Revert "Adding a test framework for conference mode application in VoE."
This reverts commit fc052055e939fa93d3ab92914e0dc8ed5e5d1d90. since it was not committed correctly. I committed it from a wrong machine, which did not have the correct patch. BUG= TBR=phoglund@webrtc.org, Review URL: https://webrtc-codereview.appspot.com/56469005 Cr-Commit-Position: refs/heads/master@{#9289}
This commit is contained in:
Minyue
parent
6a1ba8c17f
commit
a4b7e5e35a
@ -1,275 +0,0 @@
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/*
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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/voice_engine/test/auto_test/fakes/conference_transport.h"
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#include <string>
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#include "webrtc/base/byteorder.h"
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#include "webrtc/base/timeutils.h"
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#include "webrtc/test/testsupport/fileutils.h"
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#include "webrtc/system_wrappers/interface/sleep.h"
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namespace {
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static const unsigned int kReflectorSsrc = 0x0000;
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static const unsigned int kLocalSsrc = 0x0001;
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static const unsigned int kFirstRemoteSsrc = 0x0002;
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static const std::string kInputFileName =
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webrtc::test::ResourcePath("voice_engine/audio_long16", "pcm");
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static const webrtc::FileFormats kInputFileFormat =
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webrtc::kFileFormatPcm16kHzFile;
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static const webrtc::CodecInst kCodecInst =
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{120, "opus", 48000, 960, 2, 64000};
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static unsigned int ParseSsrc(const void* data, size_t len, bool rtcp) {
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const size_t ssrc_pos = (!rtcp) ? 8 : 4;
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unsigned int ssrc = 0;
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if (len >= (ssrc_pos + sizeof(ssrc))) {
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ssrc = rtc::GetBE32(static_cast<const char*>(data) + ssrc_pos);
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}
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return ssrc;
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}
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} // namespace
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namespace voetest {
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ConferenceTransport::ConferenceTransport()
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: pq_crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()),
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stream_crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()),
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packet_event_(webrtc::EventWrapper::Create()),
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thread_(webrtc::ThreadWrapper::CreateThread(Run,
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this,
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"ConferenceTransport")),
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rtt_ms_(0),
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stream_count_(0) {
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local_voe_ = webrtc::VoiceEngine::Create();
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local_base_ = webrtc::VoEBase::GetInterface(local_voe_);
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local_network_ = webrtc::VoENetwork::GetInterface(local_voe_);
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local_rtp_rtcp_ = webrtc::VoERTP_RTCP::GetInterface(local_voe_);
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// In principle, we can use one VoiceEngine to achieve the same goal. Well, in
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// here, we use two engines to make it more like reality.
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remote_voe_ = webrtc::VoiceEngine::Create();
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remote_base_ = webrtc::VoEBase::GetInterface(remote_voe_);
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remote_codec_ = webrtc::VoECodec::GetInterface(remote_voe_);
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remote_network_ = webrtc::VoENetwork::GetInterface(remote_voe_);
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remote_rtp_rtcp_ = webrtc::VoERTP_RTCP::GetInterface(remote_voe_);
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remote_file_ = webrtc::VoEFile::GetInterface(remote_voe_);
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EXPECT_EQ(0, local_base_->Init());
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local_sender_ = local_base_->CreateChannel();
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EXPECT_EQ(0, local_network_->RegisterExternalTransport(local_sender_, *this));
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EXPECT_EQ(0, local_rtp_rtcp_->SetLocalSSRC(local_sender_, kLocalSsrc));
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EXPECT_EQ(0, local_base_->StartSend(local_sender_));
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EXPECT_EQ(0, remote_base_->Init());
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reflector_ = remote_base_->CreateChannel();
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EXPECT_EQ(0, remote_network_->RegisterExternalTransport(reflector_, *this));
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EXPECT_EQ(0, remote_rtp_rtcp_->SetLocalSSRC(reflector_, kReflectorSsrc));
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thread_->Start();
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thread_->SetPriority(webrtc::kHighPriority);
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}
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ConferenceTransport::~ConferenceTransport() {
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// Must stop sending, otherwise DispatchPackets() cannot quit.
