e5ad5ca06a
Reason for revert: Intend to fix perf failures and reland. Original issue's description: > Revert of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #4 id:60001 of https://codereview.webrtc.org/2774463003/ ) > > Reason for revert: > Reverting since this seems to break multiple WebRTC Perf buildbots > > Original issue's description: > > Don't hardcode MediaType::ANY in FakeNetworkPipe. > > > > Instead let each test set the appropriate media type. This simplifies > > demuxing in Call and later in RtpTransportController. > > > > BUG=webrtc:7135 > > > > Review-Url: https://codereview.webrtc.org/2774463003 > > Cr-Commit-Position: refs/heads/master@{#17418} > > Committed:9c47b00e24> > TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,nisse@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7135 > > Review-Url: https://codereview.webrtc.org/2784543002 > Cr-Commit-Position: refs/heads/master@{#17427} > Committed:3a3bd50610TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,lliuu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/2783853002 Cr-Commit-Position: refs/heads/master@{#17459}
94 lines
2.8 KiB
C++
94 lines
2.8 KiB
C++
/*
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* Copyright (c) 2013 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/test/direct_transport.h"
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#include "webrtc/call/call.h"
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#include "webrtc/system_wrappers/include/clock.h"
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namespace webrtc {
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namespace test {
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DirectTransport::DirectTransport(Call* send_call, MediaType media_type)
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: DirectTransport(FakeNetworkPipe::Config(), send_call, media_type) {}
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DirectTransport::DirectTransport(const FakeNetworkPipe::Config& config,
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Call* send_call, MediaType media_type)
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: send_call_(send_call),
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packet_event_(false, false),
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thread_(NetworkProcess, this, "NetworkProcess"),
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clock_(Clock::GetRealTimeClock()),
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shutting_down_(false),
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fake_network_(clock_, config, media_type) {
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thread_.Start();
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if (send_call_) {
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send_call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
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send_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
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}
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}
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DirectTransport::~DirectTransport() { StopSending(); }
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void DirectTransport::SetConfig(const FakeNetworkPipe::Config& config) {
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fake_network_.SetConfig(config);
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}
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void DirectTransport::StopSending() {
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{
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rtc::CritScope crit(&lock_);
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shutting_down_ = true;
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}
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packet_event_.Set();
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thread_.Stop();
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}
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void DirectTransport::SetReceiver(PacketReceiver* receiver) {
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fake_network_.SetReceiver(receiver);
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}
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bool DirectTransport::SendRtp(const uint8_t* data,
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size_t length,
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const PacketOptions& options) {
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if (send_call_) {
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rtc::SentPacket sent_packet(options.packet_id,
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clock_->TimeInMilliseconds());
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send_call_->OnSentPacket(sent_packet);
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}
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fake_network_.SendPacket(data, length);
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packet_event_.Set();
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return true;
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}
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bool DirectTransport::SendRtcp(const uint8_t* data, size_t length) {
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fake_network_.SendPacket(data, length);
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packet_event_.Set();
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return true;
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}
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int DirectTransport::GetAverageDelayMs() {
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return fake_network_.AverageDelay();
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}
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bool DirectTransport::NetworkProcess(void* transport) {
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return static_cast<DirectTransport*>(transport)->SendPackets();
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}
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bool DirectTransport::SendPackets() {
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fake_network_.Process();
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int64_t wait_time_ms = fake_network_.TimeUntilNextProcess();
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if (wait_time_ms > 0) {
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packet_event_.Wait(static_cast<int>(wait_time_ms));
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}
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rtc::CritScope crit(&lock_);
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return shutting_down_ ? false : true;
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}
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} // namespace test
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} // namespace webrtc
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