Merge from Chromium at DEPS revision 278856

This commit was generated by merge_to_master.py.

Change-Id: If3807744d3e5d3ee84b897bd2d099a2b7ed2e7a3

24 files changed, 408 insertions, 115 deletions

diff --git a/media/cast/audio_sender/audio_encoder.cc b/media/cast/audio_sender/audio_encoder.cc
index f81ad26377..8860c7dd2d 100644
--- a/media/cast/audio_sender/audio_encoder.cc
+++ b/media/cast/audio_sender/audio_encoder.cc
@@ -9,14 +9,12 @@
 #include "base/bind.h"
 #include "base/bind_helpers.h"
 #include "base/location.h"
-#include "base/logging.h"
 #include "base/stl_util.h"
 #include "base/sys_byteorder.h"
 #include "base/time/time.h"
 #include "media/base/audio_bus.h"
 #include "media/cast/cast_defines.h"
 #include "media/cast/cast_environment.h"
-#include "media/cast/logging/logging_defines.h"
 #include "third_party/opus/src/include/opus.h"
 
 namespace media {
@@ -33,28 +31,6 @@ const int kFrameDurationMillis = 1000 / kFramesPerSecond;  // No remainder!
 // coming in too slow with respect to the capture timestamps.
 const int kUnderrunThresholdMillis = 3 * kFrameDurationMillis;
 
-void LogAudioFrameEncodedEvent(
-    const scoped_refptr<media::cast::CastEnvironment>& cast_environment,
-    base::TimeTicks event_time,
-    media::cast::RtpTimestamp rtp_timestamp,
-    uint32 frame_id,
-    size_t frame_size) {
-  if (!cast_environment->CurrentlyOn(CastEnvironment::MAIN)) {
-    cast_environment->PostTask(
-        CastEnvironment::MAIN,
-        FROM_HERE,
-        base::Bind(&LogAudioFrameEncodedEvent,
-                   cast_environment, event_time,
-                   rtp_timestamp, frame_id, frame_size));
-    return;
-  }
-  cast_environment->Logging()->InsertEncodedFrameEvent(
-      event_time, media::cast::FRAME_ENCODED, media::cast::AUDIO_EVENT,
-      rtp_timestamp, frame_id,
-      static_cast<int>(frame_size), /* key_frame - unused */ false,
-      /*target_bitrate - unused*/ 0);
-}
-
 }  // namespace
 
 
@@ -150,11 +126,6 @@ class AudioEncoder::ImplBase
       audio_frame->reference_time = frame_capture_time_;
 
       if (EncodeFromFilledBuffer(&audio_frame->data)) {
-        LogAudioFrameEncodedEvent(cast_environment_,
-                                  cast_environment_->Clock()->NowTicks(),
-                                  audio_frame->rtp_timestamp,
-                                  audio_frame->frame_id,
-                                  audio_frame->data.size());
         cast_environment_->PostTask(
             CastEnvironment::MAIN,
             FROM_HERE,
diff --git a/media/cast/audio_sender/audio_sender.cc b/media/cast/audio_sender/audio_sender.cc
index 27b42d0dc8..878f3456c8 100644
--- a/media/cast/audio_sender/audio_sender.cc
+++ b/media/cast/audio_sender/audio_sender.cc
@@ -8,7 +8,9 @@
 #include "base/logging.h"
 #include "base/message_loop/message_loop.h"
 #include "media/cast/audio_sender/audio_encoder.h"
-#include "media/cast/transport/cast_transport_defines.h"
+#include "media/cast/cast_defines.h"
+#include "media/cast/rtcp/rtcp_defines.h"
+#include "media/cast/transport/cast_transport_config.h"
 
 namespace media {
 namespace cast {
@@ -16,13 +18,24 @@ namespace cast {
 const int kNumAggressiveReportsSentAtStart = 100;
 const int kMinSchedulingDelayMs = 1;
 
-// TODO(mikhal): Reduce heap allocation when not needed.
+// TODO(miu): This should be specified in AudioSenderConfig, but currently it is
+// fixed to 100 FPS (i.e., 10 ms per frame), and AudioEncoder assumes this as
+// well.
+const int kAudioFrameRate = 100;
+
 AudioSender::AudioSender(scoped_refptr<CastEnvironment> cast_environment,
                          const AudioSenderConfig& audio_config,
                          transport::CastTransportSender* const transport_sender)
     : cast_environment_(cast_environment),
+      target_playout_delay_(base::TimeDelta::FromMilliseconds(
+          audio_config.rtp_config.max_delay_ms)),
       transport_sender_(transport_sender),
-      rtp_timestamp_helper_(audio_config.frequency),
+      max_unacked_frames_(
+          std::min(kMaxUnackedFrames,
+                   1 + static_cast<int>(target_playout_delay_ *
+                                        kAudioFrameRate /
+                                        base::TimeDelta::FromSeconds(1)))),
+      configured_encoder_bitrate_(audio_config.bitrate),
       rtcp_(cast_environment,
             this,
             transport_sender_,
@@ -34,10 +47,16 @@ AudioSender::AudioSender(scoped_refptr<CastEnvironment> cast_environment,
             audio_config.incoming_feedback_ssrc,
             audio_config.rtcp_c_name,
             AUDIO_EVENT),
+      rtp_timestamp_helper_(audio_config.frequency),
       num_aggressive_rtcp_reports_sent_(0),
+      last_sent_frame_id_(0),
+      latest_acked_frame_id_(0),
+      duplicate_ack_counter_(0),
       cast_initialization_status_(STATUS_AUDIO_UNINITIALIZED),
       weak_factory_(this) {
-  rtcp_.SetCastReceiverEventHistorySize(kReceiverRtcpEventHistorySize);
+  VLOG(1) << "max_unacked_frames " << max_unacked_frames_;
+  DCHECK_GT(max_unacked_frames_, 0);
+
   if (!audio_config.use_external_encoder) {
     audio_encoder_.reset(
         new AudioEncoder(cast_environment,
@@ -47,7 +66,7 @@ AudioSender::AudioSender(scoped_refptr<CastEnvironment> cast_environment,
     cast_initialization_status_ = audio_encoder_->InitializationResult();
   } else {
     NOTREACHED();  // No support for external audio encoding.
-    cast_initialization_status_ = STATUS_AUDIO_INITIALIZED;
+    cast_initialization_status_ = STATUS_AUDIO_UNINITIALIZED;
   }
 
   media::cast::transport::CastTransportAudioConfig transport_config;
@@ -55,10 +74,11 @@ AudioSender::AudioSender(scoped_refptr<CastEnvironment> cast_environment,
   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 =
-      audio_config.rtp_config.max_delay_ms / 100 + 1;
+  transport_config.rtp.max_outstanding_frames = max_unacked_frames_;
   transport_sender_->InitializeAudio(transport_config);
 
