blob: b0a356f2e64198cb952ece98bc4842f4e51f2003 [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 <algorithm>
#include <limits>
#include <list>
#include <memory>
#include <numeric>
#include <string>
#include <vector>
#include "modules/audio_device/audio_device_impl.h"
#include "modules/audio_device/include/audio_device.h"
#include "modules/audio_device/include/mock_audio_transport.h"
#include "modules/audio_device/ios/audio_device_ios.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/criticalsection.h"
#include "rtc_base/format_macros.h"
#include "rtc_base/logging.h"
#include "rtc_base/scoped_ref_ptr.h"
#include "rtc_base/timeutils.h"
#include "system_wrappers/include/event_wrapper.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/testsupport/fileutils.h"
#import "sdk/objc/Framework/Classes/Audio/RTCAudioSession+Private.h"
#import "sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.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;
// #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;
// Number of bits per PCM audio sample.
static const size_t kBitsPerSample = 16;
// Number of bytes per PCM audio sample.
static const size_t kBytesPerSample = kBitsPerSample / 8;
// 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 = 15;
// Run the full-duplex test during this time (unit is in seconds).
// Note that first |kNumIgnoreFirstCallbacks| are ignored.
static const int kFullDuplexTimeInSec = 10;
// 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.
// TODO(henrika): fine tune this setting for iOS.
static const int kImpulseFrequencyInHz = 1;
// Length of round-trip latency measurements. Number of transmitted impulses
// is kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1.
// TODO(henrika): fine tune this setting for iOS.
static const int kMeasureLatencyTimeInSec = 5;
// Utilized in round-trip latency measurements to avoid capturing noise samples.
// TODO(henrika): fine tune this setting for iOS.
static const int kImpulseThreshold = 50;
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_;
std::unique_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_;
std::unique_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)
: 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_ = rtc::TimeMillis();
}
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 = rtc::TimeMillis();
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 10.0 * (index / frames_per_buffer_) + 0.5;
}
private:
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 MockAudioTransportIOS : public test::MockAudioTransport {
public:
explicit MockAudioTransportIOS(int type)
: num_callbacks_(0),
type_(type),
play_count_(0),
rec_count_(0),
audio_stream_(nullptr) {}
virtual ~MockAudioTransportIOS() {}
// 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,
size_t 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, &MockAudioTransportIOS::RealNeedMorePlayData));
}
if (rec_mode()) {
ON_CALL(*this, RecordedDataIsAvailable(_, _, _, _, _, _, _, _, _, _))
.WillByDefault(Invoke(
this, &MockAudioTransportIOS::RealRecordedDataIsAvailable));
}
}
int32_t RealRecordedDataIsAvailable(const void* audioSamples,
const size_t nSamples,
const size_t nBytesPerSample,
const size_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()) {
if (test_is_done_) {
test_is_done_->Set();
}
}
return 0;
}
int32_t RealNeedMorePlayData(const size_t nSamples,
const size_t nBytesPerSample,
const size_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);
} else {
memset(audioSamples, 0, nSamples * nBytesPerSample);
}
if (ReceivedEnoughCallbacks()) {
if (test_is_done_) {
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_;
};
// AudioDeviceTest test fixture.
class AudioDeviceTest : public ::testing::Test {
protected:
AudioDeviceTest() : test_is_done_(EventWrapper::Create()) {
old_sev_ = rtc::LogMessage::GetLogToDebug();
// Set suitable logging level here. Change to rtc::LS_INFO for more verbose
// output. See webrtc/rtc_base/logging.h for complete list of options.
rtc::LogMessage::LogToDebug(rtc::LS_INFO);
// Add extra logging fields here (timestamps and thread id).
