| /* | 
 |  *  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 <memory> | 
 | #include <vector> | 
 |  | 
 | #include "audio/audio_state.h" | 
 | #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 "test/gtest.h" | 
 | #include "test/mock_voice_engine.h" | 
 |  | 
 | namespace webrtc { | 
 | namespace test { | 
 | namespace { | 
 |  | 
 | constexpr int kSampleRate = 16000; | 
 | constexpr int kNumberOfChannels = 1; | 
 |  | 
 | struct ConfigHelper { | 
 |   ConfigHelper() : audio_mixer(AudioMixerImpl::Create()) { | 
 |     audio_state_config.voice_engine = &mock_voice_engine; | 
 |     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; } | 
 |   MockVoiceEngine& voice_engine() { return mock_voice_engine; } | 
 |   rtc::scoped_refptr<AudioMixer> mixer() { return audio_mixer; } | 
 |  | 
 |  private: | 
 |   testing::StrictMock<MockVoiceEngine> mock_voice_engine; | 
 |   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> Create10msSilentTestData(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); | 
 |   return audio_data; | 
 | } | 
 |  | 
 | 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; | 
 |   std::unique_ptr<internal::AudioState> audio_state( | 
 |       new internal::AudioState(helper.config())); | 
 | } | 
 |  | 
 | TEST(AudioStateTest, GetVoiceEngine) { | 
 |   ConfigHelper helper; | 
 |   std::unique_ptr<internal::AudioState> audio_state( | 
 |       new internal::AudioState(helper.config())); | 
 |   EXPECT_EQ(audio_state->voice_engine(), &helper.voice_engine()); | 
 | } | 
 |  | 
 | TEST(AudioStateTest, RecordedAudioArrivesAtSingleStream) { | 
 |   ConfigHelper helper; | 
 |   std::unique_ptr<internal::AudioState> audio_state( | 
 |       new 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; | 
 |   std::unique_ptr<internal::AudioState> audio_state( | 
 |       new 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; | 
 |   std::unique_ptr<internal::AudioState> audio_state( | 
 |       new 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, InputLevelStats) { | 
 |   constexpr int kSampleRate = 16000; | 
 |   constexpr size_t kNumChannels = 1; | 
 |  | 
 |   ConfigHelper helper; | 
 |   std::unique_ptr<internal::AudioState> audio_state( | 
 |       new internal::AudioState(helper.config())); | 
 |  | 
 |   // Push a silent buffer -> Level stats should be zeros except for duration. | 
 |   { | 
 |     auto audio_data = Create10msSilentTestData(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); | 
 |     auto stats = audio_state->GetAudioInputStats(); | 
 |     EXPECT_EQ(0, stats.audio_level); | 
 |     EXPECT_EQ(0, stats.quantized_audio_level); | 
 |     EXPECT_THAT(stats.total_energy, testing::DoubleEq(0.0)); | 
 |     EXPECT_THAT(stats.total_duration, testing::DoubleEq(0.01)); | 
 |   } | 
 |  | 
 |   // Push 10 non-silent buffers -> Level stats should be non-zero. | 
 |   { | 
 |     auto audio_data = Create10msTestData(kSampleRate, kNumChannels); | 
 |     uint32_t new_mic_level = 667; | 
 |     for (int i = 0; i < 10; ++i) { | 
 |       audio_state->audio_transport()->RecordedDataIsAvailable( | 
 |           &audio_data[0], kSampleRate / 100, kNumChannels * 2, | 
 |           kNumChannels, kSampleRate, 0, 0, 0, false, new_mic_level); | 
 |     } | 
 |     auto stats = audio_state->GetAudioInputStats(); | 
 |     EXPECT_EQ(32767, stats.audio_level); | 
 |     EXPECT_EQ(9, stats.quantized_audio_level); | 
 |     EXPECT_THAT(stats.total_energy, testing::DoubleEq(0.01)); | 
 |     EXPECT_THAT(stats.total_duration, testing::DoubleEq(0.11)); | 
 |   } | 
 | } | 
 |  | 
 | 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 |