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EXPECT_EQ(0, remote_network_->DeRegisterExternalTransport(reflector_));
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EXPECT_EQ(0, local_network_->DeRegisterExternalTransport(local_sender_));
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for (auto stream : streams_) {
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RemoveStream(stream.first);
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}
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EXPECT_TRUE(thread_->Stop());
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remote_file_->Release();
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remote_rtp_rtcp_->Release();
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remote_network_->Release();
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remote_base_->Release();
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local_rtp_rtcp_->Release();
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local_network_->Release();
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local_base_->Release();
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EXPECT_TRUE(webrtc::VoiceEngine::Delete(remote_voe_));
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EXPECT_TRUE(webrtc::VoiceEngine::Delete(local_voe_));
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}
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int ConferenceTransport::SendPacket(int channel, const void* data, size_t len) {
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StorePacket(Packet::Rtp, channel, data, len);
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return static_cast<int>(len);
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}
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int ConferenceTransport::SendRTCPPacket(int channel, const void* data,
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size_t len) {
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StorePacket(Packet::Rtcp, channel, data, len);
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return static_cast<int>(len);
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}
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int ConferenceTransport::GetReceiverChannelForSsrc(unsigned int sender_ssrc)
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const {
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webrtc::CriticalSectionScoped lock(stream_crit_.get());
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auto it = streams_.find(sender_ssrc);
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if (it != streams_.end()) {
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return it->second.second;
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}
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return -1;
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}
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void ConferenceTransport::StorePacket(Packet::Type type, int channel,
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const void* data, size_t len) {
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{
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webrtc::CriticalSectionScoped lock(pq_crit_.get());
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packet_queue_.push_back(Packet(type, channel, data, len, rtc::Time()));
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}
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packet_event_->Set();
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}
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// This simulates the flow of RTP and RTCP packets. Complications like that
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// a packet is first sent to the reflector, and then forwarded to the receiver
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// are simplified, in this particular case, to a direct link between the sender
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// and the receiver.
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void ConferenceTransport::SendPacket(const Packet& packet) const {
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unsigned int sender_ssrc;
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int destination = -1;
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switch (packet.type_) {
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case Packet::Rtp:
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sender_ssrc = ParseSsrc(packet.data_, packet.len_, false);
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if (sender_ssrc == kLocalSsrc) {
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remote_network_->ReceivedRTPPacket(reflector_, packet.data_,
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packet.len_, webrtc::PacketTime());
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} else {
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destination = GetReceiverChannelForSsrc(sender_ssrc);
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if (destination != -1) {
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local_network_->ReceivedRTPPacket(destination, packet.data_,
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packet.len_,
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webrtc::PacketTime());
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}
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}
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break;
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case Packet::Rtcp:
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sender_ssrc = ParseSsrc(packet.data_, packet.len_, true);
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if (sender_ssrc == kLocalSsrc) {
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remote_network_->ReceivedRTCPPacket(reflector_, packet.data_,
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packet.len_);
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} else if (sender_ssrc == kReflectorSsrc) {
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local_network_->ReceivedRTCPPacket(local_sender_, packet.data_,
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packet.len_);
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} else {
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destination = GetReceiverChannelForSsrc(sender_ssrc);
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if (destination != -1) {
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local_network_->ReceivedRTCPPacket(destination, packet.data_,
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packet.len_);
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}
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}
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break;
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}
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}
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bool ConferenceTransport::DispatchPackets() {
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switch (packet_event_->Wait(1000)) {
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case webrtc::kEventSignaled:
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break;
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case webrtc::kEventTimeout:
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return true;
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case webrtc::kEventError:
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ADD_FAILURE() << "kEventError encountered.";
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return true;
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}
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while (true) {
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Packet packet;
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{
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webrtc::CriticalSectionScoped lock(pq_crit_.get());
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if (packet_queue_.empty())
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break;
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packet = packet_queue_.front();
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packet_queue_.pop_front();
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}
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int32 elapsed_time_ms = rtc::TimeSince(packet.send_time_ms_);
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int32 sleep_ms = rtt_ms_ / 2 - elapsed_time_ms;
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if (sleep_ms > 0) {
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// Every packet should be delayed by half of RTT.
