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Diffstat (limited to 'webrtc/modules/audio_device/android/audio_device_unittest.cc')
| -rw-r--r-- | webrtc/modules/audio_device/android/audio_device_unittest.cc | 1018 |
1 files changed, 1018 insertions, 0 deletions
diff --git a/webrtc/modules/audio_device/android/audio_device_unittest.cc b/webrtc/modules/audio_device/android/audio_device_unittest.cc new file mode 100644 index 0000000000..7b2d6354c4 --- /dev/null +++ b/webrtc/modules/audio_device/android/audio_device_unittest.cc @@ -0,0 +1,1018 @@ +/* + * 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 <algorithm> +#include <limits> +#include <list> +#include <numeric> +#include <string> +#include <vector> + +#include "testing/gmock/include/gmock/gmock.h" +#include "testing/gtest/include/gtest/gtest.h" +#include "webrtc/base/arraysize.h" +#include "webrtc/base/criticalsection.h" +#include "webrtc/base/format_macros.h" +#include "webrtc/base/scoped_ptr.h" +#include "webrtc/base/scoped_ref_ptr.h" +#include "webrtc/modules/audio_device/android/audio_common.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/modules/audio_device/audio_device_impl.h" +#include "webrtc/modules/audio_device/include/audio_device.h" +#include "webrtc/system_wrappers/include/clock.h" +#include "webrtc/system_wrappers/include/event_wrapper.h" +#include "webrtc/system_wrappers/include/sleep.h" +#include "webrtc/test/testsupport/fileutils.h" + +using std::cout; +using std::endl; +using ::testing::_; +using ::testing::AtLeast; +using ::testing::Gt; +using ::testing::Invoke; +using ::testing::NiceMock; +using ::testing::NotNull; +using ::testing::Return; +using ::testing::TestWithParam; + +// #define ENABLE_DEBUG_PRINTF +#ifdef ENABLE_DEBUG_PRINTF +#define PRINTD(...) fprintf(stderr, __VA_ARGS__); +#else +#define PRINTD(...) ((void)0) +#endif +#define PRINT(...) fprintf(stderr, __VA_ARGS__); + +namespace webrtc { + +// Number of callbacks (input or output) the tests waits for before we set +// an event indicating that the test was OK. +static const size_t kNumCallbacks = 10; +// Max amount of time we wait for an event to be set while counting callbacks. +static const int kTestTimeOutInMilliseconds = 10 * 1000; +// Average number of audio callbacks per second assuming 10ms packet size. +static const size_t kNumCallbacksPerSecond = 100; +// Play out a test file during this time (unit is in seconds). +static const int kFilePlayTimeInSec = 5; +static const size_t kBitsPerSample = 16; +static const size_t kBytesPerSample = kBitsPerSample / 8; +// Run the full-duplex test during this time (unit is in seconds). +// Note that first |kNumIgnoreFirstCallbacks| are ignored. +static const int kFullDuplexTimeInSec = 5; +// Wait for the callback sequence to stabilize by ignoring this amount of the +// initial callbacks (avoids initial FIFO access). +// Only used in the RunPlayoutAndRecordingInFullDuplex test. +static const size_t kNumIgnoreFirstCallbacks = 50; +// Sets the number of impulses per second in the latency test. +static const int kImpulseFrequencyInHz = 1; +// Length of round-trip latency measurements. Number of transmitted impulses +// is kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1. +static const int kMeasureLatencyTimeInSec = 11; +// Utilized in round-trip latency measurements to avoid capturing noise samples. +static const int kImpulseThreshold = 1000; +static const char kTag[] = "[..........] "; + +enum TransportType { + kPlayout = 0x1, + kRecording = 0x2, +}; + +// Interface for processing the audio stream. Real implementations can e.g. +// run audio in loopback, read audio from a file or perform latency +// measurements. +class AudioStreamInterface { + public: + virtual void Write(const void* source, size_t num_frames) = 0; + virtual void Read(void* destination, size_t num_frames) = 0; + protected: + virtual ~AudioStreamInterface() {} +}; + +// Reads audio samples from a PCM file where the file is stored in memory at +// construction. +class FileAudioStream : public AudioStreamInterface { + public: + FileAudioStream( + size_t num_callbacks, const std::string& file_name, int sample_rate) + : file_size_in_bytes_(0), + sample_rate_(sample_rate), + file_pos_(0) { + file_size_in_bytes_ = test::GetFileSize(file_name); + sample_rate_ = sample_rate; + EXPECT_GE(file_size_in_callbacks(), num_callbacks) + << "Size of test file is not large enough to last during the test."; + const size_t num_16bit_samples = + test::GetFileSize(file_name) / kBytesPerSample; + file_.reset(new int16_t[num_16bit_samples]); + FILE* audio_file = fopen(file_name.c_str(), "rb"); + EXPECT_NE(audio_file, nullptr); + size_t num_samples_read = fread( + file_.get(), sizeof(int16_t), num_16bit_samples, audio_file); + EXPECT_EQ(num_samples_read, num_16bit_samples); + fclose(audio_file); + } + + // AudioStreamInterface::Write() is not implemented. + void Write(const void* source, size_t num_frames) override {} + + // Read samples from file stored in memory (at construction) and copy + // |num_frames| (<=> 10ms) to the |destination| byte buffer. + void Read(void* destination, size_t num_frames) override { + memcpy(destination, + static_cast<int16_t*> (&file_[file_pos_]), + num_frames * sizeof(int16_t)); + file_pos_ += num_frames; + } + + int file_size_in_seconds() const { + return static_cast<int>( + file_size_in_bytes_ / (kBytesPerSample * sample_rate_)); + } + size_t file_size_in_callbacks() const { + return file_size_in_seconds() * kNumCallbacksPerSecond; + } + + private: + size_t file_size_in_bytes_; + int sample_rate_; + rtc::scoped_ptr<int16_t[]> file_; + size_t file_pos_; +}; + +// Simple first in first out (FIFO) class that wraps a list of 16-bit audio +// buffers of fixed size and allows Write and Read operations. The idea is to +// store recorded audio buffers (using Write) and then read (using Read) these +// stored buffers with as short delay as possible when the audio layer needs +// data to play out. The number of buffers in the FIFO will stabilize under +// normal conditions since there will be a balance between Write and Read calls. +// The container is a std::list container and access is protected with a lock +// since both sides (playout and recording) are driven by its own thread. +class FifoAudioStream : public AudioStreamInterface { + public: + explicit FifoAudioStream(size_t frames_per_buffer) + : frames_per_buffer_(frames_per_buffer), + bytes_per_buffer_(frames_per_buffer_ * sizeof(int16_t)), + fifo_(new AudioBufferList), + largest_size_(0), + total_written_elements_(0), + write_count_(0) { + EXPECT_NE(fifo_.get(), nullptr); + } + + ~FifoAudioStream() { + Flush(); + } + + // Allocate new memory, copy |num_frames| samples from |source| into memory + // and add pointer to the memory location to end of the list. + // Increases the size of the FIFO by one element. + void Write(const void* source, size_t num_frames) override { + ASSERT_EQ(num_frames, frames_per_buffer_); + PRINTD("+"); + if (write_count_++ < kNumIgnoreFirstCallbacks) { + return; + } + int16_t* memory = new int16_t[frames_per_buffer_]; + memcpy(static_cast<int16_t*> (&memory[0]), + source, + bytes_per_buffer_); + rtc::CritScope lock(&lock_); + fifo_->push_back(memory); + const size_t size = fifo_->size(); + if (size > largest_size_) { + largest_size_ = size; + PRINTD("(%" PRIuS ")", largest_size_); + } + total_written_elements_ += size; + } + + // Read pointer to data buffer from front of list, copy |num_frames| of stored + // data into |destination| and delete the utilized memory allocation. + // Decreases the size of the FIFO by one element. + void Read(void* destination, size_t num_frames) override { + ASSERT_EQ(num_frames, frames_per_buffer_); + PRINTD("-"); + rtc::CritScope lock(&lock_); + if (fifo_->empty()) { + memset(destination, 0, bytes_per_buffer_); + } else { + int16_t* memory = fifo_->front(); + fifo_->pop_front(); + memcpy(destination, + static_cast<int16_t*> (&memory[0]), + bytes_per_buffer_); + delete memory; + } + } + + size_t size() const { + return fifo_->size(); + } + + size_t largest_size() const { + return largest_size_; + } + + size_t average_size() const { + return (total_written_elements_ == 0) ? 0.0 : 0.5 + static_cast<float> ( + total_written_elements_) / (write_count_ - kNumIgnoreFirstCallbacks); + } + + private: + void Flush() { + for (auto it = fifo_->begin(); it != fifo_->end(); ++it) { + delete *it; + } + fifo_->clear(); + } + + using AudioBufferList = std::list<int16_t*>; + rtc::CriticalSection lock_; + const size_t frames_per_buffer_; + const size_t bytes_per_buffer_; + rtc::scoped_ptr<AudioBufferList> fifo_; + size_t largest_size_; + size_t total_written_elements_; + size_t write_count_; +}; + +// Inserts periodic impulses and measures the latency between the time of +// transmission and time of receiving the same impulse. +// Usage requires a special hardware called Audio Loopback Dongle. +// See http://source.android.com/devices/audio/loopback.html