blob: bf79529365c4e7326ac78c621a7d9f53d1c30e2c [file] [log] [blame]
/*
* 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 "audio/audio_state.h"
#include <memory>
#include <vector>
#include "call/test/mock_audio_send_stream.h"
#include "modules/audio_device/include/mock_audio_device.h"
#include "modules/audio_mixer/audio_mixer_impl.h"
#include "modules/audio_processing/include/mock_audio_processing.h"
#include "rtc_base/ref_counted_object.h"
#include "test/gtest.h"
namespace webrtc {
namespace test {
namespace {
constexpr int kSampleRate = 16000;
constexpr int kNumberOfChannels = 1;
struct ConfigHelper {
ConfigHelper() : audio_mixer(AudioMixerImpl::Create()) {
audio_state_config.audio_mixer = audio_mixer;
audio_state_config.audio_processing =
new rtc::RefCountedObject<testing::NiceMock<MockAudioProcessing>>();
audio_state_config.audio_device_module =
new rtc::RefCountedObject<MockAudioDeviceModule>();
}
AudioState::Config& config() { return audio_state_config; }
rtc::scoped_refptr<AudioMixer> mixer() { return audio_mixer; }
private:
AudioState::Config audio_state_config;
rtc::scoped_refptr<AudioMixer> audio_mixer;
};
class FakeAudioSource : public AudioMixer::Source {
public:
// TODO(aleloi): Valid overrides commented out, because the gmock
// methods don't use any override declarations, and we want to avoid
// warnings from -Winconsistent-missing-override. See
// http://crbug.com/428099.
int Ssrc() const /*override*/ { return 0; }
int PreferredSampleRate() const /*override*/ { return kSampleRate; }
MOCK_METHOD2(GetAudioFrameWithInfo,
AudioFrameInfo(int sample_rate_hz, AudioFrame* audio_frame));
};
std::vector<int16_t> Create10msTestData(int sample_rate_hz,
size_t num_channels) {
const int samples_per_channel = sample_rate_hz / 100;
std::vector<int16_t> audio_data(samples_per_channel * num_channels, 0);
// Fill the first channel with a 1kHz sine wave.
const float inc = (2 * 3.14159265f * 1000) / sample_rate_hz;
float w = 0.f;
for (int i = 0; i < samples_per_channel; ++i) {
audio_data[i * num_channels] = static_cast<int16_t>(32767.f * std::sin(w));
w += inc;
}
return audio_data;
}
std::vector<uint32_t> ComputeChannelLevels(AudioFrame* audio_frame) {
const size_t num_channels = audio_frame->num_channels_;
const size_t samples_per_channel = audio_frame->samples_per_channel_;
std::vector<uint32_t> levels(num_channels, 0);
for (size_t i = 0; i < samples_per_channel; ++i) {
for (size_t j = 0; j < num_channels; ++j) {
levels[j] += std::abs(audio_frame->data()[i * num_channels + j]);
}
}
return levels;
}
} // namespace
TEST(AudioStateTest, Create) {
ConfigHelper helper;
auto audio_state = AudioState::Create(helper.config());
EXPECT_TRUE(audio_state.get());
}
TEST(AudioStateTest, ConstructDestruct) {
ConfigHelper helper;
rtc::scoped_refptr<internal::AudioState> audio_state(
new rtc::RefCountedObject<internal::AudioState>(helper.config()));
}
TEST(AudioStateTest, RecordedAudioArrivesAtSingleStream) {
ConfigHelper helper;
rtc::scoped_refptr<internal::AudioState> audio_state(
new rtc::RefCountedObject<internal::AudioState>(helper.config()));
MockAudioSendStream stream;
audio_state->AddSendingStream(&stream, 8000, 2);
EXPECT_CALL(
stream,
SendAudioDataForMock(::testing::AllOf(
::testing::Field(&AudioFrame::sample_rate_hz_, ::testing::Eq(8000)),
::testing::Field(&AudioFrame::num_channels_, ::testing::Eq(2u)))))
.WillOnce(
// Verify that channels are not swapped by default.
::testing::Invoke([](AudioFrame* audio_frame) {
auto levels = ComputeChannelLevels(audio_frame);
EXPECT_LT(0u, levels[0]);
EXPECT_EQ(0u, levels[1]);
}));
MockAudioProcessing* ap =
static_cast<MockAudioProcessing*>(audio_state->audio_processing());
EXPECT_CALL(*ap, set_stream_delay_ms(0));
EXPECT_CALL(*ap, set_stream_key_pressed(false));
EXPECT_CALL(*ap, ProcessStream(::testing::_));
constexpr int kSampleRate = 16000;
constexpr size_t kNumChannels = 2;
auto audio_data = Create10msTestData(kSampleRate, kNumChannels);
uint32_t new_mic_level = 667;
audio_state->audio_transport()->RecordedDataIsAvailable(
&audio_data[0], kSampleRate / 100, kNumChannels * 2, kNumChannels,
kSampleRate, 0, 0, 0, false, new_mic_level);
EXPECT_EQ(667u, new_mic_level);
audio_state->RemoveSendingStream(&stream);
}
TEST(AudioStateTest, RecordedAudioArrivesAtMultipleStreams) {
ConfigHelper helper;
rtc::scoped_refptr<internal::AudioState> audio_state(
new rtc::RefCountedObject<internal::AudioState>(helper.config()));
MockAudioSendStream stream_1;
MockAudioSendStream stream_2;
audio_state->AddSendingStream(&stream_1, 8001, 2);
audio_state->AddSendingStream(&stream_2, 32000, 1);
EXPECT_CALL(
stream_1,
SendAudioDataForMock(::testing::AllOf(
::testing::Field(&AudioFrame::sample_rate_hz_, ::testing::Eq(16000)),
::testing::Field(&AudioFrame::num_channels_, ::testing::Eq(1u)))))
.WillOnce(
// Verify that there is output signal.
