrame(
+      base::Bind(&GotAudioFrame, &audio_frame_count, cast_receiver.get()));
+
+  FakeMediaSource media_source(task_runner,
+                               &testing_clock,
+                               video_sender_config);
+
+  // Initializing audio and video senders.
+  cast_sender->InitializeAudio(audio_sender_config,
+                               base::Bind(&AudioInitializationStatus));
+  cast_sender->InitializeVideo(media_source.get_video_config(),
+                               base::Bind(&VideoInitializationStatus),
+                               CreateDefaultVideoEncodeAcceleratorCallback(),
+                               CreateDefaultVideoEncodeMemoryCallback());
+
+  // Start sending.
+  if (!source_path.empty()) {
+    // 0 means using the FPS from the file.
+    media_source.SetSourceFile(source_path, 0);
+  }
+  media_source.Start(cast_sender->audio_frame_input(),
+                     cast_sender->video_frame_input());
+
+  // Run for 3 minutes.
+  base::TimeDelta elapsed_time;
+  while (elapsed_time.InMinutes() < 3) {
+    // Each step is 100us.
+    base::TimeDelta step = base::TimeDelta::FromMicroseconds(100);
+    task_runner->Sleep(step);
+    elapsed_time += step;
+  }
+
+  LOG(INFO) << "Audio frame count: " << audio_frame_count;
+  LOG(INFO) << "Video frame count: " << video_frame_count;
+  LOG(INFO) << "Writing log: " << output_path.value();
+
+  // Truncate file and then write serialized log.
+  {
+    base::ScopedFILE file(base::OpenFile(output_path, "wb"));
+    if (!file.get()) {
+      LOG(INFO) << "Cannot write to log.";
+      return;
+    }
+  }
+  AppendLog(&video_event_subscriber, extra_data, output_path);
+  AppendLog(&audio_event_subscriber, extra_data, output_path);
+}
+
+NetworkSimulationModel DefaultModel() {
+  NetworkSimulationModel model;
+  model.set_type(cast::proto::INTERRUPTED_POISSON_PROCESS);
+  IPPModel* ipp = model.mutable_ipp();
+  ipp->set_coef_burstiness(0.609);
+  ipp->set_coef_variance(4.1);
+
+  ipp->add_average_rate(0.609);
+  ipp->add_average_rate(0.495);
+  ipp->add_average_rate(0.561);
+  ipp->add_average_rate(0.458);
+  ipp->add_average_rate(0.538);
+  ipp->add_average_rate(0.513);
+  ipp->add_average_rate(0.585);
+  ipp->add_average_rate(0.592);
+  ipp->add_average_rate(0.658);
+  ipp->add_average_rate(0.556);
+  ipp->add_average_rate(0.371);
+  ipp->add_average_rate(0.595);
+  ipp->add_average_rate(0.490);
+  ipp->add_average_rate(0.980);
+  ipp->add_average_rate(0.781);
+  ipp->add_average_rate(0.463);
+
+  return model;
+}
+
+bool IsModelValid(const NetworkSimulationModel& model) {
+  if (!model.has_type())
+    return false;
+  NetworkSimulationModelType type = model.type();
+  if (type == media::cast::proto::INTERRUPTED_POISSON_PROCESS) {
+    if (!model.has_ipp())
+      return false;
+    const IPPModel& ipp = model.ipp();
+    if (ipp.coef_burstiness() <= 0.0 || ipp.coef_variance() <= 0.0)
+      return false;
+    if (ipp.average_rate_size() == 0)
+      return false;
+    for (int i = 0; i < ipp.average_rate_size(); i++) {
+      if (ipp.average_rate(i) <= 0.0)
+        return false;
+    }
+  }
+
+  return true;
+}
+
+NetworkSimulationModel LoadModel(const base::FilePath& model_path) {
+  if (model_path.empty()) {
+    LOG(ERROR) << "Model path not set.";
+    return DefaultModel();
+  }
+  std::string model_str;
+  if (!base::ReadFileToString(model_path, &model_str)) {
+    LOG(ERROR) << "Failed to read model file.";
+    return DefaultModel();
+  }
+
+  NetworkSimulationModel model;
+  if (!model.ParseFromString(model_str)) {
+    LOG(ERROR) << "Failed to parse model.";
+    return DefaultModel();
+  }
+  if (!IsModelValid(model)) {
+    LOG(ERROR) << "Invalid model.";
+    return DefaultModel();
+  }
+
+  return model;
+}
+
+}  // namespace
+}  // namespace cast
+}  // namespace media
+
+int main(int argc, char** argv) {
+  base::AtExitManager at_exit;
+  CommandLine::Init(argc, argv);
+  InitLogging(logging::LoggingSettings());
+
+  const CommandLine* cmd = CommandLine::ForCurrentProcess();
+  base::FilePath media_path = cmd->GetSwitchValuePath(media::cast::kLibDir);
+  if (media_path.empty()) {
+    if (!PathService::Get(base::DIR_MODULE, &media_path)) {
+      LOG(ERROR) << "Failed to load FFmpeg.";
+      return 1;
+    }
+  }
+
+  if (!media::InitializeMediaLibrary(media_path)) {
+    LOG(ERROR) << "Failed to initialize FFmpeg.";
+    return 1;
+  }
+
+  base::FilePath source_path = cmd->GetSwitchValuePath(
+      media::cast::kSourcePath);
+  base::FilePath output_path = cmd->GetSwitchValuePath(
+      media::cast::kOutputPath);
+  if (output_path.empty()) {
+    base::GetTempDir(&output_path);
+    output_path = output_path.AppendASCII("sim-events.gz");
+  }
+  std::string sim_id = cmd->GetSwitchValueASCII(media::cast::kSimulationId);
+
+  NetworkSimulationModel model = media::cast::LoadModel(
+      cmd->GetSwitchValuePath(media::cast::kModelPath));
+
+  base::DictionaryValue values;
+  values.SetBoolean("sim", true);
+  values.SetString("sim-id", sim_id);
+
+  std::string extra_data;
+  base::JSONWriter::Write(&values, &extra_data);
+
+  // Run.
+  media::cast::RunSimulation(source_path, output_path, extra_data, model);
+  return 0;
+}
diff --git a/media/cast/test/utility/default_config.cc b/media/cast/test/utility/default_config.cc
index b5de5ed690..6743146b70 100644
--- a/media/cast/test/utility/default_config.cc
+++ b/media/cast/test/utility/default_config.cc
@@ -5,7 +5,7 @@
 #include "media/cast/test/utility/default_config.h"
 