for details. +class LatencyMeasuringAudioStream : public AudioStreamInterface { + public: + explicit LatencyMeasuringAudioStream(size_t frames_per_buffer) + : clock_(Clock::GetRealTimeClock()), + frames_per_buffer_(frames_per_buffer), + bytes_per_buffer_(frames_per_buffer_ * sizeof(int16_t)), + play_count_(0), + rec_count_(0), + pulse_time_(0) { + } + + // Insert periodic impulses in first two samples of |destination|. + void Read(void* destination, size_t num_frames) override { + ASSERT_EQ(num_frames, frames_per_buffer_); + if (play_count_ == 0) { + PRINT("["); + } + play_count_++; + memset(destination, 0, bytes_per_buffer_); + if (play_count_ % (kNumCallbacksPerSecond / kImpulseFrequencyInHz) == 0) { + if (pulse_time_ == 0) { + pulse_time_ = clock_->TimeInMilliseconds(); + } + PRINT("."); + const int16_t impulse = std::numeric_limits<int16_t>::max(); + int16_t* ptr16 = static_cast<int16_t*> (destination); + for (size_t i = 0; i < 2; ++i) { + ptr16[i] = impulse; + } + } + } + + // Detect received impulses in |source|, derive time between transmission and + // detection and add the calculated delay to list of latencies. + void Write(const void* source, size_t num_frames) override { + ASSERT_EQ(num_frames, frames_per_buffer_); + rec_count_++; + if (pulse_time_ == 0) { + // Avoid detection of new impulse response until a new impulse has + // been transmitted (sets |pulse_time_| to value larger than zero). + return; + } + const int16_t* ptr16 = static_cast<const int16_t*> (source); + std::vector<int16_t> vec(ptr16, ptr16 + num_frames); + // Find max value in the audio buffer. + int max = *std::max_element(vec.begin(), vec.end()); + // Find index (element position in vector) of the max element. + int index_of_max = std::distance(vec.begin(), + std::find(vec.begin(), vec.end(), + max)); + if (max > kImpulseThreshold) { + PRINTD("(%d,%d)", max, index_of_max); + int64_t now_time = clock_->TimeInMilliseconds(); + int extra_delay = IndexToMilliseconds(static_cast<double> (index_of_max)); + PRINTD("[%d]", static_cast<int> (now_time - pulse_time_)); + PRINTD("[%d]", extra_delay); + // Total latency is the difference between transmit time and detection + // tome plus the extra delay within the buffer in which we detected the + // received impulse. It is transmitted at sample 0 but can be received + // at sample N where N > 0. The term |extra_delay| accounts for N and it + // is a value between 0 and 10ms. + latencies_.push_back(now_time - pulse_time_ + extra_delay); + pulse_time_ = 0; + } else { + PRINTD("-"); + } + } + + size_t num_latency_values() const { + return latencies_.size(); + } + + int min_latency() const { + if (latencies_.empty()) + return 0; + return *std::min_element(latencies_.begin(), latencies_.end()); + } + + int max_latency() const { + if (latencies_.empty()) + return 0; + return *std::max_element(latencies_.begin(), latencies_.end()); + } + + int average_latency() const { + if (latencies_.empty()) + return 0; + return 0.5 + static_cast<double> ( + std::accumulate(latencies_.begin(), latencies_.end(), 0)) / + latencies_.size(); + } + + void PrintResults() const { + PRINT("] "); + for (auto it = latencies_.begin(); it != latencies_.end(); ++it) { + PRINT("%d ", *it); + } + PRINT("\n"); + PRINT("%s[min, max, avg]=[%d, %d, %d] ms\n", kTag, + min_latency(), max_latency(), average_latency()); + } + + int IndexToMilliseconds(double index) const { + return static_cast<int>(10.0 * (index / frames_per_buffer_) + 0.5); + } + + private: + Clock* clock_; + const size_t frames_per_buffer_; + const size_t bytes_per_buffer_; + size_t play_count_; + size_t rec_count_; + int64_t pulse_time_; + std::vector<int> latencies_; +}; + +// Mocks the AudioTransport object and proxies actions for the two callbacks +// (RecordedDataIsAvailable and NeedMorePlayData) to different implementations +// of AudioStreamInterface. +class MockAudioTransport : public AudioTransport { + public: + explicit MockAudioTransport(int type) + : num_callbacks_(0), + type_(type), + play_count_(0), + rec_count_(0), + audio_stream_(nullptr) {} + + virtual ~MockAudioTransport() {} + + MOCK_METHOD10(RecordedDataIsAvailable, + int32_t(const void* audioSamples, + const size_t nSamples, + const size_t nBytesPerSample, + const uint8_t nChannels, + const uint32_t samplesPerSec, + const uint32_t totalDelayMS, + const int32_t clockDrift, + const uint32_t currentMicLevel, + const bool keyPressed, + uint32_t& newMicLevel)); + MOCK_METHOD8(NeedMorePlayData, + int32_t(const