| /* |
| * Copyright (c) 2015 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/modules/pacing/packet_router.h" |
| |
| #include "webrtc/base/atomicops.h" |
| #include "webrtc/base/checks.h" |
| #include "webrtc/base/timeutils.h" |
| #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
| #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" |
| |
| namespace webrtc { |
| |
| PacketRouter::PacketRouter() |
| : last_remb_time_ms_(rtc::TimeMillis()), |
| last_send_bitrate_bps_(0), |
| transport_seq_(0) { |
| pacer_thread_checker_.DetachFromThread(); |
| } |
| |
| PacketRouter::~PacketRouter() { |
| RTC_DCHECK(rtp_send_modules_.empty()); |
| RTC_DCHECK(rtp_receive_modules_.empty()); |
| } |
| |
| void PacketRouter::AddSendRtpModule(RtpRtcp* rtp_module) { |
| rtc::CritScope cs(&modules_crit_); |
| RTC_DCHECK(std::find(rtp_send_modules_.begin(), rtp_send_modules_.end(), |
| rtp_module) == rtp_send_modules_.end()); |
| if (rtp_send_modules_.empty() && !rtp_receive_modules_.empty()) { |
| rtp_receive_modules_.front()->SetREMBStatus(false); |
| } |
| |
| // Put modules which can use regular payload packets (over rtx) instead of |
| // padding first as it's less of a waste |
| if ((rtp_module->RtxSendStatus() & kRtxRedundantPayloads) > 0) { |
| if (!rtp_send_modules_.empty()) { |
| rtp_send_modules_.front()->SetREMBStatus(false); |
| } |
| rtp_send_modules_.push_front(rtp_module); |
| rtp_module->SetREMBStatus(true); |
| } else { |
| if (rtp_send_modules_.empty()) { |
| rtp_module->SetREMBStatus(true); |
| } |
| |
| rtp_send_modules_.push_back(rtp_module); |
| } |
| } |
| |
| void PacketRouter::RemoveSendRtpModule(RtpRtcp* rtp_module) { |
| rtc::CritScope cs(&modules_crit_); |
| RTC_DCHECK(std::find(rtp_send_modules_.begin(), rtp_send_modules_.end(), |
| rtp_module) != rtp_send_modules_.end()); |
| rtp_send_modules_.remove(rtp_module); |
| rtp_module->SetREMBStatus(false); |
| if (!rtp_send_modules_.empty()) { |
| rtp_send_modules_.front()->SetREMBStatus(true); |
| } else if (!rtp_receive_modules_.empty()) { |
| rtp_receive_modules_.front()->SetREMBStatus(true); |
| } |
| } |
| |
| void PacketRouter::AddReceiveRtpModule(RtpRtcp* rtp_module) { |
| rtc::CritScope cs(&modules_crit_); |
| RTC_DCHECK(std::find(rtp_receive_modules_.begin(), rtp_receive_modules_.end(), |
| rtp_module) == rtp_receive_modules_.end()); |
| if (rtp_send_modules_.empty() && rtp_receive_modules_.empty()) { |
| rtp_module->SetREMBStatus(true); |
| } |
| rtp_receive_modules_.push_back(rtp_module); |
| } |
| |
| void PacketRouter::RemoveReceiveRtpModule(RtpRtcp* rtp_module) { |
| rtc::CritScope cs(&modules_crit_); |
| const auto& it = std::find(rtp_receive_modules_.begin(), |
| rtp_receive_modules_.end(), rtp_module); |
| RTC_DCHECK(it != rtp_receive_modules_.end()); |
| rtp_receive_modules_.erase(it); |
| if (rtp_send_modules_.empty()) { |
| rtp_module->SetREMBStatus(false); |
| if (!rtp_receive_modules_.empty()) { |
| rtp_receive_modules_.front()->SetREMBStatus(true); |
| } |
| } |
| } |
| |
| bool PacketRouter::TimeToSendPacket(uint32_t ssrc, |
| uint16_t sequence_number, |
| int64_t capture_timestamp, |
| bool retransmission, |
| const PacedPacketInfo& pacing_info) { |
| RTC_DCHECK(pacer_thread_checker_.CalledOnValidThread()); |
| rtc::CritScope cs(&modules_crit_); |
| for (auto* rtp_module : rtp_send_modules_) { |
| if (!rtp_module->SendingMedia()) |
| continue; |
| if (ssrc == rtp_module->SSRC() || ssrc == rtp_module->FlexfecSsrc()) { |
| return rtp_module->TimeToSendPacket(ssrc, sequence_number, |
| capture_timestamp, retransmission, |
| pacing_info); |
| } |
| } |
| return true; |
| } |
| |
| size_t PacketRouter::TimeToSendPadding(size_t bytes_to_send, |
| const PacedPacketInfo& pacing_info) { |
| RTC_DCHECK(pacer_thread_checker_.CalledOnValidThread()); |
| size_t total_bytes_sent = 0; |
