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}
102 lines
3.4 KiB
C++
102 lines
3.4 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/video_engine/payload_router.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/system_wrappers/include/critical_section_wrapper.h"
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namespace webrtc {
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PayloadRouter::PayloadRouter()
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: crit_(CriticalSectionWrapper::CreateCriticalSection()),
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active_(false) {}
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PayloadRouter::~PayloadRouter() {}
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size_t PayloadRouter::DefaultMaxPayloadLength() {
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const size_t kIpUdpSrtpLength = 44;
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return IP_PACKET_SIZE - kIpUdpSrtpLength;
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}
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void PayloadRouter::SetSendingRtpModules(
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const std::list<RtpRtcp*>& rtp_modules) {
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CriticalSectionScoped cs(crit_.get());
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rtp_modules_.clear();
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rtp_modules_.reserve(rtp_modules.size());
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for (auto* rtp_module : rtp_modules) {
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rtp_modules_.push_back(rtp_module);
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}
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}
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void PayloadRouter::set_active(bool active) {
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CriticalSectionScoped cs(crit_.get());
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active_ = active;
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}
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bool PayloadRouter::active() {
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CriticalSectionScoped cs(crit_.get());
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return active_ && !rtp_modules_.empty();
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}
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bool PayloadRouter::RoutePayload(FrameType frame_type,
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int8_t payload_type,
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uint32_t time_stamp,
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int64_t capture_time_ms,
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const uint8_t* payload_data,
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size_t payload_length,
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const RTPFragmentationHeader* fragmentation,
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const RTPVideoHeader* rtp_video_hdr) {
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CriticalSectionScoped cs(crit_.get());
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if (!active_ || rtp_modules_.empty())
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return false;
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// The simulcast index might actually be larger than the number of modules in
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// case the encoder was processing a frame during a codec reconfig.
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if (rtp_video_hdr != NULL &&
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rtp_video_hdr->simulcastIdx >= rtp_modules_.size())
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return false;
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int stream_idx = 0;
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if (rtp_video_hdr != NULL)
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stream_idx = rtp_video_hdr->simulcastIdx;
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return rtp_modules_[stream_idx]->SendOutgoingData(
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frame_type, payload_type, time_stamp, capture_time_ms, payload_data,
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payload_length, fragmentation, rtp_video_hdr) == 0 ? true : false;
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}
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void PayloadRouter::SetTargetSendBitrates(
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const std::vector<uint32_t>& stream_bitrates) {
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CriticalSectionScoped cs(crit_.get());
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if (stream_bitrates.size() < rtp_modules_.size()) {
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// There can be a size mis-match during codec reconfiguration.
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return;
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}
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int idx = 0;
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for (auto* rtp_module : rtp_modules_) {
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rtp_module->SetTargetSendBitrate(stream_bitrates[idx++]);
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}
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}
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size_t PayloadRouter::MaxPayloadLength() const {
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size_t min_payload_length = DefaultMaxPayloadLength();
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CriticalSectionScoped cs(crit_.get());
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for (auto* rtp_module : rtp_modules_) {
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size_t module_payload_length = rtp_module->MaxDataPayloadLength();
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if (module_payload_length < min_payload_length)
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min_payload_length = module_payload_length;
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}
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return min_payload_length;
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}
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} // namespace webrtc
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