size_t nSamples, + const size_t nBytesPerSample, + const uint8_t nChannels, + const uint32_t samplesPerSec, + void* audioSamples, + size_t& nSamplesOut, + int64_t* elapsed_time_ms, + int64_t* ntp_time_ms)); + + // Set default actions of the mock object. We are delegating to fake + // implementations (of AudioStreamInterface) here. + void HandleCallbacks(EventWrapper* test_is_done, + AudioStreamInterface* audio_stream, + int num_callbacks) { + test_is_done_ = test_is_done; + audio_stream_ = audio_stream; + num_callbacks_ = num_callbacks; + if (play_mode()) { + ON_CALL(*this, NeedMorePlayData(_, _, _, _, _, _, _, _)) + .WillByDefault( + Invoke(this, &MockAudioTransport::RealNeedMorePlayData)); + } + if (rec_mode()) { + ON_CALL(*this, RecordedDataIsAvailable(_, _, _, _, _, _, _, _, _, _)) + .WillByDefault( + Invoke(this, &MockAudioTransport::RealRecordedDataIsAvailable)); + } + } + + int32_t RealRecordedDataIsAvailable(const void* audioSamples, + const size_t nSamples, + const size_t nBytesPerSample, + const uint8_t nChannels, + const uint32_t samplesPerSec, + const uint32_t totalDelayMS, + const int32_t clockDrift, + const uint32_t currentMicLevel, + const bool keyPressed, + uint32_t& newMicLevel) { + EXPECT_TRUE(rec_mode()) << "No test is expecting these callbacks."; + rec_count_++; + // Process the recorded audio stream if an AudioStreamInterface + // implementation exists. + if (audio_stream_) { + audio_stream_->Write(audioSamples, nSamples); + } + if (ReceivedEnoughCallbacks()) { + test_is_done_->Set(); + } + return 0; + } + + int32_t RealNeedMorePlayData(const size_t nSamples, + const size_t nBytesPerSample, + const uint8_t nChannels, + const uint32_t samplesPerSec, + void* audioSamples, + size_t& nSamplesOut, + int64_t* elapsed_time_ms, + int64_t* ntp_time_ms) { + EXPECT_TRUE(play_mode()) << "No test is expecting these callbacks."; + play_count_++; + nSamplesOut = nSamples; + // Read (possibly processed) audio stream samples to be played out if an + // AudioStreamInterface implementation exists. + if (audio_stream_) { + audio_stream_->Read(audioSamples, nSamples); + } + if (ReceivedEnoughCallbacks()) { + test_is_done_->Set(); + } + return 0; + } + + bool ReceivedEnoughCallbacks() { + bool recording_done = false; + if (rec_mode()) + recording_done = rec_count_ >= num_callbacks_; + else + recording_done = true; + + bool playout_done = false; + if (play_mode()) + playout_done = play_count_ >= num_callbacks_; + else + playout_done = true; + + return recording_done && playout_done; + } + + bool play_mode() const { return type_ & kPlayout; } + bool rec_mode() const { return type_ & kRecording; } + + private: + EventWrapper* test_is_done_; + size_t num_callbacks_; + int type_; + size_t play_count_; + size_t rec_count_; + AudioStreamInterface* audio_stream_; + rtc::scoped_ptr<LatencyMeasuringAudioStream> latency_audio_stream_; +}; + +// AudioDeviceTest test fixture. +class AudioDeviceTest : public ::testing::Test { + protected: + AudioDeviceTest() + : test_is_done_(EventWrapper::Create()) { + // One-time initialization of JVM and application context. Ensures that we + // can do calls between C++ and Java. Initializes both Java and OpenSL ES + // implementations. + webrtc::audiodevicemodule::EnsureInitialized(); + // Creates an audio device using a default audio layer. + audio_device_ = CreateAudioDevice(AudioDeviceModule::kPlatformDefaultAudio); + EXPECT_NE(audio_device_.get(), nullptr); + EXPECT_EQ(0, audio_device_->Init()); + playout_parameters_ = audio_manager()->GetPlayoutAudioParameters(); + record_parameters_ = audio_manager()->GetRecordAudioParameters(); + build_info_.reset(new BuildInfo()); + } + virtual ~AudioDeviceTest() { + EXPECT_EQ(0, audio_device_->Terminate()); + } + + int playout_sample_rate() const { + return playout_parameters_.sample_rate(); + } + int record_sample_rate() const { + return record_parameters_.sample_rate(); + } + int playout_channels() const { + return playout_parameters_.channels(); + } + int record_channels() const { + return record_parameters_.channels(); + } + size_t playout_frames_per_10ms_buffer() const { + return playout_parameters_.frames_per_10ms_buffer(); + } + size_t record_frames_per_10ms_buffer() const { + return record_parameters_.frames_per_10ms_buffer(); + } + + int total_delay_ms() const { + return audio_manager()->GetDelayEstimateInMilliseconds(); + } + + rtc::scoped_refptr<AudioDeviceModule> audio_device() const { + return