blob: e82376d7ef5e61204c4424aad0489f13fab51f14 [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 <SLES/OpenSLES_Android.h>
#include "webrtc/modules/audio_device/android/audio_manager.h"
#include "webrtc/modules/audio_device/android/build_info.h"
#include "webrtc/modules/audio_device/android/ensure_initialized.h"
#include "webrtc/rtc_base/arraysize.h"
#include "webrtc/rtc_base/format_macros.h"
#include "webrtc/test/gtest.h"
#define PRINT(...) fprintf(stderr, __VA_ARGS__);
namespace webrtc {
static const char kTag[] = " ";
class AudioManagerTest : public ::testing::Test {
AudioManagerTest() {
// One-time initialization of JVM and application context. Ensures that we
// can do calls between C++ and Java.
audio_manager_.reset(new AudioManager());
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());
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,
<< "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) {
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[] = {
// 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) {
SLObjectItf engine_object = audio_manager()->GetOpenSLEngine();
EXPECT_NE(nullptr, engine_object);
// Perform a simple sanity check of the created engine object.
TEST_F(AudioManagerTest, InitClose) {
TEST_F(AudioManagerTest, IsAcousticEchoCancelerSupported) {
PRINT("%sAcoustic Echo Canceler support: %s\n", kTag,
audio_manager()->IsAcousticEchoCancelerSupported() ? "Yes" : "No");
TEST_F(AudioManagerTest, 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) {
// Verify that recording side is configured for mono by default.
TEST_F(AudioManagerTest, IsStereoRecordSupported) {
TEST_F(AudioManagerTest, ShowAudioParameterInfo) {
const bool low_latency_out = audio_manager()->IsLowLatencyPlayoutSupported();
const bool low_latency_in = audio_manager()->IsLowLatencyRecordSupported();
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: %" PRIuS "\n", kTag, playout_parameters_.channels());
PRINT("%sframes per buffer: %" PRIuS " <=> %.2f ms\n", kTag,
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: %" PRIuS "\n", kTag, record_parameters_.channels());
PRINT("%sframes per buffer: %" PRIuS " <=> %.2f ms\n", kTag,
// 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();
if (low_latency_out && low_latency_in) {
// 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_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_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),
EXPECT_EQ(kBytesPerFrame, params.GetBytesPerFrame());
EXPECT_EQ(kBytesPerFrame * kFramesPerBuffer, params.GetBytesPerBuffer());
EXPECT_EQ(kBytesPerFrame * kFramesPer10msBuffer,
EXPECT_EQ(kBufferSizeInMs, params.GetBufferSizeInMilliseconds());
} // namespace webrtc