Remove rtcp_interval from cast_config

media/cast/sender/frame_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/frame_sender.h"
 
 #include "base/trace_event/trace_event.h"
 
 namespace media {
 namespace cast {
 namespace {
 
 const int kMinSchedulingDelayMs = 1;
 const int kNumAggressiveReportsSentAtStart = 100;
 
 // The additional number of frames that can be in-flight when input exceeds the
 // maximum frame rate.
 const int kMaxFrameBurst = 5;
 
 }  // namespace
 
 // Convenience macro used in logging statements throughout this file.
 #define SENDER_SSRC (is_audio_ ? "AUDIO[" : "VIDEO[") << ssrc_ << "] "
 
 FrameSender::FrameSender(scoped_refptr<CastEnvironment> cast_environment,
                          bool is_audio,
                          CastTransportSender* const transport_sender,
-                         base::TimeDelta rtcp_interval,
                          int rtp_timebase,
                          uint32 ssrc,
                          double max_frame_rate,
                          base::TimeDelta min_playout_delay,
                          base::TimeDelta max_playout_delay,
                          CongestionControl* congestion_control)
     : cast_environment_(cast_environment),
       transport_sender_(transport_sender),
       ssrc_(ssrc),
-      rtcp_interval_(rtcp_interval),
       min_playout_delay_(min_playout_delay == base::TimeDelta() ?
                          max_playout_delay : min_playout_delay),
       max_playout_delay_(max_playout_delay),
       send_target_playout_delay_(false),
       max_frame_rate_(max_frame_rate),
       num_aggressive_rtcp_reports_sent_(0),
       last_sent_frame_id_(0),
       latest_acked_frame_id_(0),
       duplicate_ack_counter_(0),
       congestion_control_(congestion_control),
       rtp_timebase_(rtp_timebase),
       is_audio_(is_audio),
       weak_factory_(this) {
   DCHECK(transport_sender_);
   DCHECK_GT(rtp_timebase_, 0);
   DCHECK(congestion_control_);
   SetTargetPlayoutDelay(min_playout_delay_);
   send_target_playout_delay_ = false;
   memset(frame_rtp_timestamps_, 0, sizeof(frame_rtp_timestamps_));
 }
 
 FrameSender::~FrameSender() {
 }
 
 void FrameSender::ScheduleNextRtcpReport() {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
-  base::TimeDelta time_to_next = rtcp_interval_;
-
-  time_to_next = std::max(
-      time_to_next, base::TimeDelta::FromMilliseconds(kMinSchedulingDelayMs));
 
   cast_environment_->PostDelayedTask(
-      CastEnvironment::MAIN,
-      FROM_HERE,
+      CastEnvironment::MAIN, FROM_HERE,
       base::Bind(&FrameSender::SendRtcpReport, weak_factory_.GetWeakPtr(),
                  true),
-      time_to_next);
+      base::TimeDelta::FromMilliseconds(kDefaultRtcpIntervalMs));
 }
 
 void FrameSender::SendRtcpReport(bool schedule_future_reports) {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
 
   // Sanity-check: We should have sent at least the first frame by this point.
   DCHECK(!last_send_time_.is_null());
 
   // Create lip-sync info for the sender report.  The last sent frame's
   // reference time and RTP timestamp are used to estimate an RTP timestamp in
@@ skipping 291 matching lines @@
     VLOG(1) << SENDER_SSRC << "Dropping: In-flight duration would be too high.";
     return true;
   }
 
   // Next frame is accepted.
   return false;
 }
 
 }  // namespace cast
 }  // namespace media