audio_device_; + } + + AudioDeviceModuleImpl* audio_device_impl() const { + return static_cast<AudioDeviceModuleImpl*>(audio_device_.get()); + } + + AudioManager* audio_manager() const { + return audio_device_impl()->GetAndroidAudioManagerForTest(); + } + + AudioManager* GetAudioManager(AudioDeviceModule* adm) const { + return static_cast<AudioDeviceModuleImpl*>(adm)-> + GetAndroidAudioManagerForTest(); + } + + AudioDeviceBuffer* audio_device_buffer() const { + return audio_device_impl()->GetAudioDeviceBuffer(); + } + + rtc::scoped_refptr<AudioDeviceModule> CreateAudioDevice( + AudioDeviceModule::AudioLayer audio_layer) { + rtc::scoped_refptr<AudioDeviceModule> module( + AudioDeviceModuleImpl::Create(0, audio_layer)); + return module; + } + + // Returns file name relative to the resource root given a sample rate. + std::string GetFileName(int sample_rate) { + EXPECT_TRUE(sample_rate == 48000 || sample_rate == 44100); + char fname[64]; + snprintf(fname, + sizeof(fname), + "audio_device/audio_short%d", + sample_rate / 1000); + std::string file_name(webrtc::test::ResourcePath(fname, "pcm")); + EXPECT_TRUE(test::FileExists(file_name)); +#ifdef ENABLE_PRINTF + PRINT("file name: %s\n", file_name.c_str()); + const size_t bytes = test::GetFileSize(file_name); + PRINT("file size: %" PRIuS " [bytes]\n", bytes); + PRINT("file size: %" PRIuS " [samples]\n", bytes / kBytesPerSample); + const int seconds = + static_cast<int>(bytes / (sample_rate * kBytesPerSample)); + PRINT("file size: %d [secs]\n", seconds); + PRINT("file size: %" PRIuS " [callbacks]\n", + seconds * kNumCallbacksPerSecond); +#endif + return file_name; + } + + AudioDeviceModule::AudioLayer GetActiveAudioLayer() const { + AudioDeviceModule::AudioLayer audio_layer; + EXPECT_EQ(0, audio_device()->ActiveAudioLayer(&audio_layer)); + return audio_layer; + } + + int TestDelayOnAudioLayer( + const AudioDeviceModule::AudioLayer& layer_to_test) { + rtc::scoped_refptr<AudioDeviceModule> audio_device; + audio_device = CreateAudioDevice(layer_to_test); + EXPECT_NE(audio_device.get(), nullptr); + AudioManager* audio_manager = GetAudioManager(audio_device.get()); + EXPECT_NE(audio_manager, nullptr); + return audio_manager->GetDelayEstimateInMilliseconds(); + } + + AudioDeviceModule::AudioLayer TestActiveAudioLayer( + const AudioDeviceModule::AudioLayer& layer_to_test) { + rtc::scoped_refptr<AudioDeviceModule> audio_device; + audio_device = CreateAudioDevice(layer_to_test); + EXPECT_NE(audio_device.get(), nullptr); + AudioDeviceModule::AudioLayer active; + EXPECT_EQ(0, audio_device->ActiveAudioLayer(&active)); + return active; + } + + bool DisableTestForThisDevice(const std::string& model) { + return (build_info_->GetDeviceModel() == model); + } + + // Volume control is currently only supported for the Java output audio layer. + // For OpenSL ES, the internal stream volume is always on max level and there + // is no need for this test to set it to max. + bool AudioLayerSupportsVolumeControl() const { + return GetActiveAudioLayer() == AudioDeviceModule::kAndroidJavaAudio; + } + + void SetMaxPlayoutVolume() { + if (!AudioLayerSupportsVolumeControl()) + return; + uint32_t max_volume; + EXPECT_EQ(0, audio_device()->MaxSpeakerVolume(&max_volume)); + EXPECT_EQ(0, audio_device()->SetSpeakerVolume(max_volume)); + } + + void DisableBuiltInAECIfAvailable() { + if (audio_device()->BuiltInAECIsAvailable()) { + EXPECT_EQ(0, audio_device()->EnableBuiltInAEC(false)); + } + } + + void StartPlayout() { + EXPECT_FALSE(audio_device()->PlayoutIsInitialized()); + EXPECT_FALSE(audio_device()->Playing()); + EXPECT_EQ(0, audio_device()->InitPlayout()); + EXPECT_TRUE(audio_device()->PlayoutIsInitialized()); + EXPECT_EQ(0, audio_device()->StartPlayout()); + EXPECT_TRUE(audio_device()->Playing()); + } + + void StopPlayout() { + EXPECT_EQ(0, audio_device()->StopPlayout()); + EXPECT_FALSE(audio_device()->Playing()); + EXPECT_FALSE(audio_device()->PlayoutIsInitialized()); + } + + void StartRecording() { + EXPECT_FALSE(audio_device()->RecordingIsInitialized()); + EXPECT_FALSE(audio_device()->Recording()); + EXPECT_EQ(0, audio_device()->InitRecording()); + EXPECT_TRUE(audio_device()->RecordingIsInitialized()); + EXPECT_EQ(0, audio_device()->StartRecording()); + EXPECT_TRUE(audio_device()->Recording()); + } + + void StopRecording() { + EXPECT_EQ(0, audio_device()->StopRecording()); + EXPECT_FALSE(audio_device()->Recording()); + } + + int GetMaxSpeakerVolume() const { + uint32_t max_volume(0); + EXPECT_EQ(0, audio_device()->MaxSpeakerVolume(&max_volume)); + return