Move common code and variables from audio/video sender to frame sender.

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"
 #include "media/cast/net/cast_transport_config.h"
 #include "media/cast/sender/audio_encoder.h"
 
 namespace media {
 namespace cast {
 namespace {
 
 const int kNumAggressiveReportsSentAtStart = 100;
-const int kMinSchedulingDelayMs = 1;
 
 // 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;
 
 }  // namespace
 
 AudioSender::AudioSender(scoped_refptr<CastEnvironment> cast_environment,
                          const AudioSenderConfig& audio_config,
                          CastTransportSender* const transport_sender)
     : FrameSender(
         cast_environment,
         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),
       configured_encoder_bitrate_(audio_config.bitrate),
-      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) {
+  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,
@@ skipping 13 matching lines @@
   // 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,
       base::Bind(&AudioSender::OnReceivedCastFeedback,
                  weak_factory_.GetWeakPtr()),
       base::Bind(&AudioSender::OnReceivedRtt, weak_factory_.GetWeakPtr()));
-  memset(frame_id_to_rtp_timestamp_, 0, sizeof(frame_id_to_rtp_timestamp_));
 }
 
 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;
@@ skipping 52 matching lines @@
     SendRtcpReport(is_last_aggressive_report);
   }
 
   if (send_target_playout_delay_) {
     encoded_frame->new_playout_delay_ms =
         target_playout_delay_.InMilliseconds();
   }
   transport_sender_->InsertCodedAudioFrame(*encoded_frame);
 }
 
-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));
 
   if (is_rtt_available()) {
     // Having the RTT values implies the receiver sent back a receiver report
     // based on it having received a report from here.  Therefore, ensure this
     // sender stops aggressively sending reports.
     if (num_aggressive_rtcp_reports_sent_ < kNumAggressiveReportsSentAtStart) {
       VLOG(1) << "No longer a need to send reports aggressively (sent "
               << num_aggressive_rtcp_reports_sent_ << ").";
@@ skipping 57 matching lines @@
   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.";
-  last_send_time_ = cast_environment_->Clock()->NowTicks();
-  transport_sender_->ResendFrameForKickstart(ssrc_, last_sent_frame_id_);
-}
-
 }  // namespace cast
 }  // namespace media