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// Copyright 2023 The Chromium Authors
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
#include "third_party/blink/renderer/modules/webrtc/webrtc_audio_device_impl.h"
#include <memory>
#include "base/time/time.h"
#include "media/base/audio_bus.h"
#include "media/base/audio_glitch_info.h"
#include "media/base/audio_parameters.h"
#include "media/base/audio_timestamp_helper.h"
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "third_party/blink/renderer/platform/testing/task_environment.h"
#include "third_party/blink/renderer/platform/webrtc/webrtc_source.h"
#include "third_party/webrtc/rtc_base/ref_counted_object.h"
namespace blink {
namespace {
class MockAudioTransport : public webrtc::AudioTransport {
public:
MockAudioTransport() = default;
MockAudioTransport(const MockAudioTransport&) = delete;
MockAudioTransport& operator=(const MockAudioTransport&) = delete;
MOCK_METHOD10(RecordedDataIsAvailable,
int32_t(const void* audioSamples,
size_t nSamples,
size_t nBytesPerSample,
size_t nChannels,
uint32_t samplesPerSec,
uint32_t totalDelayMS,
int32_t clockDrift,
uint32_t currentMicLevel,
bool keyPressed,
uint32_t& newMicLevel));
MOCK_METHOD8(NeedMorePlayData,
int32_t(size_t nSamples,
size_t nBytesPerSample,
size_t nChannels,
uint32_t samplesPerSec,
void* audioSamples,
size_t& nSamplesOut,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms));
MOCK_METHOD7(PullRenderData,
void(int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames,
void* audio_data,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms));
};
const int kHardwareSampleRate = 44100;
const int kHardwareBufferSize = 512;
const media::AudioParameters kAudioParameters =
media::AudioParameters(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
media::ChannelLayoutConfig::Stereo(),
kHardwareSampleRate,
kHardwareBufferSize);
} // namespace
class WebRtcAudioDeviceImplTest : public testing::Test {
public:
WebRtcAudioDeviceImplTest()
: audio_device_(
new webrtc::RefCountedObject<blink::WebRtcAudioDeviceImpl>()),
audio_transport_(new MockAudioTransport()) {
audio_device_module()->Init();
audio_device_module()->RegisterAudioCallback(audio_transport_.get());
}
~WebRtcAudioDeviceImplTest() override { audio_device_module()->Terminate(); }
protected:
webrtc::AudioDeviceModule* audio_device_module() {
return static_cast<webrtc::AudioDeviceModule*>(audio_device_.get());
}
test::TaskEnvironment task_environment_;
scoped_refptr<blink::WebRtcAudioDeviceImpl> audio_device_;
std::unique_ptr<MockAudioTransport> audio_transport_;
};
// Verify that stats are accumulated during calls to RenderData and are
// available through GetStats().
TEST_F(WebRtcAudioDeviceImplTest, GetStats) {
auto audio_bus = media::AudioBus::Create(kAudioParameters);
int sample_rate = kAudioParameters.sample_rate();
auto audio_delay = base::Seconds(1);
base::TimeDelta current_time;
media::AudioGlitchInfo glitch_info;
glitch_info.duration = base::Seconds(2);
glitch_info.count = 3;
for (int i = 0; i < 10; i++) {
webrtc::AudioDeviceModule::Stats stats = *audio_device_->GetStats();
EXPECT_EQ(stats.synthesized_samples_duration_s,
(base::Seconds(2) * i).InSecondsF());
EXPECT_EQ(stats.synthesized_samples_events, 3ull * i);
EXPECT_EQ(stats.total_samples_count,
static_cast<uint64_t>(audio_bus->frames() * i));
EXPECT_EQ(stats.total_playout_delay_s,
(audio_bus->frames() * i * base::Seconds(1)).InSecondsF());
EXPECT_EQ(stats.total_samples_duration_s,
(media::AudioTimestampHelper::FramesToTime(audio_bus->frames(),
sample_rate) *
i)
.InSecondsF());
audio_device_->RenderData(audio_bus.get(), sample_rate, audio_delay,
¤t_time, glitch_info);
}
}
} // namespace blink
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