 #include "base/bind.h"
-#include "media/cast/transport/cast_transport_config.h"
+#include "media/cast/net/cast_transport_config.h"
 
 namespace {
 
@@ -35,7 +35,7 @@ FrameReceiverConfig GetDefaultAudioReceiverConfig() {
   config.frequency = 48000;
   config.channels = 2;
   config.max_frame_rate = 100;  // 10ms of signal per frame
-  config.codec.audio = media::cast::transport::kOpus;
+  config.codec = media::cast::CODEC_AUDIO_OPUS;
   return config;
 }
 
@@ -49,7 +49,44 @@ FrameReceiverConfig GetDefaultVideoReceiverConfig() {
   config.frequency = kVideoFrequency;
   config.channels = 1;
   config.max_frame_rate = kDefaultMaxFrameRate;
-  config.codec.video = media::cast::transport::kVp8;
+  config.codec = media::cast::CODEC_VIDEO_VP8;
+  return config;
+}
+
+AudioSenderConfig GetDefaultAudioSenderConfig() {
+  FrameReceiverConfig recv_config = GetDefaultAudioReceiverConfig();
+  AudioSenderConfig config;
+  config.ssrc = recv_config.incoming_ssrc;
+  config.incoming_feedback_ssrc = recv_config.feedback_ssrc;
+  config.rtp_payload_type = recv_config.rtp_payload_type;
+  config.use_external_encoder = false;
+  config.frequency = recv_config.frequency;
+  config.channels = recv_config.channels;
+  config.bitrate = kDefaultAudioEncoderBitrate;
+  config.codec = recv_config.codec;
+  config.target_playout_delay =
+      base::TimeDelta::FromMilliseconds(kDefaultRtpMaxDelayMs);
+  return config;
+}
+
+VideoSenderConfig GetDefaultVideoSenderConfig() {
+  FrameReceiverConfig recv_config = GetDefaultVideoReceiverConfig();
+  VideoSenderConfig config;
+  config.ssrc = recv_config.incoming_ssrc;
+  config.incoming_feedback_ssrc = recv_config.feedback_ssrc;
+  config.rtp_payload_type = recv_config.rtp_payload_type;
+  config.use_external_encoder = false;
+  config.width = 1280;
+  config.height = 720;
+  config.max_bitrate = 4000000;
+  config.min_bitrate = 2000000;
+  config.start_bitrate = 4000000;
+  config.max_frame_rate = recv_config.max_frame_rate;
+  config.max_number_of_video_buffers_used = 1;
+  config.codec = recv_config.codec;
+  config.number_of_encode_threads = 2;
+  config.target_playout_delay =
+      base::TimeDelta::FromMilliseconds(kDefaultRtpMaxDelayMs);
   return config;
 }
 