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webrtc::SleepMs(sleep_ms);
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}
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SendPacket(packet);
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}
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return true;
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}
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void ConferenceTransport::SetRtt(unsigned int rtt_ms) {
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rtt_ms_ = rtt_ms;
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}
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unsigned int ConferenceTransport::AddStream() {
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const int new_sender = remote_base_->CreateChannel();
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EXPECT_EQ(0, remote_network_->RegisterExternalTransport(new_sender, *this));
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const unsigned int remote_ssrc = kFirstRemoteSsrc + stream_count_++;
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EXPECT_EQ(0, remote_rtp_rtcp_->SetLocalSSRC(new_sender, remote_ssrc));
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EXPECT_EQ(0, remote_codec_->SetSendCodec(new_sender, kCodecInst));
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EXPECT_EQ(0, remote_base_->StartSend(new_sender));
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EXPECT_EQ(0, remote_file_->StartPlayingFileAsMicrophone(
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new_sender, kInputFileName.c_str(), true, false,
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kInputFileFormat, 1.0));
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const int new_receiver = local_base_->CreateChannel();
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EXPECT_EQ(0, local_base_->AssociateSendChannel(new_receiver, local_sender_));
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EXPECT_EQ(0, local_network_->RegisterExternalTransport(new_receiver, *this));
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// Receive channels have to have the same SSRC in order to send receiver
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// reports with this SSRC.
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EXPECT_EQ(0, local_rtp_rtcp_->SetLocalSSRC(new_receiver, kLocalSsrc));
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{
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webrtc::CriticalSectionScoped lock(stream_crit_.get());
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streams_[remote_ssrc] = std::make_pair(new_sender, new_receiver);
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}
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return remote_ssrc; // remote ssrc used as stream id.
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}
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bool ConferenceTransport::RemoveStream(unsigned int id) {
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webrtc::CriticalSectionScoped lock(stream_crit_.get());
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auto it = streams_.find(id);
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if (it == streams_.end()) {
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return false;
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}
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EXPECT_EQ(0, remote_network_->
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DeRegisterExternalTransport(it->second.second));
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EXPECT_EQ(0, local_network_->
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DeRegisterExternalTransport(it->second.first));
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EXPECT_EQ(0, remote_base_->DeleteChannel(it->second.second));
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EXPECT_EQ(0, local_base_->DeleteChannel(it->second.first));
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streams_.erase(it);
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return true;
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}
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bool ConferenceTransport::StartPlayout(unsigned int id) {
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int dst = GetReceiverChannelForSsrc(id);
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if (dst == -1) {
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return false;
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}
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EXPECT_EQ(0, local_base_->StartPlayout(dst));
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return true;
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}
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bool ConferenceTransport::GetReceiverStatistics(unsigned int id,
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webrtc::CallStatistics* stats) {
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int dst = GetReceiverChannelForSsrc(id);
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if (dst == -1) {
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return false;
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}
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EXPECT_EQ(0, local_rtp_rtcp_->GetRTCPStatistics(dst, *stats));
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return true;
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}
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} // namespace voetest
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@ -1,158 +0,0 @@
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/*
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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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#ifndef WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_
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#define WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_
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#include <deque>
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#include <map>
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#include <utility>
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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/base/basictypes.h"
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/common_types.h"
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#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
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#include "webrtc/system_wrappers/interface/event_wrapper.h"
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#include "webrtc/system_wrappers/interface/thread_wrapper.h"
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#include "webrtc/voice_engine/include/voe_base.h"
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#include "webrtc/voice_engine/include/voe_codec.h"
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#include "webrtc/voice_engine/include/voe_file.h"
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#include "webrtc/voice_engine/include/voe_network.h"
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#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
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static const size_t kMaxPacketSizeByte = 1500;
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namespace voetest {
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// This class is to simulate a conference call. There are two Voice Engines, one
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// for local channels and the other for remote channels. There is a simulated
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// reflector, which exchanges RTCP with local channels. For simplicity, it
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// also uses the Voice Engine for remote channels. One can add streams by
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// calling AddStream(), which creates a remote sender channel and a local
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// receive channel. The remote sender channel plays a file as microphone in a
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// looped fashion. Received streams are mixed and played.
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class ConferenceTransport: public webrtc::Transport {
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public:
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ConferenceTransport();
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virtual ~ConferenceTransport();
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/* SetRtt()
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* Set RTT between local channels and reflector.
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*
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* Input:
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* rtt_ms : RTT in milliseconds.
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*/
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void SetRtt(unsigned int rtt_ms);
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/* AddStream()
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* Adds a stream in the conference.
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*
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* Returns stream id.
