Low-latency AudioOutputStream implementation based on WASAPI for Windows.

media/audio/win/audio_low_latency_output_win_unittest.cc

+// Copyright (c) 2011 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+#include <windows.h>
+#include <mmsystem.h>
+
+#include "base/basictypes.h"
+#include "base/environment.h"
+#include "base/memory/scoped_ptr.h"
+#include "base/message_loop.h"
+#include "base/test/test_timeouts.h"
+#include "base/time.h"
+#include "base/win/scoped_com_initializer.h"
+#include "media/audio/audio_io.h"
+#include "media/audio/audio_manager.h"
+#include "media/audio/win/audio_low_latency_output_win.h"
+#include "media/base/seekable_buffer.h"
+#include "media/base/test_data_util.h"
+#include "testing/gmock_mutant.h"
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+
+using ::testing::_;
+using ::testing::AnyNumber;
+using ::testing::Between;
+using ::testing::CreateFunctor;
+using ::testing::DoAll;
+using ::testing::Gt;
+using ::testing::InvokeWithoutArgs;
+using ::testing::NotNull;
+using ::testing::Return;
+using base::win::ScopedCOMInitializer;
+
+namespace media {
+
+static const char kSpeechFile_16b_s_48k[] = "speech_16b_stereo_48kHz.raw";
+static const char kSpeechFile_16b_s_44k[] = "speech_16b_stereo_44kHz.raw";
+static const size_t kFileDurationMs = 20000;
+
+static const size_t kMaxDeltaSamples = 1000;
+static const char* kDeltaTimeMsFileName = "delta_times_ms.txt";
+
+MATCHER_P(HasValidDelay, value, "") {
+  // It is difficult to come up with a perfect test condition for the delay
+  // estimation. For now, verify that the produced output delay is always
+  // larger than the selected buffer size.
+  return arg.hardware_delay_bytes > value.hardware_delay_bytes;
+}
+
+class MockAudioSourceCallback : public AudioOutputStream::AudioSourceCallback {
+ public:
+  MOCK_METHOD4(OnMoreData, uint32(AudioOutputStream* stream,
+                                  uint8* dest,
+                                  uint32 max_size,
+                                  AudioBuffersState buffers_state));
+  MOCK_METHOD2(OnError, void(AudioOutputStream* stream, int code));
+};
+
+// This audio source implementation should be used for manual tests only since
+// it takes about 20 seconds to play out a file.
+class ReadFromFileAudioSource : public AudioOutputStream::AudioSourceCallback {
+ public:
+  explicit ReadFromFileAudioSource(const std::string& name)
+    : pos_(0),
+      previous_call_time_(base::Time::Now()),
+      text_file_(fopen(kDeltaTimeMsFileName, "wt")),

[Paweł Hajdan Jr., 2011/11/04 11:32:11]

nit: Why not use base/file_util? Up to you, but I highly recommend it if
possible.

[tommi (sloooow) - chröme, 2011/11/07 11:47:03]

+1

[henrika (OOO until Aug 14), 2011/11/07 14:09:45]

Now uses base/file_util and base/path_service.

+      elements_to_write_(0) {
+    // Reads a test file from media/test/data directory and stores it in
+    // a scoped_array.
+    ReadTestDataFile(name, &file_, &file_size_);
+    file_size_ = file_size_;
+
+    // Creates an array that will store delta times between callbacks.
+    // The content of this array will be written to a text file at
+    // destruction and can then be used for off-line analysis of the exact
+    // timing of callbacks.
+    delta_times_.reset(new int[kMaxDeltaSamples]);
+  }
+
+  virtual ~ReadFromFileAudioSource() {
+    // Write the array which contains delta times to a text file.
+    size_t elements_written = 0;
+    while (elements_written < elements_to_write_) {
+      fprintf(text_file_, "%d\n", delta_times_[elements_written]);
+      elements_written++;

[tommi (sloooow) - chröme, 2011/11/07 11:47:03]

nit: ++elements_written

[henrika (OOO until Aug 14), 2011/11/07 14:09:45]

Big difference ;-)

Done.

