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}
125 lines
3.8 KiB
C++
125 lines
3.8 KiB
C++
/*
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* Copyright (c) 2012 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/video_engine/encoder_state_feedback.h"
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#include <assert.h>
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#include "webrtc/base/checks.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
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#include "webrtc/video_engine/vie_encoder.h"
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namespace webrtc {
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// Helper class registered at the RTP module relaying callbacks to
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// EncoderStatFeedback.
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class EncoderStateFeedbackObserver : public RtcpIntraFrameObserver {
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public:
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explicit EncoderStateFeedbackObserver(EncoderStateFeedback* owner)
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: owner_(owner) {}
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~EncoderStateFeedbackObserver() {}
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// Implements RtcpIntraFrameObserver.
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virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) {
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owner_->OnReceivedIntraFrameRequest(ssrc);
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}
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virtual void OnReceivedSLI(uint32_t ssrc, uint8_t picture_id) {
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owner_->OnReceivedSLI(ssrc, picture_id);
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}
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virtual void OnReceivedRPSI(uint32_t ssrc, uint64_t picture_id) {
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owner_->OnReceivedRPSI(ssrc, picture_id);
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}
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virtual void OnLocalSsrcChanged(uint32_t old_ssrc, uint32_t new_ssrc) {
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owner_->OnLocalSsrcChanged(old_ssrc, new_ssrc);
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}
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private:
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EncoderStateFeedback* owner_;
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};
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EncoderStateFeedback::EncoderStateFeedback()
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: crit_(CriticalSectionWrapper::CreateCriticalSection()),
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observer_(new EncoderStateFeedbackObserver(this)) {}
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EncoderStateFeedback::~EncoderStateFeedback() {
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assert(encoders_.empty());
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}
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void EncoderStateFeedback::AddEncoder(const std::vector<uint32_t>& ssrcs,
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ViEEncoder* encoder) {
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RTC_DCHECK(!ssrcs.empty());
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CriticalSectionScoped lock(crit_.get());
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for (uint32_t ssrc : ssrcs) {
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RTC_DCHECK(encoders_.find(ssrc) == encoders_.end());
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encoders_[ssrc] = encoder;
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}
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}
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void EncoderStateFeedback::RemoveEncoder(const ViEEncoder* encoder) {
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CriticalSectionScoped lock(crit_.get());
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SsrcEncoderMap::iterator it = encoders_.begin();
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while (it != encoders_.end()) {
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if (it->second == encoder) {
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encoders_.erase(it++);
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} else {
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++it;
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}
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}
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}
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RtcpIntraFrameObserver* EncoderStateFeedback::GetRtcpIntraFrameObserver() {
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return observer_.get();
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}
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void EncoderStateFeedback::OnReceivedIntraFrameRequest(uint32_t ssrc) {
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CriticalSectionScoped lock(crit_.get());
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SsrcEncoderMap::iterator it = encoders_.find(ssrc);
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if (it == encoders_.end())
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return;
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it->second->OnReceivedIntraFrameRequest(ssrc);
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}
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void EncoderStateFeedback::OnReceivedSLI(uint32_t ssrc, uint8_t picture_id) {
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CriticalSectionScoped lock(crit_.get());
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SsrcEncoderMap::iterator it = encoders_.find(ssrc);
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if (it == encoders_.end())
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return;
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it->second->OnReceivedSLI(ssrc, picture_id);
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}
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void EncoderStateFeedback::OnReceivedRPSI(uint32_t ssrc, uint64_t picture_id) {
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CriticalSectionScoped lock(crit_.get());
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SsrcEncoderMap::iterator it = encoders_.find(ssrc);
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if (it == encoders_.end())
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return;
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it->second->OnReceivedRPSI(ssrc, picture_id);
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}
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void EncoderStateFeedback::OnLocalSsrcChanged(uint32_t old_ssrc,
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uint32_t new_ssrc) {
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CriticalSectionScoped lock(crit_.get());
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SsrcEncoderMap::iterator it = encoders_.find(old_ssrc);
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if (it == encoders_.end() || encoders_.find(new_ssrc) != encoders_.end()) {
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return;
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}
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ViEEncoder* encoder = it->second;
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encoders_.erase(it);
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encoders_[new_ssrc] = encoder;
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encoder->OnLocalSsrcChanged(old_ssrc, new_ssrc);
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}
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} // namespace webrtc
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