3dd5d1d84a
Instead relies on SetSendingMediaStatus() to filter out receiving RTP modules. This status is now set in VoiceEngine's SetSend() for senders along with SetSendingStatus(). BUG= R=solenberg@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/1705763002 . Cr-Commit-Position: refs/heads/master@{#11768}
107 lines
3.6 KiB
C++
107 lines
3.6 KiB
C++
/*
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* Copyright (c) 2015 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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#include "webrtc/modules/pacing/packet_router.h"
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#include "webrtc/base/atomicops.h"
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#include "webrtc/base/checks.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
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namespace webrtc {
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PacketRouter::PacketRouter() : transport_seq_(0) {
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pacer_thread_checker_.DetachFromThread();
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}
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PacketRouter::~PacketRouter() {
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RTC_DCHECK(rtp_modules_.empty());
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}
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void PacketRouter::AddRtpModule(RtpRtcp* rtp_module) {
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rtc::CritScope cs(&modules_crit_);
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RTC_DCHECK(std::find(rtp_modules_.begin(), rtp_modules_.end(), rtp_module) ==
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rtp_modules_.end());
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rtp_modules_.push_back(rtp_module);
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}
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void PacketRouter::RemoveRtpModule(RtpRtcp* rtp_module) {
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rtc::CritScope cs(&modules_crit_);
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RTC_DCHECK(std::find(rtp_modules_.begin(), rtp_modules_.end(), rtp_module) !=
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rtp_modules_.end());
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rtp_modules_.remove(rtp_module);
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}
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bool PacketRouter::TimeToSendPacket(uint32_t ssrc,
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uint16_t sequence_number,
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int64_t capture_timestamp,
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bool retransmission) {
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RTC_DCHECK(pacer_thread_checker_.CalledOnValidThread());
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rtc::CritScope cs(&modules_crit_);
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for (auto* rtp_module : rtp_modules_) {
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if (rtp_module->SendingMedia() && ssrc == rtp_module->SSRC()) {
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return rtp_module->TimeToSendPacket(ssrc, sequence_number,
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capture_timestamp, retransmission);
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}
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}
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return true;
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}
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size_t PacketRouter::TimeToSendPadding(size_t bytes_to_send) {
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RTC_DCHECK(pacer_thread_checker_.CalledOnValidThread());
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size_t total_bytes_sent = 0;
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rtc::CritScope cs(&modules_crit_);
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for (RtpRtcp* module : rtp_modules_) {
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if (module->SendingMedia()) {
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size_t bytes_sent =
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module->TimeToSendPadding(bytes_to_send - total_bytes_sent);
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total_bytes_sent += bytes_sent;
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if (total_bytes_sent >= bytes_to_send)
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break;
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}
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}
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return total_bytes_sent;
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}
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void PacketRouter::SetTransportWideSequenceNumber(uint16_t sequence_number) {
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rtc::AtomicOps::ReleaseStore(&transport_seq_, sequence_number);
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}
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uint16_t PacketRouter::AllocateSequenceNumber() {
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int prev_seq = rtc::AtomicOps::AcquireLoad(&transport_seq_);
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int desired_prev_seq;
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int new_seq;
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do {
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desired_prev_seq = prev_seq;
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new_seq = (desired_prev_seq + 1) & 0xFFFF;
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// Note: CompareAndSwap returns the actual value of transport_seq at the
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// time the CAS operation was executed. Thus, if prev_seq is returned, the
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// operation was successful - otherwise we need to retry. Saving the
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// return value saves us a load on retry.
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prev_seq = rtc::AtomicOps::CompareAndSwap(&transport_seq_, desired_prev_seq,
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new_seq);
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} while (prev_seq != desired_prev_seq);
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return new_seq;
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}
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bool PacketRouter::SendFeedback(rtcp::TransportFeedback* packet) {
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rtc::CritScope cs(&modules_crit_);
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for (auto* rtp_module : rtp_modules_) {
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packet->WithPacketSenderSsrc(rtp_module->SSRC());
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if (rtp_module->SendFeedbackPacket(*packet))
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return true;
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}
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return false;
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}
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} // namespace webrtc
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