+  rtcp_.SetCastReceiverEventHistorySize(kReceiverRtcpEventHistorySize);
+
   memset(frame_id_to_rtp_timestamp_, 0, sizeof(frame_id_to_rtp_timestamp_));
 }
 
@@ -72,16 +92,43 @@ void AudioSender::InsertAudio(scoped_ptr<AudioBus> audio_bus,
     return;
   }
   DCHECK(audio_encoder_.get()) << "Invalid internal state";
+
+  if (AreTooManyFramesInFlight()) {
+    VLOG(1) << "Dropping frame due to too many frames currently in-flight.";
+    return;
+  }
+
   audio_encoder_->InsertAudio(audio_bus.Pass(), recorded_time);
 }
 
 void AudioSender::SendEncodedAudioFrame(
-    scoped_ptr<transport::EncodedFrame> audio_frame) {
+    scoped_ptr<transport::EncodedFrame> encoded_frame) {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
 
-  DCHECK(!audio_frame->reference_time.is_null());
-  rtp_timestamp_helper_.StoreLatestTime(audio_frame->reference_time,
-                                        audio_frame->rtp_timestamp);
+  const uint32 frame_id = encoded_frame->frame_id;
+
+  const bool is_first_frame_to_be_sent = last_send_time_.is_null();
+  last_send_time_ = cast_environment_->Clock()->NowTicks();
+  last_sent_frame_id_ = frame_id;
+  // If this is the first frame about to be sent, fake the value of
+  // |latest_acked_frame_id_| to indicate the receiver starts out all caught up.
+  // Also, schedule the periodic frame re-send checks.
+  if (is_first_frame_to_be_sent) {
+    latest_acked_frame_id_ = frame_id - 1;
+    ScheduleNextResendCheck();
+  }
+
+  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,
+      configured_encoder_bitrate_);
+  // Only use lowest 8 bits as key.
+  frame_id_to_rtp_timestamp_[frame_id & 0xff] = encoded_frame->rtp_timestamp;
+
+  DCHECK(!encoded_frame->reference_time.is_null());
+  rtp_timestamp_helper_.StoreLatestTime(encoded_frame->reference_time,
+                                        encoded_frame->rtp_timestamp);
 
   // At the start of the session, it's important to send reports before each
   // frame so that the receiver can properly compute playout times.  The reason
@@ -98,15 +145,7 @@ void AudioSender::SendEncodedAudioFrame(
     SendRtcpReport(is_last_aggressive_report);
   }
 
-  frame_id_to_rtp_timestamp_[audio_frame->frame_id & 0xff] =
-      audio_frame->rtp_timestamp;
-  transport_sender_->InsertCodedAudioFrame(*audio_frame);
-}
-
-void AudioSender::ResendPackets(
-    const MissingFramesAndPacketsMap& missing_frames_and_packets) {
-  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
-  transport_sender_->ResendPackets(true, missing_frames_and_packets, false);
+  transport_sender_->InsertCodedAudioFrame(*encoded_frame);
 }
 
 void AudioSender::IncomingRtcpPacket(scoped_ptr<Packet> packet) {
@@ -146,6 +185,37 @@ void AudioSender::SendRtcpReport(bool schedule_future_reports) {
     ScheduleNextRtcpReport();
 }
 
+void AudioSender::ScheduleNextResendCheck() {
+  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
+  DCHECK(!last_send_time_.is_null());
+  base::TimeDelta time_to_next =
+      last_send_time_ - cast_environment_->Clock()->NowTicks() +
+      target_playout_delay_;
+  time_to_next = std::max(
+      time_to_next, base::TimeDelta::FromMilliseconds(kMinSchedulingDelayMs));
+  cast_environment_->PostDelayedTask(
+      CastEnvironment::MAIN,
+      FROM_HERE,
+      base::Bind(&AudioSender::ResendCheck, weak_factory_.GetWeakPtr()),
+      time_to_next);
+}
+
+void AudioSender::ResendCheck() {
+  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
+  DCHECK(!last_send_time_.is_null());
+  const base::TimeDelta time_since_last_send =
+      cast_environment_->Clock()->NowTicks() - last_send_time_;
+  if (time_since_last_send > target_playout_delay_) {
+    if (latest_acked_frame_id_ == last_sent_frame_id_) {
+      // Last frame acked, no point in doing anything
+    } else {
+      VLOG(1) << "ACK timeout; last acked frame: " << latest_acked_frame_id_;
+      ResendForKickstart();
+    }
+  }
+  ScheduleNextResendCheck();
+}
+
 void AudioSender::OnReceivedCastFeedback(const RtcpCastMessage& cast_feedback) {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
 
@@ -161,15 +231,105 @@ void AudioSender::OnReceivedCastFeedback(const RtcpCastMessage& cast_feedback) {
     }
   }
 
