Cleanup of iOS AudioDevice implementation

webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc

 /*
  *  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/logging.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/modules/audio_device/ios/audio_device_ios.h"
 #include "webrtc/system_wrappers/interface/clock.h"
 #include "webrtc/system_wrappers/interface/event_wrapper.h"
 #include "webrtc/system_wrappers/interface/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 int 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 int kBitsPerSample = 16;
+// Number of bytes per PCM audio sample.
+static const int kBytesPerSample = kBitsPerSample / 8;
 // Average number of audio callbacks per second assuming 10ms packet size.
 static const int kNumCallbacksPerSecond = 100;
 // Play out a test file during this time (unit is in seconds).
-static const int kFilePlayTimeInSec = 5;
-static const int kBitsPerSample = 16;
-static const int kBytesPerSample = kBitsPerSample / 8;
+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 = 5;
+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 int 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.
-static const int kMeasureLatencyTimeInSec = 11;
+// TODO(henrika): fine tune this setting for iOS.
+static const int kMeasureLatencyTimeInSec = 5;
 // Utilized in round-trip latency measurements to avoid capturing noise samples.
-static const int kImpulseThreshold = 1000;
+// 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, int num_frames) = 0;
   virtual void Read(void* destination, int 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(
-      int num_callbacks, const std::string& file_name, int sample_rate)
-      : file_size_in_bytes_(0),
-        sample_rate_(sample_rate),
-        file_pos_(0) {
+  FileAudioStream(int 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 int 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);
-    int num_samples_read = fread(
-        file_.get(), sizeof(int16_t), num_16bit_samples, audio_file);
+    int 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, int 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, int num_frames) override {
-    memcpy(destination,
-           static_cast<int16_t*> (&file_[file_pos_]),
+    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 (file_size_in_bytes_ / (kBytesPerSample * sample_rate_));
   }
   int file_size_in_callbacks() const {
     return file_size_in_seconds() * kNumCallbacksPerSecond;
   }
@@ skipping 18 matching lines @@
   explicit FifoAudioStream(int 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();
-  }
+  ~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, int 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_);
+    memcpy(static_cast<int16_t*>(&memory[0]), source, bytes_per_buffer_);
     rtc::CritScope lock(&lock_);
     fifo_->push_back(memory);
     const int size = fifo_->size();
     if (size > largest_size_) {
       largest_size_ = size;
       PRINTD("(%d)", 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, int 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_);
+      memcpy(destination, static_cast<int16_t*>(&memory[0]), bytes_per_buffer_);
       delete memory;
     }
   }
 
-  int size() const {
-    return fifo_->size();
-  }
+  int size() const { return fifo_->size(); }
 
-  int largest_size() const {
-    return largest_size_;
-  }
+  int largest_size() const { return largest_size_; }
 
   int average_size() const {
-    return (total_written_elements_ == 0) ? 0.0 : 0.5 + static_cast<float> (
-      total_written_elements_) / (write_count_ - kNumIgnoreFirstCallbacks);
+    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();
   }
 
@@ skipping 12 matching lines @@
 // 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(int 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) {
-  }
+        pulse_time_(0) {}
 
   // Insert periodic impulses in first two samples of |destination|.
   void Read(void* destination, int 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);
+      int16_t* ptr16 = static_cast<int16_t*>(destination);
       for (int i = 0; i < 2; ++i) {
         *ptr16++ = 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, int 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);
+    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));
+    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_));
+      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("-");
     }
   }
 
-  int num_latency_values() const {
-    return latencies_.size();
-  }
+  int 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();
+    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());
+    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:
   Clock* clock_;
   const int frames_per_buffer_;
   const int bytes_per_buffer_;
   int play_count_;
   int 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),
@@ skipping 106 matching lines @@
   bool play_mode() const { return type_ & kPlayout; }
   bool rec_mode() const { return type_ & kRecording; }
 
