[Cast] Limit frames in flight by duration, and not by number of frames.

media/cast/sender/audio_sender.cc

 // 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/sender/audio_sender.h"
 
 #include "base/bind.h"
 #include "base/logging.h"
 #include "base/message_loop/message_loop.h"
 #include "media/cast/cast_defines.h"
@@ skipping 14 matching lines @@
 AudioSender::AudioSender(scoped_refptr<CastEnvironment> cast_environment,
                          const AudioSenderConfig& audio_config,
                          CastTransportSender* const transport_sender)
     : FrameSender(
         cast_environment,
         true,
         transport_sender,
         base::TimeDelta::FromMilliseconds(audio_config.rtcp_interval),
         audio_config.frequency,
         audio_config.ssrc,
-        kAudioFrameRate * 2.0, // We lie to increase max outstanding frames.

[hubbe, 2014/09/18 17:56:05]

Does we still give more buffer to audio, or do you feel that that's not needed
now that we account for the RTT?

[miu, 2014/09/18 21:38:43]

Not needed.  According to the simulation data, the audio frame counts are also
significantly improved over the ToT implementation.

+        kAudioFrameRate,
         audio_config.min_playout_delay,
         audio_config.max_playout_delay,
         NewFixedCongestionControl(audio_config.bitrate)),
       samples_in_encoder_(0),
       weak_factory_(this) {
   cast_initialization_status_ = STATUS_AUDIO_UNINITIALIZED;
   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,
                          audio_config.channels,
                          audio_config.frequency,
                          audio_config.bitrate,
                          audio_config.codec,
                          base::Bind(&AudioSender::OnEncodedAudioFrame,
                                     weak_factory_.GetWeakPtr(),
                                     audio_config.bitrate)));
     cast_initialization_status_ = audio_encoder_->InitializationResult();
   } else {
     NOTREACHED();  // No support for external audio encoding.
     cast_initialization_status_ = STATUS_AUDIO_UNINITIALIZED;
   }
 
   media::cast::CastTransportRtpConfig transport_config;
   transport_config.ssrc = audio_config.ssrc;
   transport_config.feedback_ssrc = audio_config.incoming_feedback_ssrc;
   transport_config.rtp_payload_type = audio_config.rtp_payload_type;
-  transport_config.stored_frames =
-      std::min(kMaxUnackedFrames,
-               1 + static_cast<int>(max_playout_delay_ *
-                                    max_frame_rate_ /
-                                    base::TimeDelta::FromSeconds(1)));
   transport_config.aes_key = audio_config.aes_key;
   transport_config.aes_iv_mask = audio_config.aes_iv_mask;
 
   transport_sender->InitializeAudio(
       transport_config,
       base::Bind(&AudioSender::OnReceivedCastFeedback,
                  weak_factory_.GetWeakPtr()),
       base::Bind(&AudioSender::OnMeasuredRoundTripTime,
                  weak_factory_.GetWeakPtr()));
 }
 
 AudioSender::~AudioSender() {}
 
 void AudioSender::InsertAudio(scoped_ptr<AudioBus> audio_bus,
                               const base::TimeTicks& recorded_time) {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
   if (cast_initialization_status_ != STATUS_AUDIO_INITIALIZED) {
     NOTREACHED();
     return;
   }
   DCHECK(audio_encoder_.get()) << "Invalid internal state";
 
-  // TODO(miu): An |audio_bus| that represents more duration than a single
-  // frame's duration can defeat our logic here, causing too much data to become
-  // enqueued.  This will be addressed in a soon-upcoming change.
-  if (ShouldDropNextFrame(recorded_time)) {
-    VLOG(1) << "Dropping frame due to too many frames currently in-flight.";
+  const base::TimeDelta next_frame_duration =
+      RtpDeltaToTimeDelta(audio_bus->frames(), rtp_timebase());
+  if (ShouldDropNextFrame(next_frame_duration))
     return;
-  }
 
   samples_in_encoder_ += audio_bus->frames();
 
   audio_encoder_->InsertAudio(audio_bus.Pass(), recorded_time);
 }
 
 int AudioSender::GetNumberOfFramesInEncoder() const {
   // Note: It's possible for a partial frame to be in the encoder, but returning
   // the floor() is good enough for the "design limit" check in FrameSender.
   return samples_in_encoder_ / audio_encoder_->GetSamplesPerFrame();
 }
 
+base::TimeDelta AudioSender::GetInFlightMediaDuration() const {
+  const int samples_in_flight = samples_in_encoder_ +
+      GetUnacknowledgedFrameCount() * audio_encoder_->GetSamplesPerFrame();
+  return RtpDeltaToTimeDelta(samples_in_flight, rtp_timebase());
+}
+
 void AudioSender::OnAck(uint32 frame_id) {
 }
 
 void AudioSender::OnEncodedAudioFrame(
     int encoder_bitrate,
     scoped_ptr<EncodedFrame> encoded_frame,
     int samples_skipped) {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
 
   samples_in_encoder_ -= audio_encoder_->GetSamplesPerFrame() + samples_skipped;
   DCHECK_GE(samples_in_encoder_, 0);
 
   SendEncodedFrame(encoder_bitrate, encoded_frame.Pass());
 }
 
 }  // namespace cast
 }  // namespace media