-  if (!cast_feedback.missing_frames_and_packets_.empty()) {
-    ResendPackets(cast_feedback.missing_frames_and_packets_);
+  if (last_send_time_.is_null())
+    return;  // Cannot get an ACK without having first sent a frame.
+
+  if (cast_feedback.missing_frames_and_packets_.empty()) {
+    // We only count duplicate ACKs when we have sent newer frames.
+    if (latest_acked_frame_id_ == cast_feedback.ack_frame_id_ &&
+        latest_acked_frame_id_ != last_sent_frame_id_) {
+      duplicate_ack_counter_++;
+    } else {
+      duplicate_ack_counter_ = 0;
+    }
+    // TODO(miu): The values "2" and "3" should be derived from configuration.
+    if (duplicate_ack_counter_ >= 2 && duplicate_ack_counter_ % 3 == 2) {
+      VLOG(1) << "Received duplicate ACK for frame " << latest_acked_frame_id_;
+      ResendForKickstart();
+    }
+  } else {
+    // Only count duplicated ACKs if there is no NACK request in between.
+    // This is to avoid aggresive resend.
+    duplicate_ack_counter_ = 0;
+
+    base::TimeDelta rtt;
+    base::TimeDelta avg_rtt;
+    base::TimeDelta min_rtt;
+    base::TimeDelta max_rtt;
+    rtcp_.Rtt(&rtt, &avg_rtt, &min_rtt, &max_rtt);
+
+    // A NACK is also used to cancel pending re-transmissions.
+    transport_sender_->ResendPackets(
+        true, cast_feedback.missing_frames_and_packets_, false, min_rtt);
   }
-  uint32 acked_frame_id = static_cast<uint32>(cast_feedback.ack_frame_id_);
-  VLOG(2) << "Received audio ACK: " << acked_frame_id;
-  cast_environment_->Logging()->InsertFrameEvent(
-        cast_environment_->Clock()->NowTicks(),
-        FRAME_ACK_RECEIVED, AUDIO_EVENT,
-        frame_id_to_rtp_timestamp_[acked_frame_id & 0xff], acked_frame_id);
+
+  const base::TimeTicks now = cast_environment_->Clock()->NowTicks();
+
+  const RtpTimestamp rtp_timestamp =
+      frame_id_to_rtp_timestamp_[cast_feedback.ack_frame_id_ & 0xff];
+  cast_environment_->Logging()->InsertFrameEvent(now,
+                                                 FRAME_ACK_RECEIVED,
+                                                 AUDIO_EVENT,
+                                                 rtp_timestamp,
+                                                 cast_feedback.ack_frame_id_);
+
+  const bool is_acked_out_of_order =
+      static_cast<int32>(cast_feedback.ack_frame_id_ -
+                             latest_acked_frame_id_) < 0;
+  VLOG(2) << "Received ACK" << (is_acked_out_of_order ? " out-of-order" : "")
+          << " for frame " << cast_feedback.ack_frame_id_;
+  if (!is_acked_out_of_order) {
+    // Cancel resends of acked frames.
+    MissingFramesAndPacketsMap missing_frames_and_packets;
+    PacketIdSet missing;
+    while (latest_acked_frame_id_ != cast_feedback.ack_frame_id_) {
+      latest_acked_frame_id_++;
+      missing_frames_and_packets[latest_acked_frame_id_] = missing;
+    }
+    transport_sender_->ResendPackets(
+        true, missing_frames_and_packets, true, base::TimeDelta());
+    latest_acked_frame_id_ = cast_feedback.ack_frame_id_;
+  }
+}
+
+bool AudioSender::AreTooManyFramesInFlight() const {
+  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
+  int frames_in_flight = 0;
+  if (!last_send_time_.is_null()) {
+    frames_in_flight +=
+        static_cast<int32>(last_sent_frame_id_ - latest_acked_frame_id_);
+  }
+  VLOG(2) << frames_in_flight
+          << " frames in flight; last sent: " << last_sent_frame_id_
+          << " latest acked: " << latest_acked_frame_id_;
+  return frames_in_flight >= max_unacked_frames_;
+}
+
+void AudioSender::ResendForKickstart() {
+  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
+  DCHECK(!last_send_time_.is_null());
+  VLOG(1) << "Resending last packet of frame " << last_sent_frame_id_
+          << " to kick-start.";
+  // Send the first packet of the last encoded frame to kick start
+  // retransmission. This gives enough information to the receiver what
+  // packets and frames are missing.
+  MissingFramesAndPacketsMap missing_frames_and_packets;
+  PacketIdSet missing;
+  missing.insert(kRtcpCastLastPacket);
+  missing_frames_and_packets.insert(
+      std::make_pair(last_sent_frame_id_, missing));
+  last_send_time_ = cast_environment_->Clock()->NowTicks();
+
+  base::TimeDelta rtt;
+  base::TimeDelta avg_rtt;
+  base::TimeDelta min_rtt;
+  base::TimeDelta max_rtt;
+  rtcp_.Rtt(&rtt, &avg_rtt, &min_rtt, &max_rtt);
+
+  // Sending this extra packet is to kick-start the session. There is
+  // no need to optimize re-transmission for this case.
+  transport_sender_->ResendPackets(
+      true, missing_frames_and_packets, false, min_rtt);
 }
 
 }  // namespace cast
diff --git a/media/cast/audio_sender/audio_sender.h b/media/cast/audio_sender/audio_sender.h
index 03de633bef..80cf8a4e9e 100644
--- a/media/cast/audio_sender/audio_sender.h
+++ b/media/cast/audio_sender/audio_sender.h
@@ -14,6 +14,8 @@
 #include "base/time/time.h"
 #include "media/base/audio_bus.h"
 #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"
 
@@ -22,8 +24,12 @@ namespace cast {
 
 class AudioEncoder;
 
-// This class is not thread safe.
-// It's only called from the main cast thread.
+// Not thread safe. Only called from the main cast thread.
+// This class owns all objects related to sending audio, objects that create RTP
+// packets, congestion control, audio encoder, parsing and sending of
+// RTCP packets.
+// Additionally it posts a bunch of delayed tasks to the main thread for various
+// timeouts.
 class AudioSender : public RtcpSenderFeedback,
                     public base::NonThreadSafe,
                     public base::SupportsWeakPtr<AudioSender> {
@@ -38,6 +44,10 @@ class AudioSender : public RtcpSenderFeedback,
     return cast_initialization_status_;
   }
 
+  // Note: It is not guaranteed that |audio_frame| will actually be encoded and
+  // sent, if AudioSender detects too many frames in flight.  Therefore, clients
+  // should be careful about the rate at which this method is called.
+  //
   // Note: It is invalid to call this method if InitializationResult() returns
   // anything but STATUS_AUDIO_INITIALIZED.
   void InsertAudio(scoped_ptr<AudioBus> audio_bus,
@@ -46,31 +56,98 @@ class AudioSender : public RtcpSenderFeedback,
   // Only called from the main cast thread.
   void IncomingRtcpPacket(scoped_ptr<Packet> packet);
 
- private:
-  void ResendPackets(
-      const MissingFramesAndPacketsMap& missing_frames_and_packets);
+ protected:
+  // Protected for testability.
+  virtual void OnReceivedCastFeedback(const RtcpCastMessage& cast_feedback)
+      OVERRIDE;
 
+ private:
+  // Schedule and execute periodic sending of RTCP report.
   void ScheduleNextRtcpReport();
   void SendRtcpReport(bool schedule_future_reports);
 
+  // Schedule and execute periodic checks for re-sending packets.  If no
+  // acknowledgements have been received for "too long," AudioSender will
+  // speculatively re-send certain packets of an unacked frame to kick-start
+  // re-transmission.  This is a last resort tactic to prevent the session from
+  // getting stuck after a long outage.
+  void ScheduleNextResendCheck();
+  void ResendCheck();
+  void ResendForKickstart();
+
+  // Returns true if there are too many frames in flight, as defined by the
+  // configured target playout delay plus simple logic.  When this is true,
+  // InsertAudio() will silenty drop frames instead of sending them to the audio
+  // encoder.
+  bool AreTooManyFramesInFlight() const;
+
   // Called by the |audio_encoder_| with the next EncodedFrame to send.
   void SendEncodedAudioFrame(scoped_ptr<transport::EncodedFrame> audio_frame);
 
