Nuke SetSenderBufferingMode.

webrtc/video/vie_channel.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 17 matching lines @@
 #include "webrtc/modules/video_coding/include/video_coding.h"
 #include "webrtc/modules/video_processing/include/video_processing.h"
 #include "webrtc/modules/video_render/video_render_defines.h"
 #include "webrtc/system_wrappers/include/metrics.h"
 #include "webrtc/video/call_stats.h"
 #include "webrtc/video/payload_router.h"
 #include "webrtc/video/receive_statistics_proxy.h"
 
 namespace webrtc {
 
-static const int kMaxTargetDelayMs = 10000;
 const int kMinSendSidePacketHistorySize = 600;
 const int kMaxPacketAgeToNack = 450;
 const int kMaxNackListSize = 250;
 
 // Helper class receiving statistics callbacks.
 class ChannelStatsObserver : public CallStatsObserver {
  public:
   explicit ChannelStatsObserver(ViEChannel* owner) : owner_(owner) {}
   virtual ~ChannelStatsObserver() {}
 
@@ skipping 411 matching lines @@
   uint8_t pltype_fec = 0;
 
   for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
     rtp_rtcp->GenericFECStatus(&fec_enabled, &pltype_red, &pltype_fec);
     if (fec_enabled)
       return true;
   }
   return false;
 }
 
-int ViEChannel::SetSenderBufferingMode(int target_delay_ms) {
-  if ((target_delay_ms < 0) || (target_delay_ms > kMaxTargetDelayMs)) {
-    LOG(LS_ERROR) << "Invalid send buffer value.";
-    return -1;
-  }
-  if (target_delay_ms == 0) {
-    // Real-time mode.
-    nack_history_size_sender_ = kMinSendSidePacketHistorySize;
-  } else {
-    nack_history_size_sender_ = GetRequiredNackListSize(target_delay_ms);
-    // Don't allow a number lower than the default value.
-    if (nack_history_size_sender_ < kMinSendSidePacketHistorySize) {
-      nack_history_size_sender_ = kMinSendSidePacketHistorySize;
-    }
-  }
-  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
-    rtp_rtcp->SetStorePacketsStatus(true, nack_history_size_sender_);
-  return 0;
-}
-
 int ViEChannel::GetRequiredNackListSize(int target_delay_ms) {
   // The max size of the nack list should be large enough to accommodate the
   // the number of packets (frames) resulting from the increased delay.
   // Roughly estimating for ~40 packets per frame @ 30fps.
   return target_delay_ms * 40 * 30 / 1000;
 }
 
 int ViEChannel::SetSendTimestampOffsetStatus(bool enable, int id) {
   // Disable any previous registrations of this extension to avoid errors.
   for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
@@ skipping 459 matching lines @@
   rtc::CritScope lock(&crit_);
   receive_stats_callback_ = receive_statistics_proxy;
 }
 
 void ViEChannel::SetIncomingVideoStream(
     IncomingVideoStream* incoming_video_stream) {
   rtc::CritScope lock(&crit_);
   incoming_video_stream_ = incoming_video_stream;
 }
 }  // namespace webrtc