  private:
   EventWrapper* test_is_done_;
   int num_callbacks_;
   int type_;
   int play_count_;
   int 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();
+  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/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());
-    playout_parameters_ = audio_manager()->GetPlayoutAudioParameters();
-    record_parameters_ = audio_manager()->GetRecordAudioParameters();
-    build_info_.reset(new BuildInfo());
+    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();
-  }
+  // TODO(henrika): don't use hardcoded values below.
+  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(); }
   int playout_frames_per_10ms_buffer() const {
     return playout_parameters_.frames_per_10ms_buffer();
   }
   int record_frames_per_10ms_buffer() const {
     return record_parameters_.frames_per_10ms_buffer();
   }
 
   int total_delay_ms() const {
-    return audio_manager()->GetDelayEstimateInMilliseconds();
+    // TODO(henrika): improve this part.
+    return 100;
   }
 
   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);
+    EXPECT_TRUE(sample_rate == 48000 || sample_rate == 44100 ||
+                sample_rate == 16000);
     char fname[64];
-    snprintf(fname,
-             sizeof(fname),
-             "audio_device/audio_short%d",
+    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());
+#ifdef ENABLE_DEBUG_PRINTF
+    PRINTD("file name: %s\n", file_name.c_str());
     const int bytes = test::GetFileSize(file_name);
-    PRINT("file size: %d [bytes]\n", bytes);
-    PRINT("file size: %d [samples]\n", bytes / kBytesPerSample);
+    PRINTD("file size: %d [bytes]\n", bytes);
+    PRINTD("file size: %d [samples]\n", bytes / kBytesPerSample);
     const int seconds = bytes / (sample_rate * kBytesPerSample);
-    PRINT("file size: %d [secs]\n", seconds);
-    PRINT("file size: %d [callbacks]\n", seconds * kNumCallbacksPerSecond);
+    PRINTD("file size: %d [secs]\n", seconds);
+    PRINTD("file size: %d [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_;
+  rtc::LoggingSeverity old_sev_;
 };
 
 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());
+  // webrtc::SleepMs(5 * 1000);
   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.
 TEST_F(AudioDeviceTest, 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 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(),
-                                     _, _, _))
+                                     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_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) {
   MockAudioTransport mock(kPlayout | kRecording);
-  mock.HandleCallbacks(test_is_done_.get(), nullptr,  kNumCallbacks);
+  mock.HandleCallbacks(test_is_done_.get(), nullptr, kNumCallbacks);
   EXPECT_CALL(mock, NeedMorePlayData(playout_frames_per_10ms_buffer(),
-                                     kBytesPerSample,
-                                     playout_channels(),
-                                     playout_sample_rate(),
-                                     NotNull(),
-                                     _, _, _))
+                                     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_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<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(),
+  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<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(),
+  mock.HandleCallbacks(test_is_done_.get(), fifo_audio_stream.get(),
                        kFullDuplexTimeInSec * kNumCallbacksPerSecond);
-  SetMaxPlayoutVolume();
+  // SetMaxPlayoutVolume();
   EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
   StartRecording();
   StartPlayout();
-  test_is_done_->Wait(std::max(kTestTimeOutInMilliseconds,
-                               1000 * kFullDuplexTimeInSec));
+  test_is_done_->Wait(
+      std::max(kTestTimeOutInMilliseconds, 1000 * kFullDuplexTimeInSec));
   StopPlayout();
   StopRecording();
   EXPECT_LE(fifo_audio_stream->average_size(), 10);
   EXPECT_LE(fifo_audio_stream->largest_size(), 20);
 }
 
 // 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(),
+  mock.HandleCallbacks(test_is_done_.get(), latency_audio_stream.get(),
                        kMeasureLatencyTimeInSec * kNumCallbacksPerSecond);
   EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
-  SetMaxPlayoutVolume();
-  DisableBuiltInAECIfAvailable();
+  // SetMaxPlayoutVolume();
+  // DisableBuiltInAECIfAvailable();
   StartRecording();
   StartPlayout();
-  test_is_done_->Wait(std::max(kTestTimeOutInMilliseconds,
-                               1000 * kMeasureLatencyTimeInSec));
+  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(),
             kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1);
   latency_audio_stream->PrintResults();
 }
 
 }  // namespace webrtc