-  virtual void OnReceivedCastFeedback(const RtcpCastMessage& cast_feedback)
-      OVERRIDE;
-
-  scoped_refptr<CastEnvironment> cast_environment_;
+  const scoped_refptr<CastEnvironment> cast_environment_;
+
+  // The total amount of time between a frame's capture/recording on the sender
+  // and its playback on the receiver (i.e., shown to a user).  This is fixed as
+  // a value large enough to give the system sufficient time to encode,
+  // transmit/retransmit, receive, decode, and render; given its run-time
+  // environment (sender/receiver hardware performance, network conditions,
+  // etc.).
+  const base::TimeDelta target_playout_delay_;
+
+  // Sends encoded frames over the configured transport (e.g., UDP).  In
+  // Chromium, this could be a proxy that first sends the frames from a renderer
+  // 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_;
+
+  // Maximum number of outstanding frames before the encoding and sending of
+  // new frames shall halt.
+  const int max_unacked_frames_;
+
+  // Encodes AudioBuses into EncodedFrames.
   scoped_ptr<AudioEncoder> audio_encoder_;
-  RtpTimestampHelper rtp_timestamp_helper_;
+  const int configured_encoder_bitrate_;
+
+  // Manages sending/receiving of RTCP packets, including sender/receiver
+  // reports.
   Rtcp rtcp_;
+
+  // Records lip-sync (i.e., mapping of RTP <--> NTP timestamps), and
+  // extrapolates this mapping to any other point in time.
+  RtpTimestampHelper rtp_timestamp_helper_;
+
+  // Counts how many RTCP reports are being "aggressively" sent (i.e., one per
+  // frame) at the start of the session.  Once a threshold is reached, RTCP
+  // reports are instead sent at the configured interval + random drift.
   int num_aggressive_rtcp_reports_sent_;
 
+  // This is "null" until the first frame is sent.  Thereafter, this tracks the
+  // last time any frame was sent or re-sent.
+  base::TimeTicks last_send_time_;
+
+  // The ID of the last frame sent.  Logic throughout AudioSender assumes this
+  // can safely wrap-around.  This member is invalid until
+  // |!last_send_time_.is_null()|.
+  uint32 last_sent_frame_id_;
+
+  // The ID of the latest (not necessarily the last) frame that has been
+  // acknowledged.  Logic throughout AudioSender assumes this can safely
+  // wrap-around.  This member is invalid until |!last_send_time_.is_null()|.
+  uint32 latest_acked_frame_id_;
+
+  // Counts the number of duplicate ACK that are being received.  When this
+  // number reaches a threshold, the sender will take this as a sign that the
+  // receiver hasn't yet received the first packet of the next frame.  In this
+  // case, AudioSender will trigger a re-send of the next frame.
+  int duplicate_ack_counter_;
+
   // If this sender is ready for use, this is STATUS_AUDIO_INITIALIZED.
   CastInitializationStatus cast_initialization_status_;
 
-  // Used to map the lower 8 bits of the frame id to a RTP timestamp. This is
-  // good enough as we only use it for logging.
+  // This is a "good enough" mapping for finding the RTP timestamp associated
+  // with a video frame. The key is the lowest 8 bits of frame id (which is
+  // what is sent via RTCP). This map is used for logging purposes.
   RtpTimestamp frame_id_to_rtp_timestamp_[256];
 
   // NOTE: Weak pointers must be invalidated before all other member variables.
diff --git a/media/cast/logging/log_deserializer.cc b/media/cast/logging/log_deserializer.cc
index a4c79b3de9..1c6dd57224 100644
--- a/media/cast/logging/log_deserializer.cc
+++ b/media/cast/logging/log_deserializer.cc
@@ -35,7 +35,12 @@ void MergePacketEvent(const AggregatedPacketEvent& from,
     for (int j = 0; j < to->base_packet_event_size(); j++) {
       BasePacketEvent* to_base_event = to->mutable_base_packet_event(j);
       if (from_base_event.packet_id() == to_base_event->packet_id()) {
+        int packet_size = std::max(
+            from_base_event.size(), to_base_event->size());
+        // Need special merge logic here because we need to prevent a valid
+        // packet size (> 0) from being overwritten with an invalid one (= 0).
         to_base_event->MergeFrom(from_base_event);
+        to_base_event->set_size(packet_size);
         merged = true;
         break;
       }
diff --git a/media/cast/logging/logging_defines.cc b/media/cast/logging/logging_defines.cc
index d0dd5c8e57..05ceeb9521 100644
--- a/media/cast/logging/logging_defines.cc
+++ b/media/cast/logging/logging_defines.cc
@@ -25,6 +25,7 @@ const char* CastLoggingToString(CastLoggingEvent event) {
     ENUM_TO_STRING(FRAME_PLAYOUT);
     ENUM_TO_STRING(PACKET_SENT_TO_NETWORK);
     ENUM_TO_STRING(PACKET_RETRANSMITTED);
+    ENUM_TO_STRING(PACKET_RTX_REJECTED);
     ENUM_TO_STRING(PACKET_RECEIVED);
   }
   NOTREACHED();
diff --git a/media/cast/logging/logging_defines.h b/media/cast/logging/logging_defines.h
index b3f3841ffb..021a3c99a7 100644
--- a/media/cast/logging/logging_defines.h
+++ b/media/cast/logging/logging_defines.h
@@ -32,6 +32,7 @@ enum CastLoggingEvent {
   // Sender side packet events.
   PACKET_SENT_TO_NETWORK,
   PACKET_RETRANSMITTED,
+  PACKET_RTX_REJECTED,
   // Receiver side packet events.
   PACKET_RECEIVED,
   kNumOfLoggingEvents = PACKET_RECEIVED
diff --git a/media/cast/logging/proto/proto_utils.cc b/media/cast/logging/proto/proto_utils.cc
index 1f05616d83..03251e64c0 100644
--- a/media/cast/logging/proto/proto_utils.cc
+++ b/media/cast/logging/proto/proto_utils.cc
@@ -25,6 +25,7 @@ proto::EventType ToProtoEventType(CastLoggingEvent event) {
     TO_PROTO_ENUM(FRAME_PLAYOUT);
     TO_PROTO_ENUM(PACKET_SENT_TO_NETWORK);
     TO_PROTO_ENUM(PACKET_RETRANSMITTED);
+    TO_PROTO_ENUM(PACKET_RTX_REJECTED);
     TO_PROTO_ENUM(PACKET_RECEIVED);
   }
   NOTREACHED();
diff --git a/media/cast/logging/proto/raw_events.proto b/media/cast/logging/proto/raw_events.proto
index 08bf53d19a..1d2c537db8 100644
--- a/media/cast/logging/proto/raw_events.proto
+++ b/media/cast/logging/proto/raw_events.proto
@@ -66,6 +66,7 @@ enum EventType {
   PACKET_SENT_TO_NETWORK = 36;
   PACKET_RETRANSMITTED = 37;
   PACKET_RECEIVED = 38;
+  PACKET_RTX_REJECTED = 39;
 }
 
