[Cast] Halt AudioSender transmission when too many frames are in-flight.

media/cast/audio_sender/audio_sender.cc

Index: media/cast/audio_sender/audio_sender.cc
diff --git a/media/cast/audio_sender/audio_sender.cc b/media/cast/audio_sender/audio_sender.cc
index 27b42d0dc8c2ac08757cb561d208f849c5835baa..ea44218d25b127a0c316642f600698f0138751f0 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 +231,92 @@ 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;
+
+    // A NACK is also used to cancel pending re-transmissions.
+    transport_sender_->ResendPackets(
+        true, cast_feedback.missing_frames_and_packets_, true);
+  }
+
+  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);
+    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_);
   }
-  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);
+  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();
+
+  // 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);
 }
 
 }  // namespace cast