diff --git a/media/cast/test/utility/default_config.h b/media/cast/test/utility/default_config.h
index eaa3c96415..2cc52a7a72 100644
--- a/media/cast/test/utility/default_config.h
+++ b/media/cast/test/utility/default_config.h
@@ -20,6 +20,15 @@ FrameReceiverConfig GetDefaultAudioReceiverConfig();
 // name.
 FrameReceiverConfig GetDefaultVideoReceiverConfig();
 
+// Returns a AudioSenderConfig initialized to default values. This means
+// 48 kHz, 2-channel Opus-coded audio. Default values for SSRCs and payload
+// type.
+AudioSenderConfig GetDefaultAudioSenderConfig();
+
+// Returns a VideoSenderConfig initialized to default values. This means
+// 30 Hz VP8 coded code. Default values for SSRCs and payload type.
+VideoSenderConfig GetDefaultVideoSenderConfig();
+
 // Returns a callback that does nothing.
 CreateVideoEncodeAcceleratorCallback
     CreateDefaultVideoEncodeAcceleratorCallback();
diff --git a/media/cast/test/utility/in_process_receiver.cc b/media/cast/test/utility/in_process_receiver.cc
index cfcc1fcb46..6c739ecdb6 100644
--- a/media/cast/test/utility/in_process_receiver.cc
+++ b/media/cast/test/utility/in_process_receiver.cc
@@ -11,11 +11,11 @@
 #include "media/cast/cast_config.h"
 #include "media/cast/cast_environment.h"
 #include "media/cast/cast_receiver.h"
-#include "media/cast/transport/cast_transport_config.h"
-#include "media/cast/transport/transport/udp_transport.h"
+#include "media/cast/net/cast_transport_config.h"
+#include "media/cast/net/udp_transport.h"
 
-using media::cast::transport::CastTransportStatus;
-using media::cast::transport::UdpTransport;
+using media::cast::CastTransportStatus;
+using media::cast::UdpTransport;
 
 namespace media {
 namespace cast {
@@ -67,7 +67,7 @@ void InProcessReceiver::StopOnMainThread(base::WaitableEvent* event) {
 }
 
 void InProcessReceiver::UpdateCastTransportStatus(CastTransportStatus status) {
-  LOG_IF(ERROR, status == media::cast::transport::TRANSPORT_SOCKET_ERROR)
+  LOG_IF(ERROR, status == media::cast::TRANSPORT_SOCKET_ERROR)
       << "Transport socket error occurred.  InProcessReceiver is likely dead.";
   VLOG(1) << "CastTransportStatus is now " << status;
 }
diff --git a/media/cast/test/utility/in_process_receiver.h b/media/cast/test/utility/in_process_receiver.h
index cf25da9cee..454dc937ae 100644
--- a/media/cast/test/utility/in_process_receiver.h
+++ b/media/cast/test/utility/in_process_receiver.h
@@ -10,7 +10,8 @@
 #include "base/memory/weak_ptr.h"
 #include "media/base/audio_bus.h"
 #include "media/cast/cast_config.h"
-#include "media/cast/transport/cast_transport_config.h"
+#include "media/cast/net/cast_transport_config.h"
+#include "net/base/ip_endpoint.h"
 
 namespace base {
 class TimeTicks;
@@ -29,10 +30,7 @@ namespace cast {
 
 class CastEnvironment;
 class CastReceiver;
-
-namespace transport {
 class UdpTransport;
-}  // namespace transport
 
