| Index: webrtc/video/vie_remb.cc
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| diff --git a/webrtc/video/vie_remb.cc b/webrtc/video/vie_remb.cc
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| deleted file mode 100644
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| index be751016d1479fee16668e2b6c087c8ce60440b5..0000000000000000000000000000000000000000
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| --- a/webrtc/video/vie_remb.cc
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| +++ /dev/null
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| @@ -1,135 +0,0 @@
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| -/*
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| - *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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| - *
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| - *  Use of this source code is governed by a BSD-style license
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| - *  that can be found in the LICENSE file in the root of the source
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| - *  tree. An additional intellectual property rights grant can be found
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| - *  in the file PATENTS.  All contributing project authors may
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| - *  be found in the AUTHORS file in the root of the source tree.
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| - */
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| -
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| -#include "webrtc/video/vie_remb.h"
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| -
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| -#include <assert.h>
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| -
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| -#include <algorithm>
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| -
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| -#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
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| -#include "webrtc/modules/utility/include/process_thread.h"
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| -
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| -namespace webrtc {
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| -
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| -const int kRembSendIntervalMs = 200;
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| -
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| -// % threshold for if we should send a new REMB asap.
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| -const uint32_t kSendThresholdPercent = 97;
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| -
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| -VieRemb::VieRemb(Clock* clock)
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| -    : clock_(clock),
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| -      last_remb_time_(clock_->TimeInMilliseconds()),
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| -      last_send_bitrate_(0),
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| -      bitrate_(0) {}
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| -
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| -VieRemb::~VieRemb() {}
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| -
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| -void VieRemb::AddReceiveChannel(RtpRtcp* rtp_rtcp) {
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| -  assert(rtp_rtcp);
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| -
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| -  rtc::CritScope lock(&list_crit_);
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| -  if (std::find(receive_modules_.begin(), receive_modules_.end(), rtp_rtcp) !=
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| -      receive_modules_.end())
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| -    return;
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| -
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| -  // The module probably doesn't have a remote SSRC yet, so don't add it to the
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| -  // map.
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| -  receive_modules_.push_back(rtp_rtcp);
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| -}
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| -
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| -void VieRemb::RemoveReceiveChannel(RtpRtcp* rtp_rtcp) {
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| -  assert(rtp_rtcp);
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| -
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| -  rtc::CritScope lock(&list_crit_);
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| -  for (RtpModules::iterator it = receive_modules_.begin();
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| -       it != receive_modules_.end(); ++it) {
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| -    if ((*it) == rtp_rtcp) {
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| -      receive_modules_.erase(it);
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| -      break;
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| -    }
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| -  }
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| -}
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| -
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| -void VieRemb::AddRembSender(RtpRtcp* rtp_rtcp) {
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| -  assert(rtp_rtcp);
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| -
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| -  rtc::CritScope lock(&list_crit_);
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| -
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| -  // Verify this module hasn't been added earlier.
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| -  if (std::find(rtcp_sender_.begin(), rtcp_sender_.end(), rtp_rtcp) !=
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| -      rtcp_sender_.end())
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| -    return;
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| -  rtcp_sender_.push_back(rtp_rtcp);
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| -}
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| -
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| -void VieRemb::RemoveRembSender(RtpRtcp* rtp_rtcp) {
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| -  assert(rtp_rtcp);
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| -
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| -  rtc::CritScope lock(&list_crit_);
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| -  for (RtpModules::iterator it = rtcp_sender_.begin();
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| -       it != rtcp_sender_.end(); ++it) {
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| -    if ((*it) == rtp_rtcp) {
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| -      rtcp_sender_.erase(it);
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| -      return;
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| -    }
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| -  }
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| -}
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| -
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| -bool VieRemb::InUse() const {
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| -  rtc::CritScope lock(&list_crit_);
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| -  return !receive_modules_.empty() || !rtcp_sender_.empty();
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| -}
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| -
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| -void VieRemb::OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs,
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| -                                      uint32_t bitrate) {
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| -  RtpRtcp* sender = nullptr;
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| -  {
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| -    rtc::CritScope lock(&list_crit_);
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| -    // If we already have an estimate, check if the new total estimate is below
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| -    // kSendThresholdPercent of the previous estimate.
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| -    if (last_send_bitrate_ > 0) {
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| -      uint32_t new_remb_bitrate = last_send_bitrate_ - bitrate_ + bitrate;
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| -
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| -      if (new_remb_bitrate < kSendThresholdPercent * last_send_bitrate_ / 100) {
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| -        // The new bitrate estimate is less than kSendThresholdPercent % of the
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| -        // last report. Send a REMB asap.
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| -        last_remb_time_ = clock_->TimeInMilliseconds() - kRembSendIntervalMs;
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| -      }
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| -    }
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| -    bitrate_ = bitrate;
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| -
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| -    // Calculate total receive bitrate estimate.
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| -    int64_t now = clock_->TimeInMilliseconds();
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| -
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| -    if (now - last_remb_time_ < kRembSendIntervalMs) {
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| -      return;
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| -    }
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| -    last_remb_time_ = now;
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| -
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| -    if (ssrcs.empty() || receive_modules_.empty()) {
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| -      return;
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| -    }
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| -
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| -    // Send a REMB packet.
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| -    if (!rtcp_sender_.empty()) {
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| -      sender = rtcp_sender_.front();
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| -    } else {
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| -      sender = receive_modules_.front();
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| -    }
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| -    last_send_bitrate_ = bitrate_;
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| -  }
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| -
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| -  if (sender) {
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| -    sender->SetREMBData(bitrate_, ssrcs);
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| -  }
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| -}
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| -
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| -}  // namespace webrtc
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| 
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