 // Contains information independent of the stream that describes the system
diff --git a/media/cast/rtcp/rtcp_sender_unittest.cc b/media/cast/rtcp/rtcp_sender_unittest.cc
index 2bd807f380..0b0c7d3ab8 100644
--- a/media/cast/rtcp/rtcp_sender_unittest.cc
+++ b/media/cast/rtcp/rtcp_sender_unittest.cc
@@ -59,7 +59,8 @@ class TestRtcpTransport : public transport::PacedPacketSender {
     return false;
   }
   virtual bool ResendPackets(
-      const transport::SendPacketVector& packets) OVERRIDE {
+      const transport::SendPacketVector& packets,
+      base::TimeDelta dedupe_window) OVERRIDE {
     return false;
   }
 
diff --git a/media/cast/rtcp/rtcp_unittest.cc b/media/cast/rtcp/rtcp_unittest.cc
index d5bf312c3e..095e6d24df 100644
--- a/media/cast/rtcp/rtcp_unittest.cc
+++ b/media/cast/rtcp/rtcp_unittest.cc
@@ -104,7 +104,8 @@ class LocalRtcpTransport : public transport::PacedPacketSender {
   }
 
   virtual bool ResendPackets(
-      const transport::SendPacketVector& packets) OVERRIDE {
+      const transport::SendPacketVector& packets,
+      base::TimeDelta dedupe_window) OVERRIDE {
     return false;
   }
 
diff --git a/media/cast/test/cast_benchmarks.cc b/media/cast/test/cast_benchmarks.cc
index c3468a2cdb..66257626bd 100644
--- a/media/cast/test/cast_benchmarks.cc
+++ b/media/cast/test/cast_benchmarks.cc
@@ -212,9 +212,10 @@ class CastTransportSenderWrapper : public transport::CastTransportSender {
   virtual void ResendPackets(
       bool is_audio,
       const MissingFramesAndPacketsMap& missing_packets,
-      bool cancel_rtx_if_not_in_list) OVERRIDE {
+      bool cancel_rtx_if_not_in_list,
+      base::TimeDelta dedupe_window) OVERRIDE {
     transport_->ResendPackets(
-        is_audio, missing_packets, cancel_rtx_if_not_in_list);
+        is_audio, missing_packets, cancel_rtx_if_not_in_list, dedupe_window);
   }
 
  private:
diff --git a/media/cast/transport/cast_transport_sender.h b/media/cast/transport/cast_transport_sender.h
index 2556a8bd3d..e88f2f4f09 100644
--- a/media/cast/transport/cast_transport_sender.h
+++ b/media/cast/transport/cast_transport_sender.h
@@ -96,11 +96,14 @@ class CastTransportSender : public base::NonThreadSafe {
   // frame to be re-transmitted.
   // If |cancel_rtx_if_not_in_list| is used as an optimization to cancel
   // pending re-transmission requests of packets not listed in
-  // |missing_packets|.
+  // |missing_packets|. If the requested packet(s) were sent recently
+  // (how long is specified by |dedupe_window|) then this re-transmit
+  // will be ignored.
   virtual void ResendPackets(
       bool is_audio,
       const MissingFramesAndPacketsMap& missing_packets,
-      bool cancel_rtx_if_not_in_list) = 0;
+      bool cancel_rtx_if_not_in_list,
+      base::TimeDelta dedupe_window) = 0;
 };
 
 }  // namespace transport
diff --git a/media/cast/transport/cast_transport_sender_impl.cc b/media/cast/transport/cast_transport_sender_impl.cc
index 2f51a934e7..6fd848f27b 100644
--- a/media/cast/transport/cast_transport_sender_impl.cc
+++ b/media/cast/transport/cast_transport_sender_impl.cc
@@ -7,6 +7,7 @@
 #include "base/single_thread_task_runner.h"
 #include "media/cast/transport/cast_transport_config.h"
 #include "media/cast/transport/cast_transport_defines.h"
+#include "net/base/net_util.h"
 
 namespace media {
 namespace cast {
@@ -66,6 +67,11 @@ CastTransportSenderImpl::CastTransportSenderImpl(
                             this,
                             &CastTransportSenderImpl::SendRawEvents);
   }
+  if (transport_) {
+    // The default DSCP value for cast is AF41. Which gives it a higher
+    // priority over other traffic.
+    transport_->SetDscp(net::DSCP_AF41);
+  }
 }
 
 CastTransportSenderImpl::~CastTransportSenderImpl() {
@@ -178,15 +184,18 @@ void CastTransportSenderImpl::SendRtcpFromRtpSender(
 void CastTransportSenderImpl::ResendPackets(
     bool is_audio,
     const MissingFramesAndPacketsMap& missing_packets,
-    bool cancel_rtx_if_not_in_list) {
+    bool cancel_rtx_if_not_in_list,
+    base::TimeDelta dedupe_window) {
   if (is_audio) {
     DCHECK(audio_sender_) << "Audio sender uninitialized";
     audio_sender_->ResendPackets(missing_packets,
-                                 cancel_rtx_if_not_in_list);
+                                 cancel_rtx_if_not_in_list,
+                                 dedupe_window);
   } else {
     DCHECK(video_sender_) << "Video sender uninitialized";
     video_sender_->ResendPackets(missing_packets,
-                                 cancel_rtx_if_not_in_list);
+                                 cancel_rtx_if_not_in_list,
+                                 dedupe_window);
   }
 }
 
diff --git a/media/cast/transport/cast_transport_sender_impl.h b/media/cast/transport/cast_transport_sender_impl.h
index 4fc074c0b9..035ef844b6 100644
--- a/media/cast/transport/cast_transport_sender_impl.h
+++ b/media/cast/transport/cast_transport_sender_impl.h
@@ -67,7 +67,8 @@ class CastTransportSenderImpl : public CastTransportSender {
 
   virtual void ResendPackets(bool is_audio,
                              const MissingFramesAndPacketsMap& missing_packets,
-                             bool cancel_rtx_if_not_in_list)
+                             bool cancel_rtx_if_not_in_list,
+                             base::TimeDelta dedupe_window)
       OVERRIDE;
 
  private:
diff --git a/media/cast/transport/pacing/mock_paced_packet_sender.h b/media/cast/transport/pacing/mock_paced_packet_sender.h
index 9f6d204f7f..20b7647035 100644
--- a/media/cast/transport/pacing/mock_paced_packet_sender.h
+++ b/media/cast/transport/pacing/mock_paced_packet_sender.h
@@ -18,7 +18,8 @@ class MockPacedPacketSender : public PacedPacketSender {
   virtual ~MockPacedPacketSender();
 
