1 files changed, 0 insertions, 143 deletions
diff --git a/webrtc/video_engine/vie_remb.cc b/webrtc/video_engine/vie_remb.cc
deleted file mode 100644
index b347f2ee00..0000000000
--- a/webrtc/video_engine/vie_remb.cc
+++ /dev/null
@@ -1,143 +0,0 @@
-/*
- *  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.
- */
-
-#include "webrtc/video_engine/vie_remb.h"
-
-#include <assert.h>
-
-#include <algorithm>
-
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
-#include "webrtc/system_wrappers/include/trace.h"
-
-namespace webrtc {
-
-const int kRembSendIntervalMs = 200;
-
-// % threshold for if we should send a new REMB asap.
-const unsigned int kSendThresholdPercent = 97;
-
-VieRemb::VieRemb()
-    : list_crit_(CriticalSectionWrapper::CreateCriticalSection()),
-      last_remb_time_(TickTime::MillisecondTimestamp()),
-      last_send_bitrate_(0),
-      bitrate_(0) {}
-
-VieRemb::~VieRemb() {}
-
-void VieRemb::AddReceiveChannel(RtpRtcp* rtp_rtcp) {
-  assert(rtp_rtcp);
-
-  CriticalSectionScoped cs(list_crit_.get());
-  if (std::find(receive_modules_.begin(), receive_modules_.end(), rtp_rtcp) !=
-      receive_modules_.end())
-    return;
-
-  // The module probably doesn't have a remote SSRC yet, so don't add it to the
-  // map.
-  receive_modules_.push_back(rtp_rtcp);
-}
-
-void VieRemb::RemoveReceiveChannel(RtpRtcp* rtp_rtcp) {
-  assert(rtp_rtcp);
-
-  CriticalSectionScoped cs(list_crit_.get());
-  for (RtpModules::iterator it = receive_modules_.begin();
-       it != receive_modules_.end(); ++it) {
-    if ((*it) == rtp_rtcp) {
-      receive_modules_.erase(it);
-      break;
-    }
-  }
-}
-
-void VieRemb::AddRembSender(RtpRtcp* rtp_rtcp) {
-  assert(rtp_rtcp);
-
-  CriticalSectionScoped cs(list_crit_.get());
-
-  // Verify this module hasn't been added earlier.
-  if (std::find(rtcp_sender_.begin(), rtcp_sender_.end(), rtp_rtcp) !=
-      rtcp_sender_.end())
-    return;
-  rtcp_sender_.push_back(rtp_rtcp);
-}
-
-void VieRemb::RemoveRembSender(RtpRtcp* rtp_rtcp) {
-  assert(rtp_rtcp);
-
-  CriticalSectionScoped cs(list_crit_.get());
-  for (RtpModules::iterator it = rtcp_sender_.begin();
-       it != rtcp_sender_.end(); ++it) {
-    if ((*it) == rtp_rtcp) {
-      rtcp_sender_.erase(it);
-      return;
-    }
-  }
-}
-
-bool VieRemb::InUse() const {
-  CriticalSectionScoped cs(list_crit_.get());
-  if (receive_modules_.empty() && rtcp_sender_.empty())
-    return false;
-  else
-    return true;
-}
-
-void VieRemb::OnReceiveBitrateChanged(const std::vector<unsigned int>& ssrcs,
-                                      unsigned int bitrate) {
-  list_crit_->Enter();
-  // If we already have an estimate, check if the new total estimate is below
-  // kSendThresholdPercent of the previous estimate.
-  if (last_send_bitrate_ > 0) {
-    unsigned int new_remb_bitrate = last_send_bitrate_ - bitrate_ + bitrate;
-
-    if (new_remb_bitrate < kSendThresholdPercent * last_send_bitrate_ / 100) {
-      // The new bitrate estimate is less than kSendThresholdPercent % of the
-      // last report. Send a REMB asap.
-      last_remb_time_ = TickTime::MillisecondTimestamp() - kRembSendIntervalMs;
-    }
-  }
-  bitrate_ = bitrate;
-
-  // Calculate total receive bitrate estimate.
-  int64_t now = TickTime::MillisecondTimestamp();
-
-  if (now - last_remb_time_ < kRembSendIntervalMs) {
-    list_crit_->Leave();
-    return;
-  }
-  last_remb_time_ = now;
-
-  if (ssrcs.empty() || receive_modules_.empty()) {
-    list_crit_->Leave();
-    return;
-  }
-
-  // Send a REMB packet.
-  RtpRtcp* sender = NULL;
-  if (!rtcp_sender_.empty()) {
-    sender = rtcp_sender_.front();
-  } else {
-    sender = receive_modules_.front();
-  }
-  last_send_bitrate_ = bitrate_;
-
-  list_crit_->Leave();
-
-  if (sender) {
-    sender->SetREMBData(bitrate_, ssrcs);
-  }
-}
-
-}  // namespace webrtc