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*/
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unsigned int AddStream();
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/* RemoveStream()
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* Removes a stream with specified ID from the conference.
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*
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* Input:
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* id : stream id.
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*
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* Returns false if the specified stream does not exist, true if succeeds.
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*/
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bool RemoveStream(unsigned int id);
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/* StartPlayout()
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* Starts playing out the stream with specified ID, using the default device.
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*
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* Input:
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* id : stream id.
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*
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* Returns false if the specified stream does not exist, true if succeeds.
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*/
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bool StartPlayout(unsigned int id);
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/* GetReceiverStatistics()
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* Gets RTCP statistics of the stream with specified ID.
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*
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* Input:
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* id : stream id;
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||||||
* stats : pointer to a CallStatistics to store the result.
|
|
||||||
*
|
|
||||||
* Returns false if the specified stream does not exist, true if succeeds.
|
|
||||||
*/
|
|
||||||
bool GetReceiverStatistics(unsigned int id, webrtc::CallStatistics* stats);
|
|
||||||
|
|
||||||
// Inherit from class webrtc::Transport.
|
|
||||||
int SendPacket(int channel, const void *data, size_t len) override;
|
|
||||||
int SendRTCPPacket(int channel, const void *data, size_t len) override;
|
|
||||||
|
|
||||||
private:
|
|
||||||
struct Packet {
|
|
||||||
enum Type { Rtp, Rtcp, } type_;
|
|
||||||
|
|
||||||
Packet() : len_(0) {}
|
|
||||||
Packet(Type type, int channel, const void* data, size_t len, uint32 time_ms)
|
|
||||||
: type_(type),
|
|
||||||
channel_(channel),
|
|
||||||
len_(len),
|
|
||||||
send_time_ms_(time_ms) {
|
|
||||||
EXPECT_LE(len_, kMaxPacketSizeByte);
|
|
||||||
memcpy(data_, data, len_);
|
|
||||||
}
|
|
||||||
|
|
||||||
int channel_;
|
|
||||||
uint8_t data_[kMaxPacketSizeByte];
|
|
||||||
size_t len_;
|
|
||||||
uint32 send_time_ms_;
|
|
||||||
};
|
|
||||||
|
|
||||||
static bool Run(void* transport) {
|
|
||||||
return static_cast<ConferenceTransport*>(transport)->DispatchPackets();
|
|
||||||
}
|
|
||||||
|
|
||||||
int GetReceiverChannelForSsrc(unsigned int sender_ssrc) const;
|
|
||||||
void StorePacket(Packet::Type type, int channel, const void* data,
|
|
||||||
size_t len);
|
|
||||||
void SendPacket(const Packet& packet) const;
|
|
||||||
bool DispatchPackets();
|
|
||||||
|
|
||||||
const rtc::scoped_ptr<webrtc::CriticalSectionWrapper> pq_crit_;
|
|
||||||
const rtc::scoped_ptr<webrtc::CriticalSectionWrapper> stream_crit_;
|
|
||||||
const rtc::scoped_ptr<webrtc::EventWrapper> packet_event_;
|
|
||||||
const rtc::scoped_ptr<webrtc::ThreadWrapper> thread_;
|
|
||||||
|
|
||||||
unsigned int rtt_ms_;
|
|
||||||
unsigned int stream_count_;
|
|
||||||
|
|
||||||
std::map<unsigned int, std::pair<int, int>> streams_
|
|
||||||
GUARDED_BY(stream_crit_.get());
|
|
||||||
std::deque<Packet> packet_queue_ GUARDED_BY(pq_crit_.get());
|
|
||||||
|
|
||||||
int local_sender_; // Channel Id of local sender
|
|
||||||
int reflector_;
|
|
||||||
|
|
||||||
webrtc::VoiceEngine* local_voe_;
|
|
||||||
webrtc::VoEBase* local_base_;
|
|
||||||
webrtc::VoERTP_RTCP* local_rtp_rtcp_;
|
|
||||||
webrtc::VoENetwork* local_network_;
|
|
||||||
|
|
||||||
webrtc::VoiceEngine* remote_voe_;
|
|
||||||
webrtc::VoEBase* remote_base_;
|
|
||||||
webrtc::VoECodec* remote_codec_;
|
|
||||||
webrtc::VoERTP_RTCP* remote_rtp_rtcp_;
|
|
||||||
webrtc::VoENetwork* remote_network_;
|
|
||||||
webrtc::VoEFile* remote_file_;
|
|
||||||
};
|
|
||||||
} // namespace voetest
|
|
||||||
|
|
||||||
#endif // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_
|
|
||||||
@ -1,108 +0,0 @@
|
|||||||
/*
|
|
||||||
* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
|
|
||||||
*
|
|
||||||
* Use of this source code is governed by a BSD-style license
|
|
||||||
* that can be found in the LICENSE file in the root of the source
|
|
||||||
* tree. An additional intellectual property rights grant can be found
|
|
||||||
* in the file PATENTS. All contributing project authors may
|
|
||||||
* be found in the AUTHORS file in the root of the source tree.
