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Diffstat (limited to 'webrtc/video/vie_remb.cc')
-rw-r--r-- | webrtc/video/vie_remb.cc | 144 |
1 files changed, 144 insertions, 0 deletions
diff --git a/webrtc/video/vie_remb.cc b/webrtc/video/vie_remb.cc new file mode 100644 index 0000000000..95c2f1e130 --- /dev/null +++ b/webrtc/video/vie_remb.cc @@ -0,0 +1,144 @@ +/* + * 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(Clock* clock) + : clock_(clock), + list_crit_(CriticalSectionWrapper::CreateCriticalSection()), + last_remb_time_(clock_->TimeInMilliseconds()), + 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_ = clock_->TimeInMilliseconds() - kRembSendIntervalMs; + } + } + bitrate_ = bitrate; + + // Calculate total receive bitrate estimate. + int64_t now = clock_->TimeInMilliseconds(); + + 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 |