Adds unit test for ADM on Linux

webrtc/modules/audio_device/audio_device_unittest.cc

+/*
+ *  Copyright (c) 2017 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 <cstring>
+
+#include "webrtc/base/event.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/modules/audio_device/audio_device_impl.h"
+#include "webrtc/modules/audio_device/include/audio_device.h"
+#include "webrtc/modules/audio_device/include/mock_audio_transport.h"
+#include "webrtc/system_wrappers/include/sleep.h"
+#include "webrtc/test/gmock.h"
+#include "webrtc/test/gtest.h"
+
+using ::testing::_;
+using ::testing::AtLeast;
+using ::testing::Ge;
+using ::testing::Invoke;
+using ::testing::NiceMock;
+using ::testing::NotNull;
+
+namespace webrtc {
+namespace {
+
+// Don't run these tests in combination with sanitizers.
+#if !defined(ADDRESS_SANITIZER) && !defined(MEMORY_SANITIZER)
+#define SKIP_TEST_IF_NOT(requirements_satisfied) \
+  do {                                           \
+    if (!requirements_satisfied) {               \
+      return;                                    \
+    }                                            \
+  } while (false)
+#else
+// Or if other audio-related requirements are not met.
+#define SKIP_TEST_IF_NOT(requirements_satisfied) \
+  do {                                           \
+    return;                                      \
+  } while (false)
+#endif
+
+// 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;
+
+enum class TransportType {
+  kInvalid,
+  kPlay,
+  kRecord,
+  kPlayAndRecord,
+};
+}  // namespace
+
+// Mocks the AudioTransport object and proxies actions for the two callbacks
+// (RecordedDataIsAvailable and NeedMorePlayData) to different implementations
+// of AudioStreamInterface.
+class MockAudioTransport : public test::MockAudioTransport {
+ public:
+  explicit MockAudioTransport(TransportType type) : type_(type) {}
+  ~MockAudioTransport() {}
+
+  // Set default actions of the mock object. We are delegating to fake
+  // implementation where the number of callbacks is counted and an event
+  // is set after a certain number of callbacks. Audio parameters are also
+  // checked.
+  void HandleCallbacks(rtc::Event* event, int num_callbacks) {
+    event_ = event;
+    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* audio_buffer,
+                                      const size_t samples_per_channel,
+                                      const size_t bytes_per_frame,
+                                      const size_t channels,
+                                      const uint32_t sample_rate,
+                                      const uint32_t total_delay_ms,
+                                      const int32_t clock_drift,
+                                      const uint32_t current_mic_level,
+                                      const bool typing_status,
+                                      uint32_t& new_mic_level) {
+    EXPECT_TRUE(rec_mode()) << "No test is expecting these callbacks.";
+    LOG(INFO) << "+";
+    // Store audio parameters once in the first callback. For all other
+    // callbacks, verify that the provided audio parameters are maintained and
+    // that each callback corresponds to 10ms for any given sample rate.
+    if (!record_parameters_.is_complete()) {
+      record_parameters_.reset(sample_rate, channels, samples_per_channel);
+    } else {
+      EXPECT_EQ(samples_per_channel, record_parameters_.frames_per_buffer());
+      EXPECT_EQ(bytes_per_frame, record_parameters_.GetBytesPerFrame());
+      EXPECT_EQ(channels, record_parameters_.channels());
+      EXPECT_EQ(static_cast<int>(sample_rate),
+                record_parameters_.sample_rate());
+      EXPECT_EQ(samples_per_channel,
+                record_parameters_.frames_per_10ms_buffer());
+    }
+    rec_count_++;
+    // Signal the event after given amount of callbacks.
+    if (ReceivedEnoughCallbacks()) {
+      event_->Set();
+    }
+    return 0;
+  }
+
+  int32_t RealNeedMorePlayData(const size_t samples_per_channel,
+                               const size_t bytes_per_frame,
+                               const size_t channels,
+                               const uint32_t sample_rate,
+                               void* audio_buffer,
+                               size_t& samples_per_channel_out,
+                               int64_t* elapsed_time_ms,
+                               int64_t* ntp_time_ms) {
+    EXPECT_TRUE(play_mode()) << "No test is expecting these callbacks.";
+    LOG(INFO) << "-";
+    // Store audio parameters once in the first callback. For all other
+    // callbacks, verify that the provided audio parameters are maintained and
+    // that each callback corresponds to 10ms for any given sample rate.
+    if (!playout_parameters_.is_complete()) {
+      playout_parameters_.reset(sample_rate, channels, samples_per_channel);
+    } else {
+      EXPECT_EQ(samples_per_channel, playout_parameters_.frames_per_buffer());
+      EXPECT_EQ(bytes_per_frame, playout_parameters_.GetBytesPerFrame());
+      EXPECT_EQ(channels, playout_parameters_.channels());
+      EXPECT_EQ(static_cast<int>(sample_rate),
+                playout_parameters_.sample_rate());
+      EXPECT_EQ(samples_per_channel,
+                playout_parameters_.frames_per_10ms_buffer());
+    }
+    play_count_++;
+    samples_per_channel_out = samples_per_channel;
+    // Fill the audio buffer with zeros to avoid disturbing audio.
+    const size_t num_bytes = samples_per_channel * bytes_per_frame;
+    std::memset(audio_buffer, 0, num_bytes);
+    // Signal the event after given amount of callbacks.
+    if (ReceivedEnoughCallbacks()) {
+      event_->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_ == TransportType::kPlay ||
+           type_ == TransportType::kPlayAndRecord;
+  }
+
+  bool rec_mode() const {
+    return type_ == TransportType::kRecord ||
+           type_ == TransportType::kPlayAndRecord;
+  }
+
+ private:
+  TransportType type_;

