|  | /* | 
|  | *  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 <SLES/OpenSLES_Android.h> | 
|  |  | 
|  | #include "modules/audio_device/android/build_info.h" | 
|  | #include "modules/audio_device/android/ensure_initialized.h" | 
|  | #include "rtc_base/arraysize.h" | 
|  | #include "rtc_base/format_macros.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<AudioManager> 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: %" RTC_PRIuS "\n", kTag, playout_parameters_.channels()); | 
|  | PRINT("%sframes per buffer: %" RTC_PRIuS " <=> %.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: %" RTC_PRIuS "\n", kTag, record_parameters_.channels()); | 
|  | PRINT("%sframes per buffer: %" RTC_PRIuS " <=> %.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<size_t>(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 |