 // Common base functionality for an in-process Cast receiver.  This is meant to
 // be subclassed with the OnAudioFrame() and OnVideoFrame() methods implemented,
@@ -85,7 +83,7 @@ class InProcessReceiver {
 
   // Callback for the transport to notify of status changes.  A default
   // implementation is provided here that simply logs socket errors.
-  virtual void UpdateCastTransportStatus(transport::CastTransportStatus status);
+  virtual void UpdateCastTransportStatus(CastTransportStatus status);
 
  private:
   friend class base::RefCountedThreadSafe<InProcessReceiver>;
@@ -108,7 +106,7 @@ class InProcessReceiver {
   const FrameReceiverConfig audio_config_;
   const FrameReceiverConfig video_config_;
 
-  scoped_ptr<transport::UdpTransport> transport_;
+  scoped_ptr<UdpTransport> transport_;
   scoped_ptr<CastReceiver> cast_receiver_;
 
   // NOTE: Weak pointers must be invalidated before all other member variables.
diff --git a/media/cast/test/utility/udp_proxy.cc b/media/cast/test/utility/udp_proxy.cc
index 9fc3b4a44d..4714b7ed67 100644
--- a/media/cast/test/utility/udp_proxy.cc
+++ b/media/cast/test/utility/udp_proxy.cc
@@ -2,12 +2,13 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
+#include <math.h>
 #include <stdlib.h>
+#include <vector>
 
 #include "media/cast/test/utility/udp_proxy.h"
 
 #include "base/logging.h"
-#include "base/memory/linked_ptr.h"
 #include "base/rand_util.h"
 #include "base/synchronization/waitable_event.h"
 #include "base/threading/thread.h"
@@ -55,10 +56,10 @@ class Buffer : public PacketPipe {
     CHECK_GT(max_megabits_per_second, 0);
   }
 
-  virtual void Send(scoped_ptr<transport::Packet> packet) OVERRIDE {
+  virtual void Send(scoped_ptr<Packet> packet) OVERRIDE {
     if (packet->size() + buffer_size_ <= max_buffer_size_) {
       buffer_size_ += packet->size();
-      buffer_.push_back(linked_ptr<transport::Packet>(packet.release()));
+      buffer_.push_back(linked_ptr<Packet>(packet.release()));
       if (buffer_.size() == 1) {
         Schedule();
       }
@@ -78,7 +79,7 @@ class Buffer : public PacketPipe {
 
   void ProcessBuffer() {
     CHECK(!buffer_.empty());
-    scoped_ptr<transport::Packet> packet(buffer_.front().release());
+    scoped_ptr<Packet> packet(buffer_.front().release());
     buffer_size_ -= packet->size();
     buffer_.pop_front();
     pipe_->Send(packet.Pass());
@@ -87,7 +88,7 @@ class Buffer : public PacketPipe {
     }
   }
 
-  std::deque<linked_ptr<transport::Packet> > buffer_;
+  std::deque<linked_ptr<Packet> > buffer_;
   size_t buffer_size_;
   size_t max_buffer_size_;
   double max_megabits_per_second_;  // megabits per second
@@ -103,7 +104,7 @@ class RandomDrop : public PacketPipe {
   RandomDrop(double drop_fraction)
       : drop_fraction_(static_cast<int>(drop_fraction * RAND_MAX)) {}
 
-  virtual void Send(scoped_ptr<transport::Packet> packet) OVERRIDE {
+  virtual void Send(scoped_ptr<Packet> packet) OVERRIDE {
     if (rand() > drop_fraction_) {
       pipe_->Send(packet.Pass());
     }
@@ -122,7 +123,7 @@ class SimpleDelayBase : public PacketPipe {
   SimpleDelayBase() : weak_factory_(this) {}
   virtual ~SimpleDelayBase() {}
 
-  virtual void Send(scoped_ptr<transport::Packet> packet) OVERRIDE {
+  virtual void Send(scoped_ptr<Packet> packet) OVERRIDE {
     double seconds = GetDelay();
     task_runner_->PostDelayedTask(
         FROM_HERE,
@@ -135,7 +136,7 @@ class SimpleDelayBase : public PacketPipe {
   virtual double GetDelay() = 0;
 
  private:
-  virtual void SendInternal(scoped_ptr<transport::Packet> packet) {
+  virtual void SendInternal(scoped_ptr<Packet> packet) {
     pipe_->Send(packet.Pass());
   }
 