+    }
+    fclose(text_file_);
+  }
+
+  // AudioOutputStream::AudioSourceCallback implementation.
+  virtual uint32 OnMoreData(AudioOutputStream* stream,
+                            uint8* dest,
+                            uint32 max_size,
+                            AudioBuffersState buffers_state) {
+    // Store time difference between two successive callbacks in an array.
+    // These values will be written to a file in the destructor.
+    int diff = (base::Time::Now() - previous_call_time_).InMilliseconds();
+    previous_call_time_ = base::Time::Now();
+    if (elements_to_write_ < kMaxDeltaSamples) {
+      delta_times_[elements_to_write_] = diff;
+      elements_to_write_++;

[tommi (sloooow) - chröme, 2011/11/07 11:47:03]

nit: and here

[henrika (OOO until Aug 14), 2011/11/07 14:09:45]

Done.

+    }
+
+    // Use samples read from a data file and fill up the audio buffer
+    // provided to us in the callback.
+    if (pos_ + static_cast<int>(max_size) > file_size_)
+      max_size = file_size_ - pos_;
+    if (max_size) {
+      memcpy(dest, &file_[pos_], max_size);
+      pos_ += max_size;
+    }
+    return max_size;
+  }
+
+  virtual void OnError(AudioOutputStream* stream, int code) {}
+
+  int file_size() { return file_size_; }
+
+ private:
+  scoped_array<uint8> file_;
+  scoped_array<int> delta_times_;
+  int file_size_;
+  int pos_;
+  base::Time previous_call_time_;
+  FILE* text_file_;
+  size_t elements_to_write_;
+};
+
+// Convenience method which ensures that we are not running on the build
+// bots and that at least one valid output device can be found.
+static bool CanRunAudioTests() {
+  scoped_ptr<base::Environment> env(base::Environment::Create());
+  if (env->HasVar("CHROME_HEADLESS"))
+    return false;
+  AudioManager* audio_man = AudioManager::GetAudioManager();
+  if (NULL == audio_man)
+    return false;
+  // TODO(henrika): note that we use Wave today to query the number of
+  // existing output devices.
+  return audio_man->HasAudioOutputDevices();
+}
+
+// Convenience method which creates a default AudioOutputStream object but
+// also allows the user to modify the default settings.
+class AudioOutputStreamWrapper {
+ public:
+  AudioOutputStreamWrapper()
+      : com_init_(ScopedCOMInitializer::kMTA),
+        audio_man_(AudioManager::GetAudioManager()),
+        format_(AudioParameters::AUDIO_PCM_LOW_LATENCY),
+        channel_layout_(CHANNEL_LAYOUT_STEREO),
+        bits_per_sample_(16) {
+    // Use native/mixing sample rate and 10ms frame size as default.
+    sample_rate_ = static_cast<int>(
+        WASAPIAudioOutputStream::HardwareSampleRate(eConsole));
+    samples_per_packet_ = sample_rate_ / 100;
+    DCHECK(sample_rate_);
+  }
+
+  ~AudioOutputStreamWrapper() {}
+
+  // Creates AudioOutputStream object using default parameters.
+  AudioOutputStream* Create() {
+    return CreateOutputStream();
+  }
+
+  // Creates AudioOutputStream object using non-default parameters where the
+  // frame size is modified.
+  AudioOutputStream* Create(int samples_per_packet) {
+    samples_per_packet_ = samples_per_packet;
+    return CreateOutputStream();
+  }
+
+  // Creates AudioOutputStream object using non-default parameters where the
+  // channel layout is modified.
+  AudioOutputStream* Create(ChannelLayout channel_layout) {
+    channel_layout_ = channel_layout;
+    return CreateOutputStream();
+  }
+
+  AudioParameters::Format format() const { return format_; }
+  int channels() const { return ChannelLayoutToChannelCount(channel_layout_); }
+  int bits_per_sample() const { return bits_per_sample_; }
+  int sample_rate() const { return sample_rate_; }
+  int samples_per_packet() const { return samples_per_packet_; }
+
+ private:
+  AudioOutputStream* CreateOutputStream() {
+    AudioOutputStream* aos = audio_man_->MakeAudioOutputStream(
+        AudioParameters(format_, channel_layout_, sample_rate_,
+                        bits_per_sample_, samples_per_packet_));
+    EXPECT_TRUE(aos);
+    return aos;
+  }
+
+  ScopedCOMInitializer com_init_;
+  AudioManager* audio_man_;
+  AudioParameters::Format format_;
+  ChannelLayout channel_layout_;
+  int bits_per_sample_;
+  int sample_rate_;
+  int samples_per_packet_;
+};
+
+// Convenience method which creates a default AudioOutputStream object.
+static AudioOutputStream* CreateDefaultAudioOutputStream() {
+  AudioOutputStreamWrapper aosw;
+  AudioOutputStream* aos = aosw.Create();
+  return aos;
+}
+
+static void QuitMessageLoop(base::MessageLoopProxy* proxy) {
+  proxy->PostTask(FROM_HERE, new MessageLoop::QuitTask());
+}
+
+// Verify that we can retrieve the current hardware/mixing sample rate
+// for all supported device roles. The ERole enumeration defines constants
+// that indicate the role that the system/user has assigned to an audio
+// endpoint device.
+// TODO(henrika): modify this test when we support full device enumeration.
+TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestHardwareSampleRate) {
+  if (!CanRunAudioTests())
+    return;
+
+  ScopedCOMInitializer com_init(ScopedCOMInitializer::kMTA);
+
+  // Default device intended for games, system notification sounds,
+  // and voice commands.
+  int fs = static_cast<int>(
+      WASAPIAudioOutputStream::HardwareSampleRate(eConsole));
+  EXPECT_GE(fs, 0);
+
+  // Default communication device intended for e.g. VoIP communication.
+  fs = static_cast<int>(
+      WASAPIAudioOutputStream::HardwareSampleRate(eCommunications));
+  EXPECT_GE(fs, 0);
+
+  // Multimedia device for music, movies and live music recording.
+  fs = static_cast<int>(
+      WASAPIAudioOutputStream::HardwareSampleRate(eMultimedia));
+  EXPECT_GE(fs, 0);
+}
+
+// Test Create(), Close() calling sequence.
+TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestCreateAndClose) {
+  if (!CanRunAudioTests())
+    return;
+  AudioOutputStream* aos = CreateDefaultAudioOutputStream();
+  aos->Close();
+}
+
+// Test Open(), Close() calling sequence.
+TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestOpenAndClose) {
+  if (!CanRunAudioTests())
+    return;
+  AudioOutputStream* aos = CreateDefaultAudioOutputStream();
+  EXPECT_TRUE(aos->Open());
+  aos->Close();
+}
+
+// Test Open(), Start(), Close() calling sequence.
+TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestOpenStartAndClose) {
+  if (!CanRunAudioTests())
+    return;
+  AudioOutputStream* aos = CreateDefaultAudioOutputStream();
+  EXPECT_TRUE(aos->Open());
+  MockAudioSourceCallback source;
+  EXPECT_CALL(source, OnError(aos, _))
+      .Times(0);
+  aos->Start(&source);
+  aos->Close();
+}
+
+// Test Open(), Start(), Stop(), Close() calling sequence.
+TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestOpenStartStopAndClose) {
+  if (!CanRunAudioTests())
+    return;
+  AudioOutputStream* aos = CreateDefaultAudioOutputStream();
+  EXPECT_TRUE(aos->Open());
+  MockAudioSourceCallback source;
+  EXPECT_CALL(source, OnError(aos, _))
+      .Times(0);
+  aos->Start(&source);
+  aos->Stop();
+  aos->Close();
+}
+
+// Test SetVolume(), GetVolume()
+TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestVolume) {
+  if (!CanRunAudioTests())
+    return;
+  AudioOutputStream* aos = CreateDefaultAudioOutputStream();
+
+  // Initial volume should be full volume (1.0).
+  double volume = 0.0;
+  aos->GetVolume(&volume);
+  EXPECT_TRUE(volume == 1.0);