   MOCK_METHOD1(SendPackets, bool(const SendPacketVector& packets));
-  MOCK_METHOD1(ResendPackets, bool(const SendPacketVector& packets));
+  MOCK_METHOD2(ResendPackets, bool(const SendPacketVector& packets,
+                                   base::TimeDelta dedupe_window));
   MOCK_METHOD2(SendRtcpPacket, bool(unsigned int ssrc, PacketRef packet));
   MOCK_METHOD1(CancelSendingPacket, void(const PacketKey& packet_key));
 };
diff --git a/media/cast/transport/pacing/paced_sender.cc b/media/cast/transport/pacing/paced_sender.cc
index 10b4224075..20cbde85be 100644
--- a/media/cast/transport/pacing/paced_sender.cc
+++ b/media/cast/transport/pacing/paced_sender.cc
@@ -20,6 +20,7 @@ static const int64 kPacingIntervalMs = 10;
 static const size_t kPacingMaxBurstsPerFrame = 3;
 static const size_t kTargetBurstSize = 10;
 static const size_t kMaxBurstSize = 20;
+static const size_t kMaxDedupeWindowMs = 500;
 
 }  // namespace
 
@@ -73,11 +74,21 @@ bool PacedSender::SendPackets(const SendPacketVector& packets) {
   return true;
 }
 
-bool PacedSender::ResendPackets(const SendPacketVector& packets) {
+bool PacedSender::ResendPackets(const SendPacketVector& packets,
+                                base::TimeDelta dedupe_window) {
   if (packets.empty()) {
     return true;
   }
+  base::TimeTicks now = clock_->NowTicks();
   for (size_t i = 0; i < packets.size(); i++) {
+    std::map<PacketKey, base::TimeTicks>::const_iterator j =
+        sent_time_.find(packets[i].first);
+
+    if (j != sent_time_.end() && now - j->second < dedupe_window) {
+      LogPacketEvent(packets[i].second->data, PACKET_RTX_REJECTED);
+      continue;
+    }
+
     packet_list_[packets[i].first] =
         make_pair(PacketType_Resend, packets[i].second);
   }
@@ -108,11 +119,13 @@ void PacedSender::CancelSendingPacket(const PacketKey& packet_key) {
   packet_list_.erase(packet_key);
 }
 
-PacketRef PacedSender::GetNextPacket(PacketType* packet_type) {
+PacketRef PacedSender::GetNextPacket(PacketType* packet_type,
+                                     PacketKey* packet_key) {
   std::map<PacketKey, std::pair<PacketType, PacketRef> >::iterator i;
   i = packet_list_.begin();
   DCHECK(i != packet_list_.end());
   *packet_type = i->second.first;
+  *packet_key = i->first;
   PacketRef ret = i->second.second;
   packet_list_.erase(i);
   return ret;
@@ -185,14 +198,17 @@ void PacedSender::SendStoredPackets() {
       return;
     }
     PacketType packet_type;
-    PacketRef packet = GetNextPacket(&packet_type);
+    PacketKey packet_key;
+    PacketRef packet = GetNextPacket(&packet_type, &packet_key);
+    sent_time_[packet_key] = now;
+    sent_time_buffer_[packet_key] = now;
 
     switch (packet_type) {
       case PacketType_Resend:
-        LogPacketEvent(packet->data, true);
+        LogPacketEvent(packet->data, PACKET_RETRANSMITTED);
         break;
       case PacketType_Normal:
-        LogPacketEvent(packet->data, false);
+        LogPacketEvent(packet->data, PACKET_SENT_TO_NETWORK);
         break;
       case PacketType_RTCP:
         break;
@@ -203,10 +219,20 @@ void PacedSender::SendStoredPackets() {
     }
     current_burst_size_++;
   }
+  // Keep ~0.5 seconds of data (1000 packets)
+  if (sent_time_buffer_.size() >=
+      kMaxBurstSize * kMaxDedupeWindowMs / kPacingIntervalMs) {
+    sent_time_.swap(sent_time_buffer_);
+    sent_time_buffer_.clear();
+  }
+  DCHECK_LE(sent_time_buffer_.size(),
+            kMaxBurstSize * kMaxDedupeWindowMs / kPacingIntervalMs);
+  DCHECK_LE(sent_time_.size(),
+            2 * kMaxBurstSize * kMaxDedupeWindowMs / kPacingIntervalMs);
   state_ = State_Unblocked;
 }
 
-void PacedSender::LogPacketEvent(const Packet& packet, bool retransmit) {
+void PacedSender::LogPacketEvent(const Packet& packet, CastLoggingEvent event) {
   // Get SSRC from packet and compare with the audio_ssrc / video_ssrc to see
   // if the packet is audio or video.
   DCHECK_GE(packet.size(), 12u);
@@ -224,8 +250,6 @@ void PacedSender::LogPacketEvent(const Packet& packet, bool retransmit) {
     return;
   }
 
-  CastLoggingEvent event = retransmit ?
-      PACKET_RETRANSMITTED : PACKET_SENT_TO_NETWORK;
   EventMediaType media_type = is_audio ? AUDIO_EVENT : VIDEO_EVENT;
   logging_->InsertSinglePacketEvent(clock_->NowTicks(), event, media_type,
       packet);
diff --git a/media/cast/transport/pacing/paced_sender.h b/media/cast/transport/pacing/paced_sender.h
index 2373fb5966..9fc0c8b8b8 100644
--- a/media/cast/transport/pacing/paced_sender.h
+++ b/media/cast/transport/pacing/paced_sender.h
@@ -41,7 +41,8 @@ typedef std::vector<std::pair<PacketKey, PacketRef> > SendPacketVector;
 class PacedPacketSender {
  public:
   virtual bool SendPackets(const SendPacketVector& packets) = 0;
-  virtual bool ResendPackets(const SendPacketVector& packets) = 0;
+  virtual bool ResendPackets(const SendPacketVector& packets,
+                             base::TimeDelta dedupe_window) = 0;
   virtual bool SendRtcpPacket(uint32 ssrc, PacketRef packet) = 0;
   virtual void CancelSendingPacket(const PacketKey& packet_key) = 0;
 
