/* * 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 "modules/audio_device/android/audio_manager.h" #include #include "modules/audio_device/android/build_info.h" #include "modules/audio_device/android/ensure_initialized.h" #include "rtc_base/arraysize.h" #include "test/gtest.h" #define PRINT(...) fprintf(stderr, __VA_ARGS__); namespace webrtc { static const char kTag[] = " "; class AudioManagerTest : public ::testing::Test { protected: AudioManagerTest() { // One-time initialization of JVM and application context. Ensures that we // can do calls between C++ and Java. webrtc::audiodevicemodule::EnsureInitialized(); audio_manager_.reset(new AudioManager()); SetActiveAudioLayer(); playout_parameters_ = audio_manager()->GetPlayoutAudioParameters(); record_parameters_ = audio_manager()->GetRecordAudioParameters(); } AudioManager* audio_manager() const { return audio_manager_.get(); } // A valid audio layer must always be set before calling Init(), hence we // might as well make it a part of the test fixture. void SetActiveAudioLayer() { EXPECT_EQ(0, audio_manager()->GetDelayEstimateInMilliseconds()); audio_manager()->SetActiveAudioLayer(AudioDeviceModule::kAndroidJavaAudio); EXPECT_NE(0, audio_manager()->GetDelayEstimateInMilliseconds()); } // One way to ensure that the engine object is valid is to create an // SL Engine interface since it exposes creation methods of all the OpenSL ES // object types and it is only supported on the engine object. This method // also verifies that the engine interface supports at least one interface. // Note that, the test below is not a full test of the SLEngineItf object // but only a simple sanity test to check that the global engine object is OK. void ValidateSLEngine(SLObjectItf engine_object) { EXPECT_NE(nullptr, engine_object); // Get the SL Engine interface which is exposed by the engine object. SLEngineItf engine; SLresult result = (*engine_object)->GetInterface(engine_object, SL_IID_ENGINE, &engine); EXPECT_EQ(result, SL_RESULT_SUCCESS) << "GetInterface() on engine failed"; // Ensure that the SL Engine interface exposes at least one interface. SLuint32 object_id = SL_OBJECTID_ENGINE; SLuint32 num_supported_interfaces = 0; result = (*engine)->QueryNumSupportedInterfaces(engine, object_id, &num_supported_interfaces); EXPECT_EQ(result, SL_RESULT_SUCCESS) << "QueryNumSupportedInterfaces() failed"; EXPECT_GE(num_supported_interfaces, 1u); } std::unique_ptr audio_manager_; AudioParameters playout_parameters_; AudioParameters record_parameters_; }; TEST_F(AudioManagerTest, ConstructDestruct) {} // It should not be possible to create an OpenSL engine object if Java based // audio is requested in both directions. TEST_F(AudioManagerTest, GetOpenSLEngineShouldFailForJavaAudioLayer) { audio_manager()->SetActiveAudioLayer(AudioDeviceModule::kAndroidJavaAudio); SLObjectItf engine_object = audio_manager()->GetOpenSLEngine(); EXPECT_EQ(nullptr, engine_object); } // It should be possible to create an OpenSL engine object if OpenSL ES based // audio is requested in any direction. TEST_F(AudioManagerTest, GetOpenSLEngineShouldSucceedForOpenSLESAudioLayer) { // List of supported audio layers that uses OpenSL ES audio. const AudioDeviceModule::AudioLayer opensles_audio[] = { AudioDeviceModule::kAndroidOpenSLESAudio, AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio}; // Verify that the global (singleton) OpenSL Engine can be acquired for all // audio layes that uses OpenSL ES. Note that the engine is only created once. for (const AudioDeviceModule::AudioLayer audio_layer : opensles_audio) { audio_manager()->SetActiveAudioLayer(audio_layer); SLObjectItf engine_object = audio_manager()->GetOpenSLEngine(); EXPECT_NE(nullptr, engine_object); // Perform a simple sanity check of the created engine object. ValidateSLEngine(engine_object); } } TEST_F(AudioManagerTest, InitClose) { EXPECT_TRUE(audio_manager()->Init()); EXPECT_TRUE(audio_manager()->Close()); } TEST_F(AudioManagerTest, IsAcousticEchoCancelerSupported) { PRINT("%sAcoustic Echo Canceler support: %s\n", kTag, audio_manager()->IsAcousticEchoCancelerSupported() ? "Yes" : "No"); } TEST_F(AudioManagerTest, IsAutomaticGainControlSupported) { EXPECT_FALSE(audio_manager()->IsAutomaticGainControlSupported()); } TEST_F(AudioManagerTest, IsNoiseSuppressorSupported) { PRINT("%sNoise Suppressor support: %s\n", kTag, audio_manager()->IsNoiseSuppressorSupported() ? "Yes" : "No"); } TEST_F(AudioManagerTest, IsLowLatencyPlayoutSupported) { PRINT("%sLow latency output support: %s\n", kTag, audio_manager()->IsLowLatencyPlayoutSupported() ? "Yes" : "No"); } TEST_F(AudioManagerTest, IsLowLatencyRecordSupported) { PRINT("%sLow