[tommi (sloooow) - chröme, 2011/11/07 11:47:03]

EXPECT_EQ (below too)

[henrika (OOO until Aug 14), 2011/11/07 14:09:45]

Done.

+
+  // Verify some valid volume settings.
+  aos->SetVolume(0.0);
+  aos->GetVolume(&volume);
+  EXPECT_TRUE(volume == 0.0);
+
+  aos->SetVolume(0.5);
+  aos->GetVolume(&volume);
+  EXPECT_TRUE(volume == 0.5);
+
+  aos->SetVolume(1.0);
+  aos->GetVolume(&volume);
+  EXPECT_TRUE(volume == 1.0);
+
+  // Ensure that invalid volume setting have no effect.
+  aos->SetVolume(1.5);
+  aos->GetVolume(&volume);
+  EXPECT_TRUE(volume == 1.0);
+
+  aos->SetVolume(-0.5);
+  aos->GetVolume(&volume);
+  EXPECT_TRUE(volume == 1.0);
+
+  aos->Close();
+}
+
+// Test some additional calling sequences.
+TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestMiscCallingSequences) {
+  if (!CanRunAudioTests())
+    return;
+  AudioOutputStream* aos = CreateDefaultAudioOutputStream();
+  WASAPIAudioOutputStream* waos = static_cast<WASAPIAudioOutputStream*>(aos);
+
+  // Open(), Open() should fail the second time.
+  EXPECT_TRUE(aos->Open());
+  EXPECT_FALSE(aos->Open());

[tommi (sloooow) - chröme, 2011/11/07 11:47:03]

It is a bit confusing that Open fails the second time around since the state
_is_ open and you even call aos->Start() right after Open() returned false.

