Cast: First stab at implementing adaptive latency

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 15 matching lines @@
                          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.
-        audio_config.target_playout_delay,
+        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;
-  // 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.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()));
 }
@@ skipping 38 matching lines @@
   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