::testing::Invoke([](AudioFrame* audio_frame) {
auto levels = ComputeChannelLevels(audio_frame);
EXPECT_LT(0u, levels[0]);
}));
EXPECT_CALL(
stream_2,
SendAudioDataForMock(::testing::AllOf(
::testing::Field(&AudioFrame::sample_rate_hz_, ::testing::Eq(16000)),
::testing::Field(&AudioFrame::num_channels_, ::testing::Eq(1u)))))
.WillOnce(
// Verify that there is output signal.
::testing::Invoke([](AudioFrame* audio_frame) {
auto levels = ComputeChannelLevels(audio_frame);
EXPECT_LT(0u, levels[0]);
}));
MockAudioProcessing* ap =
static_cast<MockAudioProcessing*>(audio_state->audio_processing());
EXPECT_CALL(*ap, set_stream_delay_ms(5));
EXPECT_CALL(*ap, set_stream_key_pressed(true));
EXPECT_CALL(*ap, ProcessStream(::testing::_));
constexpr int kSampleRate = 16000;
constexpr size_t kNumChannels = 1;
auto audio_data = Create10msTestData(kSampleRate, kNumChannels);
uint32_t new_mic_level = 667;
audio_state->audio_transport()->RecordedDataIsAvailable(
&audio_data[0], kSampleRate / 100, kNumChannels * 2, kNumChannels,
kSampleRate, 5, 0, 0, true, new_mic_level);
EXPECT_EQ(667u, new_mic_level);
audio_state->RemoveSendingStream(&stream_1);
audio_state->RemoveSendingStream(&stream_2);
}
TEST(AudioStateTest, EnableChannelSwap) {
constexpr int kSampleRate = 16000;
constexpr size_t kNumChannels = 2;
ConfigHelper helper;
rtc::scoped_refptr<internal::AudioState> audio_state(
new rtc::RefCountedObject<internal::AudioState>(helper.config()));
audio_state->SetStereoChannelSwapping(true);
MockAudioSendStream stream;
audio_state->AddSendingStream(&stream, kSampleRate, kNumChannels);
EXPECT_CALL(stream, SendAudioDataForMock(::testing::_))
.WillOnce(
// Verify that channels are swapped.
::testing::Invoke([](AudioFrame* audio_frame) {
auto levels = ComputeChannelLevels(audio_frame);
EXPECT_EQ(0u, levels[0]);
EXPECT_LT(0u, levels[1]);
}));
auto audio_data = Create10msTestData(kSampleRate, kNumChannels);
uint32_t new_mic_level = 667;
audio_state->audio_transport()->RecordedDataIsAvailable(
&audio_data[0], kSampleRate / 100, kNumChannels * 2, kNumChannels,
kSampleRate, 0, 0, 0, false, new_mic_level);
EXPECT_EQ(667u, new_mic_level);
audio_state->RemoveSendingStream(&stream);
}
TEST(AudioStateTest,
QueryingTransportForAudioShouldResultInGetAudioCallOnMixerSource) {
ConfigHelper helper;
auto audio_state = AudioState::Create(helper.config());
FakeAudioSource fake_source;
helper.mixer()->AddSource(&fake_source);
EXPECT_CALL(fake_source, GetAudioFrameWithInfo(::testing::_, ::testing::_))
.WillOnce(
::testing::Invoke([](int sample_rate_hz, AudioFrame* audio_frame) {
audio_frame->sample_rate_hz_ = sample_rate_hz;
audio_frame->samples_per_channel_ = sample_rate_hz / 100;
audio_frame->num_channels_ = kNumberOfChannels;
return AudioMixer::Source::AudioFrameInfo::kNormal;
}));
int16_t audio_buffer[kSampleRate / 100 * kNumberOfChannels];
size_t n_samples_out;
int64_t elapsed_time_ms;
int64_t ntp_time_ms;
audio_state->audio_transport()->NeedMorePlayData(
kSampleRate / 100, kNumberOfChannels * 2, kNumberOfChannels, kSampleRate,
audio_buffer, n_samples_out, &elapsed_time_ms, &ntp_time_ms);
}
} // namespace test
} // namespace webrtc