Remove unused voe_output_test.cc.
BUG=webrtc:4690 Review-Url: https://codereview.webrtc.org/2684933007 Cr-Commit-Position: refs/heads/master@{#16506}
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@ -387,7 +387,6 @@ if (rtc_include_tests) {
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"test/auto_test/voe_conference_test.cc",
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"test/auto_test/voe_cpu_test.cc",
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"test/auto_test/voe_cpu_test.h",
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"test/auto_test/voe_output_test.cc",
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"test/auto_test/voe_standard_test.cc",
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"test/auto_test/voe_standard_test.h",
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"test/auto_test/voe_stress_test.cc",
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@ -1,202 +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/base/random.h"
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#include "webrtc/base/timeutils.h"
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#include "webrtc/system_wrappers/include/sleep.h"
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#include "webrtc/test/gtest.h"
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#include "webrtc/test/testsupport/fileutils.h"
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#include "webrtc/voice_engine/test/auto_test/voe_standard_test.h"
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#include "webrtc/voice_engine/test/channel_transport/channel_transport.h"
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namespace {
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const char kIp[] = "127.0.0.1";
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const int kPort = 1234;
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const webrtc::CodecInst kCodecInst = {120, "opus", 48000, 960, 2, 64000};
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} // namespace
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namespace voetest {
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using webrtc::Random;
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using webrtc::test::VoiceChannelTransport;
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// This test allows a check on the output signal in an end-to-end call.
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class OutputTest {
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public:
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OutputTest(int16_t lower_bound, int16_t upper_bound);
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~OutputTest();
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void Start();
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void EnableOutputCheck();
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void DisableOutputCheck();
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void SetOutputBound(int16_t lower_bound, int16_t upper_bound);
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void Mute();
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void Unmute();
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void SetBitRate(int rate);
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private:
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// This class checks all output values and count the number of samples that
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// go out of a defined range.
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class VoEOutputCheckMediaProcess : public VoEMediaProcess {
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public:
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VoEOutputCheckMediaProcess(int16_t lower_bound, int16_t upper_bound);
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void set_enabled(bool enabled) { enabled_ = enabled; }
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void Process(int channel,
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ProcessingTypes type,
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int16_t audio10ms[],
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size_t length,
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int samplingFreq,
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bool isStereo) override;
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private:
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bool enabled_;
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int16_t lower_bound_;
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int16_t upper_bound_;
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};
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VoETestManager manager_;
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VoEOutputCheckMediaProcess output_checker_;
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int channel_;
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};
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OutputTest::OutputTest(int16_t lower_bound, int16_t upper_bound)
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: output_checker_(lower_bound, upper_bound) {
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EXPECT_TRUE(manager_.Init());
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manager_.GetInterfaces();
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VoEBase* base = manager_.BasePtr();
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VoECodec* codec = manager_.CodecPtr();
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VoENetwork* network = manager_.NetworkPtr();
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EXPECT_EQ(0, base->Init());
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channel_ = base->CreateChannel();
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// |network| will take care of the life time of |transport|.
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VoiceChannelTransport* transport =
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new VoiceChannelTransport(network, channel_);
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EXPECT_EQ(0, transport->SetSendDestination(kIp, kPort));
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EXPECT_EQ(0, transport->SetLocalReceiver(kPort));
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EXPECT_EQ(0, codec->SetSendCodec(channel_, kCodecInst));
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EXPECT_EQ(0, codec->SetOpusDtx(channel_, true));
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EXPECT_EQ(0, manager_.VolumeControlPtr()->SetSpeakerVolume(255));
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manager_.ExternalMediaPtr()->RegisterExternalMediaProcessing(
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channel_, ProcessingTypes::kPlaybackPerChannel, output_checker_);
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}
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OutputTest::~OutputTest() {
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EXPECT_EQ(0, manager_.NetworkPtr()->DeRegisterExternalTransport(channel_));
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EXPECT_EQ(0, manager_.ReleaseInterfaces());
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}
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void OutputTest::Start() {
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const std::string file_name =
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webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
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const webrtc::FileFormats kInputFormat = webrtc::kFileFormatPcm32kHzFile;
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ASSERT_EQ(0, manager_.FilePtr()->StartPlayingFileAsMicrophone(
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channel_, file_name.c_str(), true, false, kInputFormat, 1.0));
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VoEBase* base = manager_.BasePtr();
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ASSERT_EQ(0, base->StartPlayout(channel_));
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ASSERT_EQ(0, base->StartSend(channel_));
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}
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void OutputTest::EnableOutputCheck() {
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output_checker_.set_enabled(true);
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}
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void OutputTest::DisableOutputCheck() {
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output_checker_.set_enabled(false);
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}
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void OutputTest::Mute() {
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manager_.VolumeControlPtr()->SetInputMute(channel_, true);
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}
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void OutputTest::Unmute() {
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manager_.VolumeControlPtr()->SetInputMute(channel_, false);
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}
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void OutputTest::SetBitRate(int rate) {
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manager_.CodecPtr()->SetBitRate(channel_, rate);
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}
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OutputTest::VoEOutputCheckMediaProcess::VoEOutputCheckMediaProcess(
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int16_t lower_bound, int16_t upper_bound)
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: enabled_(false),
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lower_bound_(lower_bound),
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upper_bound_(upper_bound) {}
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void OutputTest::VoEOutputCheckMediaProcess::Process(int channel,
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ProcessingTypes type,
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int16_t* audio10ms,
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size_t length,
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int samplingFreq,
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bool isStereo) {
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if (!enabled_)
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return;
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const int num_channels = isStereo ? 2 : 1;
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for (size_t i = 0; i < length; ++i) {
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for (int c = 0; c < num_channels; ++c) {
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ASSERT_GE(audio10ms[i * num_channels + c], lower_bound_);
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ASSERT_LE(audio10ms[i * num_channels + c], upper_bound_);
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}
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}
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}
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// This test checks if the Opus does not produce high noise (noise pump) when
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// DTX is enabled. The microphone is toggled on and off, and values of the
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// output signal during muting should be bounded.
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// We do not run this test on bots. Developers that want to see the result
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// and/or listen to sound quality can run this test manually.
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TEST(OutputTest, DISABLED_OpusDtxHasNoNoisePump) {
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const int kRuntimeMs = 20000;
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const uint32_t kUnmuteTimeMs = 1000;
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const int kCheckAfterMute = 2000;
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const uint32_t kCheckTimeMs = 2000;
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const int kMinOpusRate = 6000;
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const int kMaxOpusRate = 64000;
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#if defined(OPUS_FIXED_POINT)
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const int16_t kDtxBoundForSilence = 20;
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#else
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const int16_t kDtxBoundForSilence = 2;
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#endif
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OutputTest test(-kDtxBoundForSilence, kDtxBoundForSilence);
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Random random(1234ull);
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int64_t start_time = rtc::TimeMillis();
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test.Start();
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while (rtc::TimeSince(start_time) < kRuntimeMs) {
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webrtc::SleepMs(random.Rand(kUnmuteTimeMs - kUnmuteTimeMs / 10,
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kUnmuteTimeMs + kUnmuteTimeMs / 10));
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test.Mute();
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webrtc::SleepMs(kCheckAfterMute);
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test.EnableOutputCheck();
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webrtc::SleepMs(random.Rand(kCheckTimeMs - kCheckTimeMs / 10,
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kCheckTimeMs + kCheckTimeMs / 10));
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test.DisableOutputCheck();
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test.SetBitRate(random.Rand(kMinOpusRate, kMaxOpusRate));
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test.Unmute();
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}
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}
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} // namespace voetest
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