@@ -72,14 +73,15 @@ class PacedSender : public PacedPacketSender,
 
   // PacedPacketSender implementation.
   virtual bool SendPackets(const SendPacketVector& packets) OVERRIDE;
-  virtual bool ResendPackets(const SendPacketVector& packets) OVERRIDE;
+  virtual bool ResendPackets(const SendPacketVector& packets,
+                             base::TimeDelta dedupe_window) OVERRIDE;
   virtual bool SendRtcpPacket(uint32 ssrc, PacketRef packet) OVERRIDE;
   virtual void CancelSendingPacket(const PacketKey& packet_key) OVERRIDE;
 
  private:
   // Actually sends the packets to the transport.
   void SendStoredPackets();
-  void LogPacketEvent(const Packet& packet, bool retransmit);
+  void LogPacketEvent(const Packet& packet, CastLoggingEvent event);
 
   enum PacketType {
     PacketType_RTCP,
@@ -108,7 +110,8 @@ 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 GetNextPacket(PacketType* packet_type,
+                          PacketKey* packet_key);
 
   base::TickClock* const clock_;  // Not owned by this class.
   LoggingImpl* const logging_;    // Not owned by this class.
@@ -117,6 +120,8 @@ class PacedSender : public PacedPacketSender,
   uint32 audio_ssrc_;
   uint32 video_ssrc_;
   std::map<PacketKey, std::pair<PacketType, PacketRef> > packet_list_;
+  std::map<PacketKey, base::TimeTicks> sent_time_;
+  std::map<PacketKey, base::TimeTicks> sent_time_buffer_;
 
   // Maximum burst size for the next three bursts.
   size_t max_burst_size_;
diff --git a/media/cast/transport/pacing/paced_sender_unittest.cc b/media/cast/transport/pacing/paced_sender_unittest.cc
index ef9d89b5f0..5e24fca4b5 100644
--- a/media/cast/transport/pacing/paced_sender_unittest.cc
+++ b/media/cast/transport/pacing/paced_sender_unittest.cc
@@ -129,7 +129,7 @@ TEST_F(PacedSenderTest, PassThroughRtcp) {
   SendPacketVector packets = CreateSendPacketVector(kSize1, 1, true);
 
   EXPECT_TRUE(paced_sender_->SendPackets(packets));
-  EXPECT_TRUE(paced_sender_->ResendPackets(packets));
+  EXPECT_TRUE(paced_sender_->ResendPackets(packets, base::TimeDelta()));
 
   mock_transport_.AddExpectedSize(kSize2, 1);
   Packet tmp(kSize2, kValue);
@@ -202,7 +202,7 @@ TEST_F(PacedSenderTest, PaceWithNack) {
   EXPECT_TRUE(paced_sender_->SendPackets(first_frame_packets));
 
   // Add first NACK request.
-  EXPECT_TRUE(paced_sender_->ResendPackets(nack_packets));
+  EXPECT_TRUE(paced_sender_->ResendPackets(nack_packets, base::TimeDelta()));
 
   // Check that we get the first NACK burst.
   mock_transport_.AddExpectedSize(kNackSize, 10);
@@ -211,7 +211,7 @@ TEST_F(PacedSenderTest, PaceWithNack) {
   task_runner_->RunTasks();
 
   // Add second NACK request.
-  EXPECT_TRUE(paced_sender_->ResendPackets(nack_packets));
+  EXPECT_TRUE(paced_sender_->ResendPackets(nack_packets, base::TimeDelta()));
 
   // Check that we get the next NACK burst.
   mock_transport_.AddExpectedSize(kNackSize, 10);
diff --git a/media/cast/transport/rtp_sender/rtp_sender.cc b/media/cast/transport/rtp_sender/rtp_sender.cc
index 2604d253de..b807b34757 100644
--- a/media/cast/transport/rtp_sender/rtp_sender.cc
+++ b/media/cast/transport/rtp_sender/rtp_sender.cc
@@ -4,6 +4,7 @@
 
 #include "media/cast/transport/rtp_sender/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"
@@ -75,7 +76,8 @@ void RtpSender::SendFrame(const EncodedFrame& frame) {
 
 void RtpSender::ResendPackets(
     const MissingFramesAndPacketsMap& missing_frames_and_packets,
-    bool cancel_rtx_if_not_in_list) {
+    bool cancel_rtx_if_not_in_list,
+    base::TimeDelta dedupe_window) {
   DCHECK(storage_);
   // Iterate over all frames in the list.
   for (MissingFramesAndPacketsMap::const_iterator it =
@@ -129,15 +131,18 @@ void RtpSender::ResendPackets(
         transport_->CancelSendingPacket(it->first);
       }
     }
-    transport_->ResendPackets(packets_to_resend);
+    transport_->ResendPackets(packets_to_resend, dedupe_window);
   }
 }
 
 void RtpSender::UpdateSequenceNumber(Packet* packet) {
-  uint16 new_sequence_number = packetizer_->NextSequenceNumber();
-  int index = 2;
-  (*packet)[index] = (static_cast<uint8>(new_sequence_number));
-  (*packet)[index + 1] = (static_cast<uint8>(new_sequence_number >> 8));
+  // TODO(miu): This is an abstraction violation.  This needs to be a part of
+  // the overall packet (de)serialization consolidation.
+  static const int kByteOffsetToSequenceNumber = 2;
+  base::BigEndianWriter big_endian_writer(
+      reinterpret_cast<char*>((&packet->front()) + kByteOffsetToSequenceNumber),
+      sizeof(uint16));
+  big_endian_writer.WriteU16(packetizer_->NextSequenceNumber());
 }
 
 }  // namespace transport
diff --git a/media/cast/transport/rtp_sender/rtp_sender.h b/media/cast/transport/rtp_sender/rtp_sender.h
index bfb46cb09e..e65326abf1 100644
--- a/media/cast/transport/rtp_sender/rtp_sender.h
+++ b/media/cast/transport/rtp_sender/rtp_sender.h
@@ -51,7 +51,8 @@ class RtpSender {
   void SendFrame(const EncodedFrame& frame);
 
   void ResendPackets(const MissingFramesAndPacketsMap& missing_packets,
-                     bool cancel_rtx_if_not_in_list);
+                     bool cancel_rtx_if_not_in_list,
+                     base::TimeDelta dedupe_window);
 
   size_t send_packet_count() const {
     return packetizer_ ? packetizer_->send_packet_count() : 0;
diff --git a/media/cast/transport/transport/udp_transport.cc b/media/cast/transport/transport/udp_transport.cc
index bcce4f72f3..9669b17d43 100644
--- a/media/cast/transport/transport/udp_transport.cc
+++ b/media/cast/transport/transport/udp_transport.cc
@@ -54,6 +54,7 @@ UdpTransport::UdpTransport(
       send_pending_(false),
       receive_pending_(false),
       client_connected_(false),
+      next_dscp_value_(net::DSCP_NO_CHANGE),
       status_callback_(status_callback),
       weak_factory_(this) {
   DCHECK(!IsEmpty(local_end_point) || !IsEmpty(remote_end_point));
@@ -88,6 +89,11 @@ void UdpTransport::StartReceiving(
   ScheduleReceiveNextPacket();
 }
 