[the sun, 2017/03/16 15:24:14]

= kInvalid

(or remove kInvalid from the enum)

[henrika_webrtc, 2017/03/17 09:54:03]

Done.

+  rtc::Event* event_ = nullptr;
+  size_t num_callbacks_ = 0;
+  size_t play_count_ = 0;
+  size_t rec_count_ = 0;
+  AudioParameters playout_parameters_;
+  AudioParameters record_parameters_;
+};
+
+// AudioDeviceTest test fixture.
+class AudioDeviceTest : public ::testing::Test {

[the sun, 2017/03/16 15:24:14]

I had one more comment, which appears to have been lost.

I'd usually prefer to not use a fixture, but instead a test helper that you put
on the stack at the beginning of each test case (see e.g.
audio_send_stream_unittest.cc). That makes the code easier to follow, and more
flexible, e.g. if a certain test case would require an argument to the ctor, it
is a simple matter to add one. With a gtest fixture, you would end up with
either a whole separate base class, a "reconfig" method, or (worst case)
implementation inheritance in multiple steps.

I'm fine with this as is - just wanted to mention an alternative, because doing
test fixtures this way has spread across the code base as a bit of an
anti-pattern IMO.

[henrika_webrtc, 2017/03/17 09:54:03]

Thanks. I'll consider that next time. I have written some tests before and using
fixtures has (in Chrome) always been the recommended method. Do see your points
though.

+ protected:
+  AudioDeviceTest() : event_(false, false) {
+#if !defined(ADDRESS_SANITIZER) && !defined(MEMORY_SANITIZER)
+    rtc::LogMessage::LogToDebug(rtc::LS_INFO);
+    // Add extra logging fields here if needed for debugging.
+    // rtc::LogMessage::LogTimestamps();
+    // rtc::LogMessage::LogThreads();
+    audio_device_ =
+        AudioDeviceModule::Create(0, AudioDeviceModule::kPlatformDefaultAudio);
+    EXPECT_NE(audio_device_.get(), nullptr);
+    AudioDeviceModule::AudioLayer audio_layer;
+    EXPECT_EQ(0, audio_device_->ActiveAudioLayer(&audio_layer));
+    if (audio_layer == AudioDeviceModule::kLinuxAlsaAudio) {
+      requirements_satisfied_ = false;
+    }
+    if (requirements_satisfied_) {
+      EXPECT_EQ(0, audio_device_->Init());
+      const int16_t num_playout_devices = audio_device_->PlayoutDevices();
+      const int16_t num_record_devices = audio_device_->RecordingDevices();
+      requirements_satisfied_ =
+          num_playout_devices > 0 && num_record_devices > 0;
+    }
+#else
+    requirements_satisfied_ = false;
+#endif
+  }
+  virtual ~AudioDeviceTest() {}
+
+  // Combine all required initialization and do all of it in the test setup.
+  // This test suite does not focus on testing all components of the ADM but
+  // instead focus on the "pig picture" and how it is used in VoiceEngine.
+  // TODO(henrika): perhaps use SetUpTestCase() to share initial setup between

[the sun, 2017/03/16 15:24:14]

We're still sharding test cases in separate processes, so I don't think that
would do a lot for speed.

In fact, couldn't you fold these into the ctor/dtor to make the code a little
simpler?

[henrika_webrtc, 2017/03/17 09:54:02]

I removed the use of SetUp/TearDown. 