[henrika (OOO until Aug 14), 2011/11/07 14:09:45]

Good point. Modified. Second call to Open() now returns true and does nothing.

+
+  MockAudioSourceCallback source;
+
+  // Start(), Start() is a valid calling sequence (second call does nothing).
+  aos->Start(&source);
+  EXPECT_TRUE(waos->started());
+  aos->Start(&source);
+  EXPECT_TRUE(waos->started());
+
+  // Stop(), Stop() is a valid calling sequence (second call does nothing).
+  aos->Stop();
+  EXPECT_FALSE(waos->started());
+  aos->Stop();
+  EXPECT_FALSE(waos->started());
+
+  aos->Close();
+}
+
+// Use default packet size (10ms) and verify that rendering starts.
+TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestPacketSizeInMilliseconds) {
+  if (!CanRunAudioTests())
+    return;
+
+  MessageLoopForUI loop;
+  scoped_refptr<base::MessageLoopProxy> proxy(loop.message_loop_proxy());
+
+  MockAudioSourceCallback source;
+
+  // Create default WASAPI output stream which plays out in stereo using
+  // the shared mixing rate. The default buffer size is 10ms.
+  AudioOutputStreamWrapper aosw;
+  AudioOutputStream* aos = aosw.Create();
+  EXPECT_TRUE(aos->Open());
+
+  // Derive the expected size in bytes of each packet.
+  uint32 bytes_per_packet = aosw.channels() * aosw.samples_per_packet() *
+                           (aosw.bits_per_sample() / 8);
+
+  // Set up expected minimum delay estimation.
+  AudioBuffersState state(0, bytes_per_packet);
+
+  // Wait for the first callback and verify its parameters.
+  EXPECT_CALL(source, OnMoreData(aos, NotNull(), bytes_per_packet,
+                                 HasValidDelay(state)))
+      .WillOnce(
+          DoAll(
+              InvokeWithoutArgs(
+                  CreateFunctor(&QuitMessageLoop, proxy.get())),
+              Return(bytes_per_packet)));
+
+  aos->Start(&source);
+  loop.PostDelayedTask(FROM_HERE, new MessageLoop::QuitTask(),
+                       TestTimeouts::action_timeout_ms());
+  loop.Run();
+  aos->Stop();
+  aos->Close();
+}
+
+// Use a fixed packets size (independent of sample rate) and verify
+// that rendering starts.
+TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestPacketSizeInSamples) {
+  if (!CanRunAudioTests())
+    return;
+
+  MessageLoopForUI loop;
+  scoped_refptr<base::MessageLoopProxy> proxy(loop.message_loop_proxy());
+
+  MockAudioSourceCallback source;
+
+  // Create default WASAPI output stream which plays out in stereo using
+  // the shared mixing rate. The buffer size is set to 1024 samples.
+  AudioOutputStreamWrapper aosw;
+  AudioOutputStream* aos = aosw.Create(1024);
+  EXPECT_TRUE(aos->Open());
+
+  // Derive the expected size in bytes of each packet.
+  uint32 bytes_per_packet = aosw.channels() * aosw.samples_per_packet() *
+                           (aosw.bits_per_sample() / 8);
+
+  // Set up expected minimum delay estimation.
+  AudioBuffersState state(0, bytes_per_packet);
+
+  // Wait for the first callback and verify its parameters.
+  EXPECT_CALL(source, OnMoreData(aos, NotNull(), bytes_per_packet,
+                                 HasValidDelay(state)))
+      .WillOnce(
+          DoAll(
+              InvokeWithoutArgs(
+                  CreateFunctor(&QuitMessageLoop, proxy.get())),
+              Return(bytes_per_packet)));
+
+  aos->Start(&source);
+  loop.PostDelayedTask(FROM_HERE, new MessageLoop::QuitTask(),
+                       TestTimeouts::action_timeout_ms());
+  loop.Run();
+  aos->Stop();
+  aos->Close();
+}
+
+TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestMono) {
+  if (!CanRunAudioTests())
+    return;
+
+  MessageLoopForUI loop;
+  scoped_refptr<base::MessageLoopProxy> proxy(loop.message_loop_proxy());
+
+  MockAudioSourceCallback source;
+
+  // Create default WASAPI output stream which plays out in *mono* using
+  // the shared mixing rate. The default buffer size is 10ms.
+  AudioOutputStreamWrapper aosw;
+  AudioOutputStream* aos = aosw.Create(CHANNEL_LAYOUT_MONO);
+  EXPECT_TRUE(aos->Open());
+
+  // Derive the expected size in bytes of each packet.
+  uint32 bytes_per_packet = aosw.channels() * aosw.samples_per_packet() *
+                           (aosw.bits_per_sample() / 8);
+
+  // Set up expected minimum delay estimation.
+  AudioBuffersState state(0, bytes_per_packet);
+
+  EXPECT_CALL(source, OnMoreData(aos, NotNull(), bytes_per_packet,
+                                 HasValidDelay(state)))
+      .WillOnce(
+          DoAll(
+              InvokeWithoutArgs(
+                  CreateFunctor(&QuitMessageLoop, proxy.get())),
+              Return(bytes_per_packet)));
+
+  aos->Start(&source);
+  loop.PostDelayedTask(FROM_HERE, new MessageLoop::QuitTask(),
+                       TestTimeouts::action_timeout_ms());
+  loop.Run();
+  aos->Stop();
+  aos->Close();
+}
+
+// This test is intended for manual tests and should only be enabled
+// when it is required to store the captured data on a local file.
+// By default, GTest will print out YOU HAVE 1 DISABLED TEST.
+// To include disabled tests in test execution, just invoke the test program
+// with --gtest_also_run_disabled_tests or set the GTEST_ALSO_RUN_DISABLED_TESTS
+// environment variable to a value greater than 0.
+// The test files are approximately 20 seconds long.
+TEST(WinAudioOutputTest, DISABLED_WASAPIAudioOutputStreamReadFromFile) {