| rtc::CritScope cs(&modules_crit_); |
| // Rtp modules are ordered by which stream can most benefit from padding. |
| for (RtpRtcp* module : rtp_send_modules_) { |
| if (module->SendingMedia() && module->HasBweExtensions()) { |
| size_t bytes_sent = module->TimeToSendPadding( |
| bytes_to_send - total_bytes_sent, pacing_info); |
| total_bytes_sent += bytes_sent; |
| if (total_bytes_sent >= bytes_to_send) |
| break; |
| } |
| } |
| return total_bytes_sent; |
| } |
| |
| void PacketRouter::SetTransportWideSequenceNumber(uint16_t sequence_number) { |
| rtc::AtomicOps::ReleaseStore(&transport_seq_, sequence_number); |
| } |
| |
| uint16_t PacketRouter::AllocateSequenceNumber() { |
| int prev_seq = rtc::AtomicOps::AcquireLoad(&transport_seq_); |
| int desired_prev_seq; |
| int new_seq; |
| do { |
| desired_prev_seq = prev_seq; |
| new_seq = (desired_prev_seq + 1) & 0xFFFF; |
| // Note: CompareAndSwap returns the actual value of transport_seq at the |
| // time the CAS operation was executed. Thus, if prev_seq is returned, the |
| // operation was successful - otherwise we need to retry. Saving the |
| // return value saves us a load on retry. |
| prev_seq = rtc::AtomicOps::CompareAndSwap(&transport_seq_, desired_prev_seq, |
| new_seq); |
| } while (prev_seq != desired_prev_seq); |
| |
| return new_seq; |
| } |
| |
| void PacketRouter::OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs, |
| uint32_t bitrate_bps) { |
| const int kRembSendIntervalMs = 200; |
| |
| // % threshold for if we should send a new REMB asap. |
| const uint32_t kSendThresholdPercent = 97; |
| |
| int64_t now_ms = rtc::TimeMillis(); |
| { |
| rtc::CritScope lock(&remb_crit_); |
| |
| // If we already have an estimate, check if the new total estimate is below |
| // kSendThresholdPercent of the previous estimate. |
| if (last_send_bitrate_bps_ > 0) { |
| uint32_t new_remb_bitrate_bps = |
| last_send_bitrate_bps_ - bitrate_bps_ + bitrate_bps; |
| |
| if (new_remb_bitrate_bps < |
| kSendThresholdPercent * last_send_bitrate_bps_ / 100) { |
| // The new bitrate estimate is less than kSendThresholdPercent % of the |
| // last report. Send a REMB asap. |
| last_remb_time_ms_ = now_ms - kRembSendIntervalMs; |
| } |
| } |
| bitrate_bps_ = bitrate_bps; |
| |
| if (now_ms - last_remb_time_ms_ < kRembSendIntervalMs) { |
| return; |
| } |
| // NOTE: Updated if we intend to send the data; we might not have |
| // a module to actually send it. |
| last_remb_time_ms_ = now_ms; |
| last_send_bitrate_bps_ = bitrate_bps; |
| } |
| SendRemb(bitrate_bps, ssrcs); |
| } |
| |
| bool PacketRouter::SendRemb(uint32_t bitrate_bps, |
| const std::vector<uint32_t>& ssrcs) { |
| rtc::CritScope lock(&modules_crit_); |
| RtpRtcp* remb_module; |
| if (!rtp_send_modules_.empty()) |
| remb_module = rtp_send_modules_.front(); |
| else if (!rtp_receive_modules_.empty()) |
| remb_module = rtp_receive_modules_.front(); |
| else |
| return false; |
| // The Add* and Remove* methods above ensure that this (and only this) module |
| // has REMB enabled. REMB should be disabled on all other modules, because |
| // otherwise, they will send REMB with stale info. |
| RTC_DCHECK(remb_module->REMB()); |
| remb_module->SetREMBData(bitrate_bps, ssrcs); |
| return true; |
| } |
| |
| bool PacketRouter::SendTransportFeedback(rtcp::TransportFeedback* packet) { |
| RTC_DCHECK(pacer_thread_checker_.CalledOnValidThread()); |
| rtc::CritScope cs(&modules_crit_); |
| // Prefer send modules. |
| for (auto* rtp_module : rtp_send_modules_) { |
| packet->SetSenderSsrc(rtp_module->SSRC()); |
| if (rtp_module->SendFeedbackPacket(*packet)) |
| return true; |
| } |
| for (auto* rtp_module : rtp_receive_modules_) { |
| packet->SetSenderSsrc(rtp_module->SSRC()); |
| if (rtp_module->SendFeedbackPacket(*packet)) |
| return true; |
| } |
| return false; |
| } |
| |
| } // namespace webrtc |