// rtc::LogMessage::LogTimestamps();
rtc::LogMessage::LogThreads();
// 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());
EXPECT_EQ(0,
audio_device()->GetPlayoutAudioParameters(&playout_parameters_));
EXPECT_EQ(0, audio_device()->GetRecordAudioParameters(&record_parameters_));
}
virtual ~AudioDeviceTest() {
EXPECT_EQ(0, audio_device_->Terminate());
rtc::LogMessage::LogToDebug(old_sev_);
}
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();
}
rtc::scoped_refptr<AudioDeviceModule> audio_device() const {
return audio_device_;
}
AudioDeviceModuleImpl* audio_device_impl() const {
return static_cast<AudioDeviceModuleImpl*>(audio_device_.get());
}
AudioDeviceBuffer* audio_device_buffer() const {
return audio_device_impl()->GetAudioDeviceBuffer();
}
rtc::scoped_refptr<AudioDeviceModule> CreateAudioDevice(
AudioDeviceModule::AudioLayer audio_layer) {
rtc::scoped_refptr<AudioDeviceModule> module(
AudioDeviceModule::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 ||
sample_rate == 16000);
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_DEBUG_PRINTF
PRINTD("file name: %s\n", file_name.c_str());
const size_t bytes = test::GetFileSize(file_name);
PRINTD("file size: %" PRIuS " [bytes]\n", bytes);
PRINTD("file size: %" PRIuS " [samples]\n", bytes / kBytesPerSample);
const int seconds =
static_cast<int>(bytes / (sample_rate * kBytesPerSample));
PRINTD("file size: %d [secs]\n", seconds);
PRINTD("file size: %" PRIuS " [callbacks]\n",
seconds * kNumCallbacksPerSecond);
#endif
return file_name;
}
void StartPlayout() {
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());
}
void StartRecording() {
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());
}
std::unique_ptr<EventWrapper> test_is_done_;
rtc::scoped_refptr<AudioDeviceModule> audio_device_;
AudioParameters playout_parameters_;
AudioParameters record_parameters_;
rtc::LoggingSeverity old_sev_;
};
TEST_F(AudioDeviceTest, ConstructDestruct) {
// Using the test fixture to create and destruct the audio device module.
}
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());
}
// Tests that playout can be initiated, started and stopped. No audio callback
// is registered in this test.
// Failing when running on real iOS devices: bugs.webrtc.org/6889.
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.
// Can sometimes fail when running on real devices: bugs.webrtc.org/7888.
TEST_F(AudioDeviceTest, DISABLED_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());
}
// Verify that we can create two ADMs and start playing on the second ADM.
// Only the first active instance shall activate an audio session and the
// last active instance shall deactivate the audio session. The test does not
// explicitly verify correct audio session calls but instead focuses on
// ensuring that audio starts for both ADMs.
// Failing when running on real iOS devices: bugs.webrtc.org/6889.
TEST_F(AudioDeviceTest, DISABLED_StartPlayoutOnTwoInstances) {
// Create and initialize a second/extra ADM instance. The default ADM is
// created by the test harness.
rtc::scoped_refptr<AudioDeviceModule> second_audio_device =
CreateAudioDevice(AudioDeviceModule::kPlatformDefaultAudio);
EXPECT_NE(second_audio_device.get(), nullptr);
EXPECT_EQ(0, second_audio_device->Init());
// Start playout for the default ADM but don't wait here. Instead use the
// upcoming second stream for that. We set the same expectation on number
// of callbacks as for the second stream.
NiceMock<MockAudioTransportIOS> mock(kPlayout);
mock.HandleCallbacks(nullptr, nullptr, 0);
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();
// Initialize playout for the second ADM. If all is OK, the second ADM shall
// reuse the audio session activated when the first ADM started playing.
// This call will also ensure that we avoid a problem related to initializing
// two different audio unit instances back to back (see webrtc:5166 for
// details).
EXPECT_EQ(0, second_audio_device->InitPlayout());
EXPECT_TRUE(second_audio_device->PlayoutIsInitialized());
// Start playout for the second ADM and verify that it starts as intended.
// Passing this test ensures that initialization of the second audio unit
// has been done successfully and that there is no conflict with the already
// playing first ADM.