+void UdpTransport::SetDscp(net::DiffServCodePoint dscp) {
+  DCHECK(io_thread_proxy_->RunsTasksOnCurrentThread());
+  next_dscp_value_ = dscp;
+}
+
 void UdpTransport::ScheduleReceiveNextPacket() {
   DCHECK(io_thread_proxy_->RunsTasksOnCurrentThread());
   if (!packet_receiver_.is_null() && !receive_pending_) {
@@ -162,6 +168,16 @@ bool UdpTransport::SendPacket(PacketRef packet, const base::Closure& cb) {
     return true;
   }
 
+  if (next_dscp_value_ != net::DSCP_NO_CHANGE) {
+    int result = udp_socket_->SetDiffServCodePoint(next_dscp_value_);
+    if (result != net::OK) {
+      LOG(ERROR) << "Unable to set DSCP: " << next_dscp_value_
+                 << " to socket; Error: " << result;
+    }
+    // Don't change DSCP in next send.
+    next_dscp_value_ = net::DSCP_NO_CHANGE;
+  }
+
   scoped_refptr<net::IOBuffer> buf =
       new net::WrappedIOBuffer(reinterpret_cast<char*>(&packet->data.front()));
 
diff --git a/media/cast/transport/transport/udp_transport.h b/media/cast/transport/transport/udp_transport.h
index 17b0d77599..1a568501d5 100644
--- a/media/cast/transport/transport/udp_transport.h
+++ b/media/cast/transport/transport/udp_transport.h
@@ -12,6 +12,7 @@
 #include "media/cast/transport/cast_transport_config.h"
 #include "media/cast/transport/cast_transport_sender.h"
 #include "net/base/ip_endpoint.h"
+#include "net/base/net_util.h"
 #include "net/udp/udp_socket.h"
 
 namespace net {
@@ -46,6 +47,10 @@ class UdpTransport : public PacketSender {
   // Start receiving packets. Packets are submitted to |packet_receiver|.
   void StartReceiving(const PacketReceiverCallback& packet_receiver);
 
+  // Set a new DSCP value to the socket. The value will be set right before
+  // the next send.
+  void SetDscp(net::DiffServCodePoint dscp);
+
   // PacketSender implementations.
   virtual bool SendPacket(PacketRef packet,
                           const base::Closure& cb) OVERRIDE;
@@ -72,6 +77,7 @@ class UdpTransport : public PacketSender {
   bool send_pending_;
   bool receive_pending_;
   bool client_connected_;
+  net::DiffServCodePoint next_dscp_value_;
   scoped_ptr<Packet> next_packet_;
   scoped_refptr<net::WrappedIOBuffer> recv_buf_;
   net::IPEndPoint recv_addr_;
diff --git a/media/cast/video_sender/video_sender.cc b/media/cast/video_sender/video_sender.cc
index cc8b158950..cf050b7f10 100644
--- a/media/cast/video_sender/video_sender.cc
+++ b/media/cast/video_sender/video_sender.cc
@@ -78,7 +78,7 @@ VideoSender::VideoSender(
   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_ + 1;
+  transport_config.rtp.max_outstanding_frames = max_unacked_frames_;
   transport_sender_->InitializeVideo(transport_config);
 
   rtcp_.SetCastReceiverEventHistorySize(kReceiverRtcpEventHistorySize);
@@ -321,7 +321,7 @@ void VideoSender::OnReceivedCastFeedback(const RtcpCastMessage& cast_feedback) {
 
     // A NACK is also used to cancel pending re-transmissions.
     transport_sender_->ResendPackets(
-        false, cast_feedback.missing_frames_and_packets_, true);
+        false, cast_feedback.missing_frames_and_packets_, true, rtt);
   }
 
   base::TimeTicks now = cast_environment_->Clock()->NowTicks();
@@ -348,7 +348,8 @@ void VideoSender::OnReceivedCastFeedback(const RtcpCastMessage& cast_feedback) {
       latest_acked_frame_id_++;
       missing_frames_and_packets[latest_acked_frame_id_] = missing;
     }
-    transport_sender_->ResendPackets(false, missing_frames_and_packets, true);
+    transport_sender_->ResendPackets(
+        false, missing_frames_and_packets, true, rtt);
     latest_acked_frame_id_ = cast_feedback.ack_frame_id_;
   }
 }
@@ -382,10 +383,16 @@ void VideoSender::ResendForKickstart() {
       std::make_pair(last_sent_frame_id_, missing));
   last_send_time_ = cast_environment_->Clock()->NowTicks();
 
+  base::TimeDelta rtt;
+  base::TimeDelta avg_rtt;
+  base::TimeDelta min_rtt;
+  base::TimeDelta max_rtt;
+  rtcp_.Rtt(&rtt, &avg_rtt, &min_rtt, &max_rtt);
+
   // Sending this extra packet is to kick-start the session. There is
   // no need to optimize re-transmission for this case.
   transport_sender_->ResendPackets(false, missing_frames_and_packets,
-                                   false);
+                                   false, rtt);
 }
 
 }  // namespace cast
diff --git a/media/cast/video_sender/video_sender.h b/media/cast/video_sender/video_sender.h
index 30066cbbf8..cf8d27511c 100644
--- a/media/cast/video_sender/video_sender.h
+++ b/media/cast/video_sender/video_sender.h
@@ -76,15 +76,13 @@ class VideoSender : public RtcpSenderFeedback,
   void ScheduleNextRtcpReport();
   void SendRtcpReport(bool schedule_future_reports);
 
-  // Schedule and execute periodic checks for re-sending frames.  If no
+  // Schedule and execute periodic checks for re-sending packets.  If no
   // acknowledgements have been received for "too long," VideoSender will
-  // speculatively re-send the frame just after |latest_acked_frame_id_| (the
-  // whole frame).  This is a last resort tactic to prevent the session from
+  // speculatively re-send certain packets of an unacked frame to kick-start
+  // re-transmission.  This is a last resort tactic to prevent the session from
   // getting stuck after a long outage.
   void ScheduleNextResendCheck();
   void ResendCheck();
-
-  // Resend certain packets of an unacked frame to kick start re-transmission.
   void ResendForKickstart();
 
   // Returns true if there are too many frames in flight, as defined by the
@@ -169,10 +167,7 @@ class VideoSender : public RtcpSenderFeedback,
 
   // This is a "good enough" mapping for finding the RTP timestamp associated
   // with a video frame. The key is the lowest 8 bits of frame id (which is
-  // what is sent via RTCP). This map is used for logging purposes. The only
-  // time when this mapping will be incorrect is when it receives an ACK for a
-  // old enough frame such that 8-bit wrap around has already occurred, which
-  // should be pretty rare.
+  // what is sent via RTCP). This map is used for logging purposes.
   RtpTimestamp frame_id_to_rtp_timestamp_[256];
 
   // NOTE: Weak pointers must be invalidated before all other member variables.