50e28166af
This cl just moves the logic form the default module SetTargetSendBitrates to PayloadRouter. There might be glitch / mismatch in size between trate the vector and rtp modules. This was the same in the default module and is quite hard to protect from before we have the new video API. I also removed some test form rtp_rtcp_impl_unittest that were affected by this change. The test tests code that isn't implemented, hence the DISABLED_, and this will never be implemented in the RTP module, rather the payload router in the future. BUG=769 R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/42419004 Cr-Commit-Position: refs/heads/master@{#8453} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8453 4adac7df-926f-26a2-2b94-8c16560cd09d
125 lines
4.2 KiB
C++
125 lines
4.2 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/interface/rtp_rtcp.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
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#include "webrtc/system_wrappers/interface/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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bool PayloadRouter::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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CriticalSectionScoped cs(crit_.get());
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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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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::TimeToSendPadding(size_t bytes) {
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CriticalSectionScoped cs(crit_.get());
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for(auto* rtp_module : rtp_modules_) {
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if (rtp_module->SendingMedia())
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return rtp_module->TimeToSendPadding(bytes);
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}
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return 0;
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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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