|
|
||||||
*/
|
|
||||||
|
|
||||||
#include <queue>
|
|
||||||
|
|
||||||
#include "testing/gtest/include/gtest/gtest.h"
|
|
||||||
#include "webrtc/base/format_macros.h"
|
|
||||||
#include "webrtc/base/timeutils.h"
|
|
||||||
#include "webrtc/system_wrappers/interface/sleep.h"
|
|
||||||
#include "webrtc/voice_engine/test/auto_test/fakes/conference_transport.h"
|
|
||||||
|
|
||||||
namespace {
|
|
||||||
static const int kRttMs = 25;
|
|
||||||
|
|
||||||
static bool IsNear(int ref, int comp, int error) {
|
|
||||||
return (ref - comp <= error) && (comp - ref >= -error);
|
|
||||||
}
|
|
||||||
}
|
|
||||||
|
|
||||||
namespace voetest {
|
|
||||||
|
|
||||||
TEST(VoeConferenceTest, RttAndStartNtpTime) {
|
|
||||||
struct Stats {
|
|
||||||
Stats(int64_t rtt_receiver_1, int64_t rtt_receiver_2, int64_t ntp_delay)
|
|
||||||
: rtt_receiver_1_(rtt_receiver_1),
|
|
||||||
rtt_receiver_2_(rtt_receiver_2),
|
|
||||||
ntp_delay_(ntp_delay) {
|
|
||||||
}
|
|
||||||
int64_t rtt_receiver_1_;
|
|
||||||
int64_t rtt_receiver_2_;
|
|
||||||
int64_t ntp_delay_;
|
|
||||||
};
|
|
||||||
|
|
||||||
const int kDelayMs = 987;
|
|
||||||
ConferenceTransport trans;
|
|
||||||
trans.SetRtt(kRttMs);
|
|
||||||
|
|
||||||
unsigned int id_1 = trans.AddStream();
|
|
||||||
unsigned int id_2 = trans.AddStream();
|
|
||||||
|
|
||||||
EXPECT_TRUE(trans.StartPlayout(id_1));
|
|
||||||
// Start NTP time is the time when a stream is played out, rather than
|
|
||||||
// when it is added.
|
|
||||||
webrtc::SleepMs(kDelayMs);
|
|
||||||
EXPECT_TRUE(trans.StartPlayout(id_2));
|
|
||||||
|
|
||||||
const int kMaxRunTimeMs = 25000;
|
|
||||||
const int kNeedSuccessivePass = 3;
|
|
||||||
const int kStatsRequestIntervalMs = 1000;
|
|
||||||
const int kStatsBufferSize = 3;
|
|
||||||
|
|
||||||
uint32 deadline = rtc::TimeAfter(kMaxRunTimeMs);
|
|
||||||
// Run the following up to |kMaxRunTimeMs| milliseconds.
|
|
||||||
int successive_pass = 0;
|
|
||||||
webrtc::CallStatistics stats_1;
|
|
||||||
webrtc::CallStatistics stats_2;
|
|
||||||
std::queue<Stats> stats_buffer;
|
|
||||||
|
|
||||||
while (rtc::TimeIsLater(rtc::Time(), deadline) &&
|
|
||||||
successive_pass < kNeedSuccessivePass) {
|
|
||||||
webrtc::SleepMs(kStatsRequestIntervalMs);
|
|
||||||
|
|
||||||
EXPECT_TRUE(trans.GetReceiverStatistics(id_1, &stats_1));
|
|
||||||
EXPECT_TRUE(trans.GetReceiverStatistics(id_2, &stats_2));
|
|
||||||
|
|
||||||
// It is not easy to verify the NTP time directly. We verify it by testing
|
|
||||||
// the difference of two start NTP times.