max_volume; + } + + int GetMinSpeakerVolume() const { + uint32_t min_volume(0); + EXPECT_EQ(0, audio_device()->MinSpeakerVolume(&min_volume)); + return min_volume; + } + + int GetSpeakerVolume() const { + uint32_t volume(0); + EXPECT_EQ(0, audio_device()->SpeakerVolume(&volume)); + return volume; + } + + rtc::scoped_ptr<EventWrapper> test_is_done_; + rtc::scoped_refptr<AudioDeviceModule> audio_device_; + AudioParameters playout_parameters_; + AudioParameters record_parameters_; + rtc::scoped_ptr<BuildInfo> build_info_; +}; + +TEST_F(AudioDeviceTest, ConstructDestruct) { + // Using the test fixture to create and destruct the audio device module. +} + +// We always ask for a default audio layer when the ADM is constructed. But the +// ADM will then internally set the best suitable combination of audio layers, +// for input and output based on if low-latency output audio in combination +// with OpenSL ES is supported or not. This test ensures that the correct +// selection is done. +TEST_F(AudioDeviceTest, VerifyDefaultAudioLayer) { + const AudioDeviceModule::AudioLayer audio_layer = GetActiveAudioLayer(); + bool low_latency_output = audio_manager()->IsLowLatencyPlayoutSupported(); + AudioDeviceModule::AudioLayer expected_audio_layer = low_latency_output ? + AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio : + AudioDeviceModule::kAndroidJavaAudio; + EXPECT_EQ(expected_audio_layer, audio_layer); +} + +// Verify that it is possible to explicitly create the two types of supported +// ADMs. These two tests overrides the default selection of native audio layer +// by ignoring if the device supports low-latency output or not. +TEST_F(AudioDeviceTest, CorrectAudioLayerIsUsedForCombinedJavaOpenSLCombo) { + AudioDeviceModule::AudioLayer expected_layer = + AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio; + AudioDeviceModule::AudioLayer active_layer = TestActiveAudioLayer( + expected_layer); + EXPECT_EQ(expected_layer, active_layer); +} + +TEST_F(AudioDeviceTest, CorrectAudioLayerIsUsedForJavaInBothDirections) { + AudioDeviceModule::AudioLayer expected_layer = + AudioDeviceModule::kAndroidJavaAudio; + AudioDeviceModule::AudioLayer active_layer = TestActiveAudioLayer( + expected_layer); + EXPECT_EQ(expected_layer, active_layer); +} + +// The Android ADM supports two different delay reporting modes. One for the +// low-latency output path (in combination with OpenSL ES), and one for the +// high-latency output path (Java backends in both directions). These two tests +// verifies that the audio manager reports correct delay estimate given the +// selected audio layer. Note that, this delay estimate will only be utilized +// if the HW AEC is disabled. +TEST_F(AudioDeviceTest, UsesCorrectDelayEstimateForHighLatencyOutputPath) { + EXPECT_EQ(kHighLatencyModeDelayEstimateInMilliseconds, + TestDelayOnAudioLayer(AudioDeviceModule::kAndroidJavaAudio)); +} + +TEST_F(AudioDeviceTest, UsesCorrectDelayEstimateForLowLatencyOutputPath) { + EXPECT_EQ(kLowLatencyModeDelayEstimateInMilliseconds, + TestDelayOnAudioLayer( + AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio)); +} + +// Ensure that the ADM internal audio device buffer is configured to use the +// correct set of parameters. +TEST_F(AudioDeviceTest, VerifyAudioDeviceBufferParameters) { + EXPECT_EQ(playout_parameters_.sample_rate(), + audio_device_buffer()->PlayoutSampleRate()); + EXPECT_EQ(record_parameters_.sample_rate(), + audio_device_buffer()->RecordingSampleRate()); + EXPECT_EQ(playout_parameters_.channels(), + audio_device_buffer()->PlayoutChannels()); + EXPECT_EQ(record_parameters_.channels(), + audio_device_buffer()->RecordingChannels()); +} + + +TEST_F(AudioDeviceTest, InitTerminate) { + // Initialization is part of the test fixture. + EXPECT_TRUE(audio_device()->Initialized()); + EXPECT_EQ(0, audio_device()->Terminate()); + EXPECT_FALSE(audio_device()->Initialized()); +} + +TEST_F(AudioDeviceTest, Devices) { + // Device enumeration is not supported. Verify fixed values only. + EXPECT_EQ(1, audio_device()->PlayoutDevices()); + EXPECT_EQ(1, audio_device()->RecordingDevices()); +} + +TEST_F(AudioDeviceTest, SpeakerVolumeShouldBeAvailable) { + // The OpenSL ES output audio path does not support volume control. + if (!AudioLayerSupportsVolumeControl()) + return; + bool available; + EXPECT_EQ(0, audio_device()->SpeakerVolumeIsAvailable(&available)); + EXPECT_TRUE(available); +} + +TEST_F(AudioDeviceTest, MaxSpeakerVolumeIsPositive) { + // The