@@ -184,8 +185,8 @@ class RandomSortedDelay : public PacketPipe {
         seconds_between_extra_delay_(seconds_between_extra_delay),
         weak_factory_(this) {}
 
-  virtual void Send(scoped_ptr<transport::Packet> packet) OVERRIDE {
-    buffer_.push_back(linked_ptr<transport::Packet>(packet.release()));
+  virtual void Send(scoped_ptr<Packet> packet) OVERRIDE {
+    buffer_.push_back(linked_ptr<Packet>(packet.release()));
     if (buffer_.size() == 1) {
       Schedule();
     }
@@ -238,7 +239,7 @@ class RandomSortedDelay : public PacketPipe {
 
   void ProcessBuffer() {
     CHECK(!buffer_.empty());
-    scoped_ptr<transport::Packet> packet(buffer_.front().release());
+    scoped_ptr<Packet> packet(buffer_.front().release());
     pipe_->Send(packet.Pass());
     buffer_.pop_front();
     if (!buffer_.empty()) {
@@ -247,7 +248,7 @@ class RandomSortedDelay : public PacketPipe {
   }
 
   base::TimeTicks block_until_;
-  std::deque<linked_ptr<transport::Packet> > buffer_;
+  std::deque<linked_ptr<Packet> > buffer_;
   double random_delay_;
   double extra_delay_;
   double seconds_between_extra_delay_;
@@ -279,7 +280,7 @@ class NetworkGlitchPipe : public PacketPipe {
     Flip();
   }
 
-  virtual void Send(scoped_ptr<transport::Packet> packet) OVERRIDE {
+  virtual void Send(scoped_ptr<Packet> packet) OVERRIDE {
     if (works_) {
       pipe_->Send(packet.Pass());
     }
@@ -310,13 +311,212 @@ scoped_ptr<PacketPipe> NewNetworkGlitchPipe(double average_work_time,
       .Pass();
 }
 