MockAudioTransportIOS mock2(kPlayout);
mock2.HandleCallbacks(test_is_done_.get(), nullptr, kNumCallbacks);
EXPECT_CALL(
mock2, NeedMorePlayData(playout_frames_per_10ms_buffer(), kBytesPerSample,
playout_channels(), playout_sample_rate(),
NotNull(), _, _, _))
.Times(AtLeast(kNumCallbacks));
EXPECT_EQ(0, second_audio_device->RegisterAudioCallback(&mock2));
EXPECT_EQ(0, second_audio_device->StartPlayout());
EXPECT_TRUE(second_audio_device->Playing());
test_is_done_->Wait(kTestTimeOutInMilliseconds);
EXPECT_EQ(0, second_audio_device->StopPlayout());
EXPECT_FALSE(second_audio_device->Playing());
EXPECT_FALSE(second_audio_device->PlayoutIsInitialized());
EXPECT_EQ(0, second_audio_device->Terminate());
}
// 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) {
MockAudioTransportIOS 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) {
MockAudioTransportIOS 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(),
_, // TODO(henrika): fix delay
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) {
MockAudioTransportIOS 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(),
_, // TODO(henrika): fix delay
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<MockAudioTransportIOS> mock(kPlayout);
const int num_callbacks = kFilePlayTimeInSec * kNumCallbacksPerSecond;
std::string file_name = GetFileName(playout_sample_rate());
std::unique_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();
}
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());
}
// 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<MockAudioTransportIOS> mock(kPlayout | kRecording);
std::unique_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<MockAudioTransportIOS> mock(kPlayout | kRecording);
std::unique_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();
}
// Verifies that the AudioDeviceIOS is_interrupted_ flag is reset correctly
// after an iOS AVAudioSessionInterruptionTypeEnded notification event.
// AudioDeviceIOS listens to RTCAudioSession interrupted notifications by:
// - In AudioDeviceIOS.InitPlayOrRecord registers its audio_session_observer_
// callback with RTCAudioSession's delegate list.
// - When RTCAudioSession receives an iOS audio interrupted notification, it
// passes the notification to callbacks in its delegate list which sets
// AudioDeviceIOS's is_interrupted_ flag to true.
// - When AudioDeviceIOS.ShutdownPlayOrRecord is called, its
// audio_session_observer_ callback is removed from RTCAudioSessions's
// delegate list.
// So if RTCAudioSession receives an iOS end audio interruption notification,
// AudioDeviceIOS is not notified as its callback is not in RTCAudioSession's
// delegate list. This causes AudioDeviceIOS's is_interrupted_ flag to be in
// the wrong (true) state and the audio session will ignore audio changes.
// As RTCAudioSession keeps its own interrupted state, the fix is to initialize
// AudioDeviceIOS's is_interrupted_ flag to RTCAudioSession's isInterrupted
// flag in AudioDeviceIOS.InitPlayOrRecord.
TEST_F(AudioDeviceTest, testInterruptedAudioSession) {
RTCAudioSession *session = [RTCAudioSession sharedInstance];
std::unique_ptr<webrtc::AudioDeviceIOS> audio_device;
audio_device.reset(new webrtc::AudioDeviceIOS());
std::unique_ptr<webrtc::AudioDeviceBuffer> audio_buffer;
audio_buffer.reset(new webrtc::AudioDeviceBuffer());
audio_device->AttachAudioBuffer(audio_buffer.get());
audio_device->Init();
audio_device->InitPlayout();
// Force interruption.
[session notifyDidBeginInterruption];
// Wait for notification to propagate.
rtc::MessageQueueManager::ProcessAllMessageQueues();
EXPECT_TRUE(audio_device->is_interrupted_);
// Force it for testing.
audio_device->playing_ = false;
audio_device->ShutdownPlayOrRecord();
// Force it for testing.
audio_device->audio_is_initialized_ = false;
[session notifyDidEndInterruptionWithShouldResumeSession:YES];
// Wait for notification to propagate.
rtc::MessageQueueManager::ProcessAllMessageQueues();
EXPECT_TRUE(audio_device->is_interrupted_);
audio_device->Init();
audio_device->InitPlayout();
EXPECT_FALSE(audio_device->is_interrupted_);
}
} // namespace webrtc