OpenSL ES output audio path does not support volume control. + if (!AudioLayerSupportsVolumeControl()) + return; + StartPlayout(); + EXPECT_GT(GetMaxSpeakerVolume(), 0); + StopPlayout(); +} + +TEST_F(AudioDeviceTest, MinSpeakerVolumeIsZero) { + // The OpenSL ES output audio path does not support volume control. + if (!AudioLayerSupportsVolumeControl()) + return; + EXPECT_EQ(GetMinSpeakerVolume(), 0); +} + +TEST_F(AudioDeviceTest, DefaultSpeakerVolumeIsWithinMinMax) { + // The OpenSL ES output audio path does not support volume control. + if (!AudioLayerSupportsVolumeControl()) + return; + const int default_volume = GetSpeakerVolume(); + EXPECT_GE(default_volume, GetMinSpeakerVolume()); + EXPECT_LE(default_volume, GetMaxSpeakerVolume()); +} + +TEST_F(AudioDeviceTest, SetSpeakerVolumeActuallySetsVolume) { + // The OpenSL ES output audio path does not support volume control. + if (!AudioLayerSupportsVolumeControl()) + return; + const int default_volume = GetSpeakerVolume(); + const int max_volume = GetMaxSpeakerVolume(); + EXPECT_EQ(0, audio_device()->SetSpeakerVolume(max_volume)); + int new_volume = GetSpeakerVolume(); + EXPECT_EQ(new_volume, max_volume); + EXPECT_EQ(0, audio_device()->SetSpeakerVolume(default_volume)); +} + +// Tests that playout can be initiated, started and stopped. No audio callback +// is registered in this test. +// Flaky on our trybots makes this test unusable. +// https://code.google.com/p/webrtc/issues/detail?id=5046 +TEST_F(AudioDeviceTest, DISABLED_StartStopPlayout) { + StartPlayout(); + StopPlayout(); + StartPlayout(); + StopPlayout(); +} + +// Tests that recording can be initiated, started and stopped. No audio callback +// is registered in this test. +TEST_F(AudioDeviceTest, StartStopRecording) { + StartRecording(); + StopRecording(); + StartRecording(); + StopRecording(); +} + +// Verify that calling StopPlayout() will leave us in an uninitialized state +// which will require a new call to InitPlayout(). This test does not call +// StartPlayout() while being uninitialized since doing so will hit a +// RTC_DCHECK. +TEST_F(AudioDeviceTest, StopPlayoutRequiresInitToRestart) { + EXPECT_EQ(0, audio_device()->InitPlayout()); + EXPECT_EQ(0, audio_device()->StartPlayout()); + EXPECT_EQ(0, audio_device()->StopPlayout()); + EXPECT_FALSE(audio_device()->PlayoutIsInitialized()); +} + +// Start playout and verify that the native audio layer starts asking for real +// audio samples to play out using the NeedMorePlayData callback. +TEST_F(AudioDeviceTest, StartPlayoutVerifyCallbacks) { + MockAudioTransport mock(kPlayout); + mock.HandleCallbacks(test_is_done_.get(), nullptr, kNumCallbacks); + EXPECT_CALL(mock, NeedMorePlayData(playout_frames_per_10ms_buffer(), + kBytesPerSample, + playout_channels(), + playout_sample_rate(), + NotNull(), + _, _, _)) + .Times(AtLeast(kNumCallbacks)); + EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock)); + StartPlayout(); + test_is_done_->Wait(kTestTimeOutInMilliseconds); + StopPlayout(); +} + +// Start recording and verify that the native audio layer starts feeding real +// audio samples via the RecordedDataIsAvailable callback. +TEST_F(AudioDeviceTest, StartRecordingVerifyCallbacks) { + MockAudioTransport mock(kRecording); + mock.HandleCallbacks(test_is_done_.get(), nullptr, kNumCallbacks); + EXPECT_CALL(mock, RecordedDataIsAvailable(NotNull(), + record_frames_per_10ms_buffer(), + kBytesPerSample, + record_channels(), + record_sample_rate(), + total_delay_ms(), + 0, + 0, + false, + _)) + .Times(AtLeast(kNumCallbacks)); + + EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock)); + StartRecording(); + test_is_done_->Wait(kTestTimeOutInMilliseconds); + StopRecording(); +} + + +// Start playout and recording (full-duplex audio) and verify that audio is +// active in both directions. +TEST_F(AudioDeviceTest, StartPlayoutAndRecordingVerifyCallbacks) { + MockAudioTransport mock(kPlayout | kRecording); + mock.HandleCallbacks(test_is_done_.get(), nullptr, kNumCallbacks); + EXPECT_CALL(mock, NeedMorePlayData(playout_frames_per_10ms_buffer(), + kBytesPerSample, + playout_channels(), + playout_sample_rate(), + NotNull(), + _, _, _)) + .Times(AtLeast(kNumCallbacks)); + EXPECT_CALL(mock, RecordedDataIsAvailable(NotNull(), + record_frames_per_10ms_buffer(), + kBytesPerSample, + record_channels(), + record_sample_rate(), + total_delay_ms(), + 0, + 