+
+// Internal buffer object for a client of the IPP model.
+class InterruptedPoissonProcess::InternalBuffer : public PacketPipe {
+ public:
+  InternalBuffer(base::WeakPtr<InterruptedPoissonProcess> ipp,
+                 size_t size)
+      : ipp_(ipp),
+        stored_size_(0),
+        stored_limit_(size),
+        clock_(NULL),
+        weak_factory_(this) {
+  }
+
+  virtual void Send(scoped_ptr<Packet> packet) OVERRIDE {
+    // Drop if buffer is full.
+    if (stored_size_ >= stored_limit_)
+      return;
+    stored_size_ += packet->size();
+    buffer_.push_back(linked_ptr<Packet>(packet.release()));
+    buffer_time_.push_back(clock_->NowTicks());
+    DCHECK(buffer_.size() == buffer_time_.size());
+  }
+
+  virtual void InitOnIOThread(
+      const scoped_refptr<base::SingleThreadTaskRunner>& task_runner,
+      base::TickClock* clock) OVERRIDE {
+    clock_ = clock;
+    if (ipp_)
+      ipp_->InitOnIOThread(task_runner, clock);
+    PacketPipe::InitOnIOThread(task_runner, clock);
+  }
+
+  void SendOnePacket() {
+    scoped_ptr<Packet> packet(buffer_.front().release());
+    stored_size_ -= packet->size();
+    buffer_.pop_front();
+    buffer_time_.pop_front();
+    pipe_->Send(packet.Pass());
+    DCHECK(buffer_.size() == buffer_time_.size());
+  }
+
+  bool Empty() const {
+    return buffer_.empty();
+  }
+
+  base::TimeTicks FirstPacketTime() const {
+    DCHECK(!buffer_time_.empty());
+    return buffer_time_.front();
+  }
+
+  base::WeakPtr<InternalBuffer> GetWeakPtr() {
+    return weak_factory_.GetWeakPtr();
+
+  }
+
+ private:
+  const base::WeakPtr<InterruptedPoissonProcess> ipp_;
+  size_t stored_size_;
+  const size_t stored_limit_;
+  std::deque<linked_ptr<Packet> > buffer_;
+  std::deque<base::TimeTicks> buffer_time_;
+  base::TickClock* clock_;
+  base::WeakPtrFactory<InternalBuffer> weak_factory_;
+
+  DISALLOW_COPY_AND_ASSIGN(InternalBuffer);
+};
+
+InterruptedPoissonProcess::InterruptedPoissonProcess(
+    const std::vector<double>& average_rates,
+    double coef_burstiness,
+    double coef_variance,
+    uint32 rand_seed)
+    : clock_(NULL),
+      average_rates_(average_rates),
+      coef_burstiness_(coef_burstiness),
+      coef_variance_(coef_variance),
+      rate_index_(0),
+      on_state_(true),
+      weak_factory_(this) {
+  mt_rand_.init_genrand(rand_seed);
+  DCHECK(!average_rates.empty());
+  ComputeRates();
+}
+
+InterruptedPoissonProcess::~InterruptedPoissonProcess() {
+}
+
+void InterruptedPoissonProcess::InitOnIOThread(
+    const scoped_refptr<base::SingleThreadTaskRunner>& task_runner,
+    base::TickClock* clock) {
+  // Already initialized and started.
+  if (task_runner_ &&  clock_)
+    return;
+  task_runner_ = task_runner;
+  clock_ = clock;
+  UpdateRates();
+  SwitchOn();
+  SendPacket();
+}
+
+scoped_ptr<PacketPipe> InterruptedPoissonProcess::NewBuffer(size_t size) {
+  scoped_ptr<InternalBuffer> buffer(
+      new InternalBuffer(weak_factory_.GetWeakPtr(), size));
+  send_buffers_.push_back(buffer->GetWeakPtr());
+  return buffer.PassAs<PacketPipe>();
+}
+
+base::TimeDelta InterruptedPoissonProcess::NextEvent(double rate) {
+  // Rate is per milliseconds.
+  // The time until next event is exponentially distributed to the
+  // inverse of |rate|.
+  return base::TimeDelta::FromMillisecondsD(
+      fabs(-log(1.0 - RandDouble()) / rate));
+}
+
+double InterruptedPoissonProcess::RandDouble() {
+  // Generate a 64-bits random number from MT19937 and then convert
+  // it to double.
+  uint64 rand = mt_rand_.genrand_int32();
+  rand <<= 32;
+  rand |= mt_rand_.genrand_int32();
+  return base::BitsToOpenEndedUnitInterval(rand);
+}
+
+void InterruptedPoissonProcess::ComputeRates() {
+  double avg_rate = average_rates_[rate_index_];
+
+  send_rate_ = avg_rate / coef_burstiness_;
+  switch_off_rate_ =
+      2 * avg_rate * (1 - coef_burstiness_) * (1 - coef_burstiness_) /
+      coef_burstiness_ / (coef_variance_ - 1);
+  switch_on_rate_ =
+      2 * avg_rate * (1 - coef_burstiness_) / (coef_variance_ - 1);
+}
+
+void InterruptedPoissonProcess::UpdateRates() {
+  ComputeRates();
+
+  // Rates are updated once per second.
+  rate_index_ = (rate_index_ + 1) % average_rates_.size();
+  task_runner_->PostDelayedTask(
+      FROM_HERE,
+      base::Bind(&InterruptedPoissonProcess::UpdateRates,
+                 weak_factory_.GetWeakPtr()),
+      base::TimeDelta::FromSeconds(1));
+}
+
+void InterruptedPoissonProcess::SwitchOff() {
+  on_state_ = false;
+  task_runner_->PostDelayedTask(
+      FROM_HERE,
+      base::Bind(&InterruptedPoissonProcess::SwitchOn,
+                 weak_factory_.GetWeakPtr()),
+      NextEvent(switch_on_rate_));
+}
+
+void InterruptedPoissonProcess::SwitchOn() {
+  on_state_ = true;
+  task_runner_->PostDelayedTask(
+      FROM_HERE,
+      base::Bind(&InterruptedPoissonProcess::SwitchOff,
+                 weak_factory_.GetWeakPtr()),
+      NextEvent(switch_off_rate_));
+}
+
+void InterruptedPoissonProcess::SendPacket() {
+  task_runner_->PostDelayedTask(
+      FROM_HERE,
+      base::Bind(&InterruptedPoissonProcess::SendPacket,
+                 weak_factory_.GetWeakPtr()),
+      NextEvent(send_rate_));
+
+  // If OFF then don't send.
+  if (!on_state_)
+    return;
+
+  // Find the earliest packet to send.
+  base::TimeTicks earliest_time;
+  for (size_t i = 0; i < send_buffers_.size(); ++i) {
+    if (!send_buffers_[i])
+      continue;
+    if (send_buffers_[i]->Empty())
+      continue;
+    if (earliest_time.is_null() ||
+        send_buffers_[i]->FirstPacketTime() < earliest_time)
+      earliest_time = send_buffers_[i]->FirstPacketTime();
+  }
+  for (size_t i = 0; i < send_buffers_.size(); ++i) {
+    if (!send_buffers_[i])
+      continue;
+    if (send_buffers_[i]->Empty())
+      continue;
+    if (send_buffers_[i]->FirstPacketTime() != earliest_time)
+      continue;
+    send_buffers_[i]->SendOnePacket();
+    break;
+  }
+}
+
 class UDPProxyImpl;
 
