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/*
* 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
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