Index: webrtc/video/vie_remb.cc |
diff --git a/webrtc/video/vie_remb.cc b/webrtc/video/vie_remb.cc |
deleted file mode 100644 |
index 923e55c58c483a6214aeb4ee0778279f8d2a3983..0000000000000000000000000000000000000000 |
--- a/webrtc/video/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/vie_remb.h" |
- |
-#include <assert.h> |
- |
-#include <algorithm> |
- |
-#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
-#include "webrtc/modules/utility/include/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 |