0, + false, + _)) + .Times(AtLeast(kNumCallbacks)); + EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock)); + StartPlayout(); + StartRecording(); + test_is_done_->Wait(kTestTimeOutInMilliseconds); + StopRecording(); + StopPlayout(); +} + +// Start playout and read audio from an external PCM file when the audio layer +// asks for data to play out. Real audio is played out in this test but it does +// not contain any explicit verification that the audio quality is perfect. +TEST_F(AudioDeviceTest, RunPlayoutWithFileAsSource) { + // TODO(henrika): extend test when mono output is supported. + EXPECT_EQ(1, playout_channels()); + NiceMock<MockAudioTransport> mock(kPlayout); + const int num_callbacks = kFilePlayTimeInSec * kNumCallbacksPerSecond; + std::string file_name = GetFileName(playout_sample_rate()); + rtc::scoped_ptr<FileAudioStream> file_audio_stream( + new FileAudioStream(num_callbacks, file_name, playout_sample_rate())); + mock.HandleCallbacks(test_is_done_.get(), + file_audio_stream.get(), + num_callbacks); + // SetMaxPlayoutVolume(); + EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock)); + StartPlayout(); + test_is_done_->Wait(kTestTimeOutInMilliseconds); + StopPlayout(); +} + +// Start playout and recording and store recorded data in an intermediate FIFO +// buffer from which the playout side then reads its samples in the same order +// as they were stored. Under ideal circumstances, a callback sequence would +// look like: ...+-+-+-+-+-+-+-..., where '+' means 'packet recorded' and '-' +// means 'packet played'. Under such conditions, the FIFO would only contain +// one packet on average. However, under more realistic conditions, the size +// of the FIFO will vary more due to an unbalance between the two sides. +// This test tries to verify that the device maintains a balanced callback- +// sequence by running in loopback for ten seconds while measuring the size +// (max and average) of the FIFO. The size of the FIFO is increased by the +// recording side and decreased by the playout side. +// TODO(henrika): tune the final test parameters after running tests on several +// different devices. +TEST_F(AudioDeviceTest, RunPlayoutAndRecordingInFullDuplex) { + EXPECT_EQ(record_channels(), playout_channels()); + EXPECT_EQ(record_sample_rate(), playout_sample_rate()); + NiceMock<MockAudioTransport> mock(kPlayout | kRecording); + rtc::scoped_ptr<FifoAudioStream> fifo_audio_stream( + new FifoAudioStream(playout_frames_per_10ms_buffer())); + mock.HandleCallbacks(test_is_done_.get(), + fifo_audio_stream.get(), + kFullDuplexTimeInSec * kNumCallbacksPerSecond); + SetMaxPlayoutVolume(); + EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock)); + StartRecording(); + StartPlayout(); + test_is_done_->Wait(std::max(kTestTimeOutInMilliseconds, + 1000 * kFullDuplexTimeInSec)); + StopPlayout(); + StopRecording(); + EXPECT_LE(fifo_audio_stream->average_size(), 10u); + EXPECT_LE(fifo_audio_stream->largest_size(), 20u); +} + +// Measures loopback latency and reports the min, max and average values for +// a full duplex audio session. +// The latency is measured like so: +// - Insert impulses periodically on the output side. +// - Detect the impulses on the input side. +// - Measure the time difference between the transmit time and receive time. +// - Store time differences in a vector and calculate min, max and average. +// This test requires a special hardware called Audio Loopback Dongle. +// See http://source.android.com/devices/audio/loopback.html for details. +TEST_F(AudioDeviceTest, DISABLED_MeasureLoopbackLatency) { + EXPECT_EQ(record_channels(), playout_channels()); + EXPECT_EQ(record_sample_rate(), playout_sample_rate()); + NiceMock<MockAudioTransport> mock(kPlayout | kRecording); + rtc::scoped_ptr<LatencyMeasuringAudioStream> latency_audio_stream( + new LatencyMeasuringAudioStream(playout_frames_per_10ms_buffer())); + mock.HandleCallbacks(test_is_done_.get(), + latency_audio_stream.get(), + kMeasureLatencyTimeInSec * kNumCallbacksPerSecond); + EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock)); + SetMaxPlayoutVolume(); + DisableBuiltInAECIfAvailable(); + StartRecording(); + StartPlayout(); + test_is_done_->Wait(std::max(kTestTimeOutInMilliseconds, + 1000 * kMeasureLatencyTimeInSec)); + StopPlayout(); + StopRecording(); + // Verify that the correct number of transmitted impulses are detected. + EXPECT_EQ(latency_audio_stream->num_latency_values(), + static_cast<size_t>( + kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1)); + latency_audio_stream->PrintResults(); +} + +} // namespace webrtc |