Delete a chain of methods in ViE, VoE and ACM

webrtc/video_engine/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 22 matching lines @@
 #include "webrtc/video/receive_statistics_proxy.h"
 #include "webrtc/video_engine/call_stats.h"
 #include "webrtc/video_engine/payload_router.h"
 #include "webrtc/video_engine/report_block_stats.h"
 #include "webrtc/video_engine/vie_defines.h"
 
 namespace webrtc {
 
 const int kMaxDecodeWaitTimeMs = 50;
 static const int kMaxTargetDelayMs = 10000;
-static const float kMaxIncompleteTimeMultiplier = 3.5f;
 
 // Helper class receiving statistics callbacks.
 class ChannelStatsObserver : public CallStatsObserver {
  public:
   explicit ChannelStatsObserver(ViEChannel* owner) : owner_(owner) {}
   virtual ~ChannelStatsObserver() {}
 
   // Implements StatsObserver.
   virtual void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) {
     owner_->OnRttUpdate(avg_rtt_ms, max_rtt_ms);
@@ skipping 514 matching lines @@
     // Don't allow a number lower than the default value.
     if (nack_history_size_sender_ < kSendSidePacketHistorySize) {
       nack_history_size_sender_ = kSendSidePacketHistorySize;
     }
   }
   for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
     rtp_rtcp->SetStorePacketsStatus(true, nack_history_size_sender_);
   return 0;
 }
 
-int ViEChannel::SetReceiverBufferingMode(int target_delay_ms) {
-  if ((target_delay_ms < 0) || (target_delay_ms > kMaxTargetDelayMs)) {
-    LOG(LS_ERROR) << "Invalid receive buffer delay value.";
-    return -1;
-  }
-  int max_nack_list_size;
-  int max_incomplete_time_ms;
-  if (target_delay_ms == 0) {
-    // Real-time mode - restore default settings.
-    max_nack_reordering_threshold_ = kMaxPacketAgeToNack;
-    max_nack_list_size = kMaxNackListSize;
-    max_incomplete_time_ms = 0;
-  } else {
-    max_nack_list_size =  3 * GetRequiredNackListSize(target_delay_ms) / 4;
-    max_nack_reordering_threshold_ = max_nack_list_size;
-    // Calculate the max incomplete time and round to int.
-    max_incomplete_time_ms = static_cast<int>(kMaxIncompleteTimeMultiplier *
-        target_delay_ms + 0.5f);
-  }
-  vcm_->SetNackSettings(max_nack_list_size, max_nack_reordering_threshold_,
-                       max_incomplete_time_ms);
-  vcm_->SetMinReceiverDelay(target_delay_ms);
-  if (vie_sync_.SetTargetBufferingDelay(target_delay_ms) < 0)
-    return -1;
-  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 629 matching lines @@
   CriticalSectionScoped cs(crit_.get());
   receive_stats_callback_ = receive_statistics_proxy;
 }
 
 void ViEChannel::SetIncomingVideoStream(
     IncomingVideoStream* incoming_video_stream) {
   CriticalSectionScoped cs(crit_.get());
   incoming_video_stream_ = incoming_video_stream;
 }
 }  // namespace webrtc