ff761fba82
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
104 lines
3.5 KiB
C++
104 lines
3.5 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/include/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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}
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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_lock_);
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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_lock_);
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auto it = std::find(rtp_modules_.begin(), rtp_modules_.end(), rtp_module);
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RTC_DCHECK(it != rtp_modules_.end());
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rtp_modules_.erase(it);
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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::CritScope cs(&modules_lock_);
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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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size_t total_bytes_sent = 0;
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rtc::CritScope cs(&modules_lock_);
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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_lock_);
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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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