| Index: webrtc/voice_engine/test/auto_test/voe_output_test.cc
|
| diff --git a/webrtc/voice_engine/test/auto_test/voe_output_test.cc b/webrtc/voice_engine/test/auto_test/voe_output_test.cc
|
| deleted file mode 100644
|
| index 6a842b80b9a0819a3f2af0675387886aa32a8a71..0000000000000000000000000000000000000000
|
| --- a/webrtc/voice_engine/test/auto_test/voe_output_test.cc
|
| +++ /dev/null
|
| @@ -1,198 +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 "testing/gtest/include/gtest/gtest.h"
|
| -#include "webrtc/base/scoped_ptr.h"
|
| -#include "webrtc/base/timeutils.h"
|
| -#include "webrtc/system_wrappers/include/sleep.h"
|
| -#include "webrtc/test/channel_transport/include/channel_transport.h"
|
| -#include "webrtc/test/random.h"
|
| -#include "webrtc/test/testsupport/fileutils.h"
|
| -#include "webrtc/voice_engine/test/auto_test/voe_standard_test.h"
|
| -
|
| -namespace {
|
| -
|
| -const char kIp[] = "127.0.0.1";
|
| -const int kPort = 1234;
|
| -const webrtc::CodecInst kCodecInst = {120, "opus", 48000, 960, 2, 64000};
|
| -
|
| -} // namespace
|
| -
|
| -namespace voetest {
|
| -
|
| -using webrtc::test::Random;
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| -using webrtc::test::VoiceChannelTransport;
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| -
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| -// This test allows a check on the output signal in an end-to-end call.
|
| -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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| -
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| - void Start();
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| -
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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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| -
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| - private:
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| - // This class checks all output values and count the number of samples that
|
| - // go out of a defined range.
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| - class VoEOutputCheckMediaProcess : public VoEMediaProcess {
|
| - public:
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| - VoEOutputCheckMediaProcess(int16_t lower_bound, int16_t upper_bound);
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| -
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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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| -
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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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| -
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| - VoETestManager manager_;
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| - VoEOutputCheckMediaProcess output_checker_;
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| -
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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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| -
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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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| -
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| - EXPECT_EQ(0, base->Init());
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| -
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| - channel_ = base->CreateChannel();
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| -
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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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| -
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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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| -
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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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| -
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| - EXPECT_EQ(0, manager_.VolumeControlPtr()->SetSpeakerVolume(255));
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| -
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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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| -
|
| -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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| -
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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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| -
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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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| -
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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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| -
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| -void OutputTest::Mute() {
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| - manager_.VolumeControlPtr()->SetInputMute(channel_, true);
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| -}
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| -
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| -void OutputTest::Unmute() {
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| - manager_.VolumeControlPtr()->SetInputMute(channel_, false);
|
| -}
|
| -
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| -void OutputTest::SetBitRate(int rate) {
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| - manager_.CodecPtr()->SetBitRate(channel_, rate);
|
| -}
|
| -
|
| -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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| -
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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;
|
| - const int num_channels = isStereo ? 2 : 1;
|
| - 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_);
|
| - }
|
| - }
|
| -}
|
| -
|
| -TEST(OutputTest, OpusDtxHasNoNoisePump) {
|
| - const int kRuntimeMs = 20000;
|
| - const uint32_t kUnmuteTimeMs = 1000;
|
| - const int kCheckAfterMute = 2000;
|
| - const uint32_t kCheckTimeMs = 2000;
|
| - const int kMinOpusRate = 6000;
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| - const int kMaxOpusRate = 64000;
|
| -
|
| -#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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| -
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| - OutputTest test(-kDtxBoundForSilence, kDtxBoundForSilence);
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| - Random random(1234ull);
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| -
|
| - uint32_t start_time = rtc::Time();
|
| - test.Start();
|
| - 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();
|
| - }
|
| -}
|
| -
|
| -} // namespace voetest
|
|
|