 class PacketSender : public PacketPipe {
  public:
   PacketSender(UDPProxyImpl* udp_proxy, const net::IPEndPoint* destination)
       : udp_proxy_(udp_proxy), destination_(destination) {}
-  virtual void Send(scoped_ptr<transport::Packet> packet) OVERRIDE;
+  virtual void Send(scoped_ptr<Packet> packet) OVERRIDE;
   virtual void AppendToPipe(scoped_ptr<PacketPipe> pipe) OVERRIDE {
     NOTREACHED();
   }
@@ -426,7 +626,7 @@ class UDPProxyImpl : public UDPProxy {
     proxy_thread_.Stop();
   }
 
-  void Send(scoped_ptr<transport::Packet> packet,
+  void Send(scoped_ptr<Packet> packet,
             const net::IPEndPoint& destination) {
     if (blocked_) {
       LOG(ERROR) << "Cannot write packet right now: blocked";
@@ -522,7 +722,7 @@ class UDPProxyImpl : public UDPProxy {
 
   void PollRead() {
     while (true) {
-      packet_.reset(new transport::Packet(kMaxPacketSize));
+      packet_.reset(new Packet(kMaxPacketSize));
       scoped_refptr<net::IOBuffer> recv_buf =
           new net::WrappedIOBuffer(reinterpret_cast<char*>(&packet_->front()));
       int len = socket_->RecvFrom(
@@ -539,7 +739,7 @@ class UDPProxyImpl : public UDPProxy {
   }
 
   void AllowWrite(scoped_refptr<net::IOBuffer> buf,
-                  scoped_ptr<transport::Packet> packet,
+                  scoped_ptr<Packet> packet,
                   int unused_len) {
     DCHECK(blocked_);
     blocked_ = false;
@@ -562,7 +762,7 @@ class UDPProxyImpl : public UDPProxy {
 
   // For receiving.
   net::IPEndPoint recv_address_;
-  scoped_ptr<transport::Packet> packet_;
+  scoped_ptr<Packet> packet_;
 
   // For sending.
   bool blocked_;
@@ -570,7 +770,7 @@ class UDPProxyImpl : public UDPProxy {
   base::WeakPtrFactory<UDPProxyImpl> weak_factory_;
 };
 
-void PacketSender::Send(scoped_ptr<transport::Packet> packet) {
+void PacketSender::Send(scoped_ptr<Packet> packet) {
   udp_proxy_->Send(packet.Pass(), *destination_);
 }
 
diff --git a/media/cast/test/utility/udp_proxy.h b/media/cast/test/utility/udp_proxy.h
index b102573a94..ea50a2c86a 100644
--- a/media/cast/test/utility/udp_proxy.h
+++ b/media/cast/test/utility/udp_proxy.h
@@ -8,11 +8,14 @@
 #include <vector>
 