latency input support: %s\n", kTag, audio_manager()->IsLowLatencyRecordSupported() ? "Yes" : "No"); } TEST_F(AudioManagerTest, IsProAudioSupported) { PRINT("%sPro audio support: %s\n", kTag, audio_manager()->IsProAudioSupported() ? "Yes" : "No"); } // Verify that playout side is configured for mono by default. TEST_F(AudioManagerTest, IsStereoPlayoutSupported) { EXPECT_FALSE(audio_manager()->IsStereoPlayoutSupported()); } // Verify that recording side is configured for mono by default. TEST_F(AudioManagerTest, IsStereoRecordSupported) { EXPECT_FALSE(audio_manager()->IsStereoRecordSupported()); } TEST_F(AudioManagerTest, ShowAudioParameterInfo) { const bool low_latency_out = audio_manager()->IsLowLatencyPlayoutSupported(); const bool low_latency_in = audio_manager()->IsLowLatencyRecordSupported(); PRINT("PLAYOUT:\n"); PRINT("%saudio layer: %s\n", kTag, low_latency_out ? "Low latency OpenSL" : "Java/JNI based AudioTrack"); PRINT("%ssample rate: %d Hz\n", kTag, playout_parameters_.sample_rate()); PRINT("%schannels: %zu\n", kTag, playout_parameters_.channels()); PRINT("%sframes per buffer: %zu <=> %.2f ms\n", kTag, playout_parameters_.frames_per_buffer(), playout_parameters_.GetBufferSizeInMilliseconds()); PRINT("RECORD: \n"); PRINT("%saudio layer: %s\n", kTag, low_latency_in ? "Low latency OpenSL" : "Java/JNI based AudioRecord"); PRINT("%ssample rate: %d Hz\n", kTag, record_parameters_.sample_rate()); PRINT("%schannels: %zu\n", kTag, record_parameters_.channels()); PRINT("%sframes per buffer: %zu <=> %.2f ms\n", kTag, record_parameters_.frames_per_buffer(), record_parameters_.GetBufferSizeInMilliseconds()); } // The audio device module only suppors the same sample rate in both directions. // In addition, in full-duplex low-latency mode (OpenSL ES), both input and // output must use the same native buffer size to allow for usage of the fast // audio track in Android. TEST_F(AudioManagerTest, VerifyAudioParameters) { const bool low_latency_out = audio_manager()->IsLowLatencyPlayoutSupported(); const bool low_latency_in = audio_manager()->IsLowLatencyRecordSupported(); EXPECT_EQ(playout_parameters_.sample_rate(), record_parameters_.sample_rate()); if (low_latency_out && low_latency_in) { EXPECT_EQ(playout_parameters_.frames_per_buffer(), record_parameters_.frames_per_buffer()); } } // Add device-specific information to the test for logging purposes. TEST_F(AudioManagerTest, ShowDeviceInfo) { BuildInfo build_info; PRINT("%smodel: %s\n", kTag, build_info.GetDeviceModel().c_str()); PRINT("%sbrand: %s\n", kTag, build_info.GetBrand().c_str()); PRINT("%smanufacturer: %s\n", kTag, build_info.GetDeviceManufacturer().c_str()); } // Add Android build information to the test for logging purposes. TEST_F(AudioManagerTest, ShowBuildInfo) { BuildInfo build_info; PRINT("%sbuild release: %s\n", kTag, build_info.GetBuildRelease().c_str()); PRINT("%sbuild id: %s\n", kTag, build_info.GetAndroidBuildId().c_str()); PRINT("%sbuild type: %s\n", kTag, build_info.GetBuildType().c_str()); PRINT("%sSDK version: %d\n", kTag, build_info.GetSdkVersion()); } // Basic test of the AudioParameters class using default construction where // all members are set to zero. TEST_F(AudioManagerTest, AudioParametersWithDefaultConstruction) { AudioParameters params; EXPECT_FALSE(params.is_valid()); EXPECT_EQ(0, params.sample_rate()); EXPECT_EQ(0U, params.channels()); EXPECT_EQ(0U, params.frames_per_buffer()); EXPECT_EQ(0U, params.frames_per_10ms_buffer()); EXPECT_EQ(0U, params.GetBytesPerFrame()); EXPECT_EQ(0U, params.GetBytesPerBuffer()); EXPECT_EQ(0U, params.GetBytesPer10msBuffer()); EXPECT_EQ(0.0f, params.GetBufferSizeInMilliseconds()); } // Basic test of the AudioParameters class using non default construction. TEST_F(AudioManagerTest, AudioParametersWithNonDefaultConstruction) { const int kSampleRate = 48000; const size_t kChannels = 1; const size_t kFramesPerBuffer = 480; const size_t kFramesPer10msBuffer = 480; const size_t kBytesPerFrame = 2; const float kBufferSizeInMs = 10.0f; AudioParameters params(kSampleRate, kChannels, kFramesPerBuffer); EXPECT_TRUE(params.is_valid()); EXPECT_EQ(kSampleRate, params.sample_rate()); EXPECT_EQ(kChannels, params.channels()); EXPECT_EQ(kFramesPerBuffer, params.frames_per_buffer()); EXPECT_EQ(static_cast(kSampleRate / 100), params.frames_per_10ms_buffer()); EXPECT_EQ(kBytesPerFrame, params.GetBytesPerFrame()); EXPECT_EQ(kBytesPerFrame * kFramesPerBuffer, params.GetBytesPerBuffer()); EXPECT_EQ(kBytesPerFrame * kFramesPer10msBuffer, params.GetBytesPer10msBuffer()); EXPECT_EQ(kBufferSizeInMs, params.GetBufferSizeInMilliseconds()); } } // namespace webrtc