290cb56dca
Thie CL moves the default RTP module logic for TimeToSendPacket and TimeToSendPadding to PayloadRouter class and asserts on usage of the default module. BUG=769 TEST=New unittest. R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/33319004 Cr-Commit-Position: refs/heads/master@{#8383} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8383 4adac7df-926f-26a2-2b94-8c16560cd09d
109 lines
3.6 KiB
C++
109 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/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_;
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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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DCHECK(rtp_video_hdr == NULL ||
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rtp_video_hdr->simulcastIdx <= rtp_modules_.size());
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if (!active_ || rtp_modules_.empty())
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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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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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