 #include "base/basictypes.h"
+#include "base/memory/linked_ptr.h"
 #include "base/memory/ref_counted.h"
 #include "base/memory/scoped_ptr.h"
+#include "base/memory/weak_ptr.h"
 #include "base/single_thread_task_runner.h"
-#include "media/cast/transport/cast_transport_config.h"
+#include "media/cast/net/cast_transport_config.h"
 #include "net/base/ip_endpoint.h"
+#include "third_party/mt19937ar/mt19937ar.h"
 
 namespace net {
 class NetLog;
@@ -30,7 +33,7 @@ class PacketPipe {
  public:
   PacketPipe();
   virtual ~PacketPipe();
-  virtual void Send(scoped_ptr<transport::Packet> packet) = 0;
+  virtual void Send(scoped_ptr<Packet> packet) = 0;
   // Allows injection of fake test runner for testing.
   virtual void InitOnIOThread(
       const scoped_refptr<base::SingleThreadTaskRunner>& task_runner,
@@ -43,6 +46,63 @@ class PacketPipe {
   base::TickClock* clock_;
 };
 
+// Implements a Interrupted Poisson Process for packet delivery.
+// The process has 2 states: ON and OFF, the rate of switching between
+// these two states are defined.
+// When in ON state packets are sent according to a defined rate.
+// When in OFF state packets are not sent.
+// The rate above is the average rate of a poisson distribution.
+class InterruptedPoissonProcess {
+ public:
+  InterruptedPoissonProcess(
+      const std::vector<double>& average_rates,
+      double coef_burstiness,
+      double coef_variance,
+      uint32 rand_seed);
+  ~InterruptedPoissonProcess();
+
+  scoped_ptr<PacketPipe> NewBuffer(size_t size);
+
+ private:
+  class InternalBuffer;
+
+  // |task_runner| is the executor of the IO thread.
+  // |clock| is the system clock.
+  void InitOnIOThread(
+      const scoped_refptr<base::SingleThreadTaskRunner>& task_runner,
+      base::TickClock* clock);
+
+  base::TimeDelta NextEvent(double rate);
+  double RandDouble();
+  void ComputeRates();
+  void UpdateRates();
+  void SwitchOff();
+  void SwitchOn();
+  void SendPacket();
+
+  scoped_refptr<base::SingleThreadTaskRunner> task_runner_;
+  base::TickClock* clock_;
+  const std::vector<double> average_rates_;
+  const double coef_burstiness_;
+  const double coef_variance_;
+  int rate_index_;
+
+  // The following rates are per milliseconds.
+  double send_rate_;
+  double switch_off_rate_;
+  double switch_on_rate_;
+  bool on_state_;
+
+  std::vector<base::WeakPtr<InternalBuffer> > send_buffers_;
+
+  // Fast pseudo random number generator.
+  MersenneTwister mt_rand_;
+
+  base::WeakPtrFactory<InterruptedPoissonProcess> weak_factory_;
+
+  DISALLOW_COPY_AND_ASSIGN(InterruptedPoissonProcess);
+};
+
 // A UDPProxy will set up a UDP socket and bind to |local_port|.
 // Packets send to that port will be forwarded to |destination|.
 // Packets send from |destination| to |local_port| will be returned
diff --git a/media/cast/test/utility/udp_proxy_main.cc b/media/cast/test/utility/udp_proxy_main.cc
index 800f09dc70..9dd047abfd 100644
--- a/media/cast/test/utility/udp_proxy_main.cc
+++ b/media/cast/test/utility/udp_proxy_main.cc
@@ -68,7 +68,7 @@ ByteCounter out_pipe_output_counter;
 class ByteCounterPipe : public media::cast::test::PacketPipe {
  public:
   ByteCounterPipe(ByteCounter* counter) : counter_(counter) {}
-  virtual void Send(scoped_ptr<media::cast::transport::Packet> packet)
+  virtual void Send(scoped_ptr<media::cast::Packet> packet)
       OVERRIDE {
     counter_->Increment(packet->size());
     pipe_->Send(packet.Pass());
author	Ben Murdoch <benm@google.com>	2014-07-20 18:25:52 -0700
committer	Ben Murdoch <benm@google.com>	2014-07-20 18:25:52 -0700
commit	116680a4aac90f2aa7413d9095a592090648e557 (patch)
tree	f7c6fed0e63d6a2804243d4a31a752dca39fb076 /media/cast
parent	1f14a4515e04c9ffc9bac4dd1e2f68611626b800 (diff)
download	chromium_org-116680a4aac90f2aa7413d9095a592090648e557.tar.gz