[tommi (sloooow) - chröme, 2011/11/07 11:47:03]

Is it necessary to have the test disabled since it also calls
CanRunAudioTests()?

[henrika (OOO until Aug 14), 2011/11/07 14:09:45]

Please advice. I figured that it was not OK to add a test that takes 20 seconds
by default.

[tommi (sloooow) - chröme, 2011/11/07 17:37:17]

Nope, you're right, 20 seconds is too long.  But do we need both DISABLED_ and
CanRunAudioTests (which checks another env variable)?

+  if (!CanRunAudioTests())
+    return;
+
+  AudioOutputStreamWrapper aosw;
+  AudioOutputStream* aos = aosw.Create();
+  EXPECT_TRUE(aos->Open());
+
+  std::string file_name;
+  if (aosw.sample_rate() == 48000) {
+    file_name = kSpeechFile_16b_s_48k;
+  } else if (aosw.sample_rate() == 44100) {
+    file_name = kSpeechFile_16b_s_44k;
+  } else if (aosw.sample_rate() == 96000) {
+    // Use 48kHz file at 96kHz as well. Will sound as Donald Duck.
+    file_name = kSpeechFile_16b_s_48k;
+  } else {
+    fprintf(stderr, "This test supports 44.1, 48kHz and 96kHz only.\n");

[Paweł Hajdan Jr., 2011/11/04 11:32:11]

Don't you want the test to fail then?

FAIL() << "This test ...";

Up to you.

[henrika (OOO until Aug 14), 2011/11/07 14:09:45]

Fixed. Thanks.

+    return;
+  }
+  ReadFromFileAudioSource file_source(file_name);
+  int file_duration_ms = kFileDurationMs;
+
+  fprintf(stderr, "               File name  : %s\n", file_name.c_str());

[Paweł Hajdan Jr., 2011/11/04 11:32:11]

nit: I think you should generally use LOG in favor of fprintf. Up to you and
other reviewers.

[tommi (sloooow) - chröme, 2011/11/07 11:47:03]

+1

[henrika (OOO until Aug 14), 2011/11/07 14:09:45]

Done.

+  fprintf(stderr, "               Sample rate: %d\n", aosw.sample_rate());
+  fprintf(stderr, "               File size  : %d\n", file_source.file_size());
+  fprintf(stderr, "               >> Listen to the file while playing...\n");
+
+  aos->Start(&file_source);
+  base::PlatformThread::Sleep(file_duration_ms);
+  aos->Stop();
+
+  fprintf(stderr, "               >> File playout has stopped.\n");
+  aos->Close();
+}
+
+}  // namespace media