|
|
||||||
int64_t captured_start_ntp_delay = stats_2.capture_start_ntp_time_ms_ -
|
|
||||||
stats_1.capture_start_ntp_time_ms_;
|
|
||||||
|
|
||||||
// For the checks of RTT and start NTP time, We allow 10% accuracy.
|
|
||||||
if (IsNear(kRttMs, stats_1.rttMs, kRttMs / 10 + 1) &&
|
|
||||||
IsNear(kRttMs, stats_2.rttMs, kRttMs / 10 + 1) &&
|
|
||||||
IsNear(kDelayMs, captured_start_ntp_delay, kDelayMs / 10 + 1)) {
|
|
||||||
successive_pass++;
|
|
||||||
} else {
|
|
||||||
successive_pass = 0;
|
|
||||||
}
|
|
||||||
if (stats_buffer.size() >= kStatsBufferSize) {
|
|
||||||
stats_buffer.pop();
|
|
||||||
}
|
|
||||||
stats_buffer.push(Stats(stats_1.rttMs, stats_2.rttMs,
|
|
||||||
captured_start_ntp_delay));
|
|
||||||
}
|
|
||||||
|
|
||||||
EXPECT_GE(successive_pass, kNeedSuccessivePass) << "Expected to get RTT and"
|
|
||||||
" start NTP time estimate within 10% of the correct value over "
|
|
||||||
<< kStatsRequestIntervalMs * kNeedSuccessivePass / 1000
|
|
||||||
<< " seconds.";
|
|
||||||
if (successive_pass < kNeedSuccessivePass) {
|
|
||||||
printf("The most recent values (RTT for receiver 1, RTT for receiver 2, "
|
|
||||||
"NTP delay between receiver 1 and 2) are (from oldest):\n");
|
|
||||||
while (!stats_buffer.empty()) {
|
|
||||||
Stats stats = stats_buffer.front();
|
|
||||||
printf("(%" PRId64 ", %" PRId64 ", %" PRId64 ")\n", stats.rtt_receiver_1_,
|
|
||||||
stats.rtt_receiver_2_, stats.ntp_delay_);
|
|
||||||
stats_buffer.pop();
|
|
||||||
}
|
|
||||||
}
|
|
||||||
}
|
|
||||||
} // namespace voetest
|
|
||||||
@ -158,8 +158,6 @@
|
|||||||
'test/auto_test/automated_mode.cc',
|
'test/auto_test/automated_mode.cc',
|
||||||
'test/auto_test/extended/agc_config_test.cc',
|
'test/auto_test/extended/agc_config_test.cc',
|
||||||
'test/auto_test/extended/ec_metrics_test.cc',
|
'test/auto_test/extended/ec_metrics_test.cc',
|
||||||
'test/auto_test/fakes/conference_transport.cc',
|
|
||||||
'test/auto_test/fakes/conference_transport.h',
|
|
||||||
'test/auto_test/fakes/fake_external_transport.cc',
|
'test/auto_test/fakes/fake_external_transport.cc',
|
||||||
'test/auto_test/fakes/fake_external_transport.h',
|
'test/auto_test/fakes/fake_external_transport.h',
|
||||||
'test/auto_test/fixtures/after_initialization_fixture.cc',
|
'test/auto_test/fixtures/after_initialization_fixture.cc',
|
||||||
@ -189,7 +187,6 @@
|
|||||||
'test/auto_test/standard/video_sync_test.cc',
|
'test/auto_test/standard/video_sync_test.cc',
|
||||||
'test/auto_test/standard/volume_test.cc',
|
'test/auto_test/standard/volume_test.cc',
|
||||||
'test/auto_test/resource_manager.cc',
|
'test/auto_test/resource_manager.cc',
|
||||||
'test/auto_test/voe_conference_test.cc',
|
|
||||||
'test/auto_test/voe_cpu_test.cc',
|
'test/auto_test/voe_cpu_test.cc',
|
||||||
'test/auto_test/voe_cpu_test.h',
|
'test/auto_test/voe_cpu_test.h',
|
||||||
'test/auto_test/voe_standard_test.cc',
|
'test/auto_test/voe_standard_test.cc',
|
||||||
|
|||||||
Loading…
x
Reference in New Issue
Block a user