+  // all tests.
+  virtual void SetUp() {
+    if (!requirements_satisfied_)
+      return;
+    EXPECT_EQ(0, audio_device_->SetPlayoutDevice(0));
+    EXPECT_EQ(0, audio_device_->InitSpeaker());
+    EXPECT_EQ(0, audio_device_->SetRecordingDevice(0));
+    EXPECT_EQ(0, audio_device_->InitMicrophone());
+    EXPECT_EQ(0, audio_device_->StereoPlayoutIsAvailable(&stereo_playout_));
+    EXPECT_EQ(0, audio_device_->SetStereoPlayout(stereo_playout_));
+    EXPECT_EQ(0, audio_device_->StereoRecordingIsAvailable(&stereo_recording_));
+    EXPECT_EQ(0, audio_device_->SetStereoRecording(stereo_recording_));
+    EXPECT_EQ(0, audio_device_->SetAGC(false));
+    EXPECT_FALSE(audio_device_->AGC());
+  }
+
+  virtual void TearDown() {
+    if (audio_device_) {
+      EXPECT_EQ(0, audio_device_->Terminate());
+    }
+  }
+
+  bool requirements_satisfied() const { return requirements_satisfied_; }
+
+  const rtc::scoped_refptr<AudioDeviceModule>& audio_device() const {
+    return audio_device_;
+  }
+
+  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());
+    EXPECT_FALSE(audio_device()->PlayoutIsInitialized());
+  }
+
+  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());
+    EXPECT_FALSE(audio_device()->RecordingIsInitialized());
+  }
+
+  bool requirements_satisfied_ = true;

[the sun, 2017/03/16 15:24:14]

Make these private: and add an accessor for event_. 

[henrika_webrtc, 2017/03/17 09:54:02]

Used raw pointer as return value.

+  rtc::Event event_;
+  rtc::scoped_refptr<AudioDeviceModule> audio_device_;
+  bool stereo_playout_ = false;
+  bool stereo_recording_ = false;
+};
+
+// Uses the test fixture to create, initialize and destruct the ADM.
+TEST_F(AudioDeviceTest, ConstructDestruct) {}
+
+TEST_F(AudioDeviceTest, InitTerminate) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  // Initialization is part of the test fixture.
+  EXPECT_TRUE(audio_device()->Initialized());
+  EXPECT_EQ(0, audio_device()->Terminate());
+  EXPECT_FALSE(audio_device()->Initialized());
+}
+
+// Tests Start/Stop playout without any registered audio callback.
+TEST_F(AudioDeviceTest, StartStopPlayout) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  StartPlayout();
+  StopPlayout();
+  StartPlayout();
+  StopPlayout();
+}
+
+// Tests Start/Stop recording without any registered audio callback.
+TEST_F(AudioDeviceTest, StartStopRecording) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  StartRecording();
+  StopRecording();
+  StartRecording();
+  StopRecording();
+}
+
+// Start playout and verify that the native audio layer starts asking for real
+// audio samples to play out using the NeedMorePlayData() callback.
+// Note that we can't add expectations on audio parameters in EXPECT_CALL
+// since parameter are not provided in the each callback. We therefore test and
+// verify the parameters in the fake audio transport implementation instead.
+TEST_F(AudioDeviceTest, StartPlayoutVerifyCallbacks) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  MockAudioTransport mock(TransportType::kPlay);
+  mock.HandleCallbacks(&event_, kNumCallbacks);
+  EXPECT_CALL(mock, NeedMorePlayData(_, _, _, _, NotNull(), _, _, _))
+      .Times(AtLeast(kNumCallbacks));
+  EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
+  StartPlayout();
+  event_.Wait(kTestTimeOutInMilliseconds);
+  StopPlayout();
+}
+
+// Start recording and verify that the native audio layer starts providing real
+// audio samples using the RecordedDataIsAvailable() callback.
+TEST_F(AudioDeviceTest, StartRecordingVerifyCallbacks) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  MockAudioTransport mock(TransportType::kRecord);
+  mock.HandleCallbacks(&event_, kNumCallbacks);
+  EXPECT_CALL(mock, RecordedDataIsAvailable(NotNull(), _, _, _, _, Ge(0u), 0, _,
+                                            false, _))
+      .Times(AtLeast(kNumCallbacks));
+  EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
+  StartRecording();
+  event_.Wait(kTestTimeOutInMilliseconds);
+  StopRecording();
+}
+
+// Start playout and recording (full-duplex audio) and verify that audio is
+// active in both directions.
+TEST_F(AudioDeviceTest, StartPlayoutAndRecordingVerifyCallbacks) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  MockAudioTransport mock(TransportType::kPlayAndRecord);
+  mock.HandleCallbacks(&event_, kNumCallbacks);
+  EXPECT_CALL(mock, NeedMorePlayData(_, _, _, _, NotNull(), _, _, _))
+      .Times(AtLeast(kNumCallbacks));
+  EXPECT_CALL(mock, RecordedDataIsAvailable(NotNull(), _, _, _, _, Ge(0u), 0, _,
+                                            false, _))
+      .Times(AtLeast(kNumCallbacks));
+  EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
+  StartPlayout();
+  StartRecording();
+  event_.Wait(kTestTimeOutInMilliseconds);
+  StopRecording();
+  StopPlayout();
+}
+
+}  // namespace webrtc