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Diffstat (limited to 'webrtc/voice_engine/test/auto_test/voe_output_test.cc')
| -rw-r--r-- | webrtc/voice_engine/test/auto_test/voe_output_test.cc | 203 |
1 files changed, 203 insertions, 0 deletions
diff --git a/webrtc/voice_engine/test/auto_test/voe_output_test.cc b/webrtc/voice_engine/test/auto_test/voe_output_test.cc new file mode 100644 index 0000000000..3bedbc3b17 --- /dev/null +++ b/webrtc/voice_engine/test/auto_test/voe_output_test.cc @@ -0,0 +1,203 @@ +/* + * 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/random.h" +#include "webrtc/base/scoped_ptr.h" +#include "webrtc/base/timeutils.h" +#include "webrtc/system_wrappers/include/sleep.h" +#include "webrtc/test/channel_transport/channel_transport.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::Random; +using webrtc::test::VoiceChannelTransport; + +// This test allows a check on the output signal in an end-to-end call. +class OutputTest { + public: + OutputTest(int16_t lower_bound, int16_t upper_bound); + ~OutputTest(); + + void Start(); + + void EnableOutputCheck(); + void DisableOutputCheck(); + void SetOutputBound(int16_t lower_bound, int16_t upper_bound); + void Mute(); + void Unmute(); + void SetBitRate(int rate); + + private: + // This class checks all output values and count the number of samples that + // go out of a defined range. + class VoEOutputCheckMediaProcess : public VoEMediaProcess { + public: + VoEOutputCheckMediaProcess(int16_t lower_bound, int16_t upper_bound); + + void set_enabled(bool enabled) { enabled_ = enabled; } + void Process(int channel, + ProcessingTypes type, + int16_t audio10ms[], + size_t length, + int samplingFreq, + bool isStereo) override; + + private: + bool enabled_; + int16_t lower_bound_; + int16_t upper_bound_; + }; + + VoETestManager manager_; + VoEOutputCheckMediaProcess output_checker_; + + int channel_; +}; + +OutputTest::OutputTest(int16_t lower_bound, int16_t upper_bound) + : output_checker_(lower_bound, upper_bound) { + EXPECT_TRUE(manager_.Init()); + manager_.GetInterfaces(); + + VoEBase* base = manager_.BasePtr(); + VoECodec* codec = manager_.CodecPtr(); + VoENetwork* network = manager_.NetworkPtr(); + + EXPECT_EQ(0, base->Init()); + + channel_ = base->CreateChannel(); + + // |network| will take care of the life time of |transport|. + VoiceChannelTransport* transport = + new VoiceChannelTransport(network, channel_); + + EXPECT_EQ(0, transport->SetSendDestination(kIp, kPort)); + EXPECT_EQ(0, transport->SetLocalReceiver(kPort)); + + EXPECT_EQ(0, codec->SetSendCodec(channel_, kCodecInst)); + EXPECT_EQ(0, codec->SetOpusDtx(channel_, true)); + + EXPECT_EQ(0, manager_.VolumeControlPtr()->SetSpeakerVolume(255)); + + manager_.ExternalMediaPtr()->RegisterExternalMediaProcessing( + channel_, ProcessingTypes::kPlaybackPerChannel, output_checker_); +} + +OutputTest::~OutputTest() { + EXPECT_EQ(0, manager_.NetworkPtr()->DeRegisterExternalTransport(channel_)); + EXPECT_EQ(0, manager_.ReleaseInterfaces()); +} + +void OutputTest::Start() { + const std::string file_name = + webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"); + const webrtc::FileFormats kInputFormat = webrtc::kFileFormatPcm32kHzFile; + + ASSERT_EQ(0, manager_.FilePtr()->StartPlayingFileAsMicrophone( + channel_, file_name.c_str(), true, false, kInputFormat, 1.0)); + + VoEBase* base = manager_.BasePtr(); + ASSERT_EQ(0, base->StartPlayout(channel_)); + ASSERT_EQ(0, base->StartSend(channel_)); +} + +void OutputTest::EnableOutputCheck() { + output_checker_.set_enabled(true); +} + +void OutputTest::DisableOutputCheck() { + output_checker_.set_enabled(false); +} + +void OutputTest::Mute() { + manager_.VolumeControlPtr()->SetInputMute(channel_, true); +} + +void OutputTest::Unmute() { + manager_.VolumeControlPtr()->SetInputMute(channel_, false); +} + +void OutputTest::SetBitRate(int rate) { + manager_.CodecPtr()->SetBitRate(channel_, rate); +} + +OutputTest::VoEOutputCheckMediaProcess::VoEOutputCheckMediaProcess( + int16_t lower_bound, int16_t upper_bound) + : enabled_(false), + lower_bound_(lower_bound), + upper_bound_(upper_bound) {} + +void OutputTest::VoEOutputCheckMediaProcess::Process(int channel, + ProcessingTypes type, + int16_t* audio10ms, + size_t length, + int samplingFreq, + bool isStereo) { + if (!enabled_) + return; + const int num_channels = isStereo ? 2 : 1; + for (size_t i = 0; i < length; ++i) { + for (int c = 0; c < num_channels; ++c) { + ASSERT_GE(audio10ms[i * num_channels + c], lower_bound_); + ASSERT_LE(audio10ms[i * num_channels + c], upper_bound_); + } + } +} + +// This test checks if the Opus does not produce high noise (noise pump) when +// DTX is enabled. The microphone is toggled on and off, and values of the +// output signal during muting should be bounded. +// We do not run this test on bots. Developers that want to see the result +// and/or listen to sound quality can run this test manually. +TEST(OutputTest, DISABLED_OpusDtxHasNoNoisePump) { + const int kRuntimeMs = 20000; + const uint32_t kUnmuteTimeMs = 1000; + const int kCheckAfterMute = 2000; + const uint32_t kCheckTimeMs = 2000; + const int kMinOpusRate = 6000; + const int kMaxOpusRate = 64000; + +#if defined(OPUS_FIXED_POINT) + const int16_t kDtxBoundForSilence = 20; +#else + const int16_t kDtxBoundForSilence = 2; +#endif + + OutputTest test(-kDtxBoundForSilence, kDtxBoundForSilence); + Random random(1234ull); + + uint32_t start_time = rtc::Time(); + test.Start(); + while (rtc::TimeSince(start_time) < kRuntimeMs) { + webrtc::SleepMs(random.Rand(kUnmuteTimeMs - kUnmuteTimeMs / 10, + kUnmuteTimeMs + kUnmuteTimeMs / 10)); + test.Mute(); + webrtc::SleepMs(kCheckAfterMute); + test.EnableOutputCheck(); + webrtc::SleepMs(random.Rand(kCheckTimeMs - kCheckTimeMs / 10, + kCheckTimeMs + kCheckTimeMs / 10)); + test.DisableOutputCheck(); + test.SetBitRate(random.Rand(kMinOpusRate, kMaxOpusRate)); + test.Unmute(); + } +} + +} // namespace voetest |