Low-latency AudioOutputStream implementation based on WASAPI for Windows.

media/audio/win/audio_low_latency_output_win_unittest.cc

Index: media/audio/win/audio_low_latency_output_win_unittest.cc
===================================================================
--- media/audio/win/audio_low_latency_output_win_unittest.cc	(revision 0)
+++ media/audio/win/audio_low_latency_output_win_unittest.cc	(revision 0)
@@ -0,0 +1,527 @@
+// 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/test/test_timeouts.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/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+
+using ::testing::_;
+using ::testing::AnyNumber;
+using ::testing::Between;
+using ::testing::Gt;
+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";
+
+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) {
+    // 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_;
+  }
+
+  virtual ~ReadFromFileAudioSource() {}
+
+  // AudioOutputStream::AudioSourceCallback implementation.
+  virtual uint32 OnMoreData(AudioOutputStream* stream,
+                            uint8* dest,
+                            uint32 max_size,
+                            AudioBuffersState buffers_state) {
+    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_;
+  int file_size_;
+  int pos_;
+};
+
+// 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;
+}
+
+// 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);
+
+  // 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());
+
+  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();
+}
+
+TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestPacketSizes) {
+  if (!CanRunAudioTests())
+    return;
+
+  // 10 ms packet size.
+
+  // 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);
+
+  MockAudioSourceCallback source;
+
+  // We use 10ms packets and will run the test for ~100ms. Given that the
+  // startup sequence takes some time, it is reasonable to expect 5-12
+  // callbacks in this time period. All should ask for the same size and
+  // contain a valid delay estimate.
+  EXPECT_CALL(source, OnMoreData(aos, NotNull(), bytes_per_packet,
+                                 HasValidDelay(state)))
+      .Times(Between(5, 10))
+      .WillRepeatedly(Return(bytes_per_packet));
+
+  aos->Start(&source);
+  base::PlatformThread::Sleep(TestTimeouts::tiny_timeout_ms());
+  aos->Stop();
+
+  // Store current packet size (to be used in the subsequent tests).
+  int samples_per_packet_10ms = aosw.samples_per_packet();
+
+  aos->Close();
+
+  // 20 ms packet size.
+
+  aos = aosw.Create(2 * samples_per_packet_10ms);
+  EXPECT_TRUE(aos->Open());
+  bytes_per_packet = aosw.channels() * aosw.samples_per_packet() *
+                    (aosw.bits_per_sample() / 8);
+
+  EXPECT_CALL(source, OnMoreData(aos, NotNull(), bytes_per_packet,
+                                 HasValidDelay(state)))
+      .Times(Between(5, 10))
+      .WillRepeatedly(Return(bytes_per_packet));
+
+  aos->Start(&source);
+  base::PlatformThread::Sleep(2 * TestTimeouts::tiny_timeout_ms());

[Paweł Hajdan Jr., 2011/11/02 09:36:59]

Please don't multiply values received from TestTimeouts. Also, you can't rely on
it being really 20ms. If you need to do either of those things, you should use
needed time intervals explicitly.

[henrika (OOO until Aug 14), 2011/11/03 15:03:35]

Modified the design. Now waits for one callback instead and verifies that it was
OK. It is to error prone anyhow to count on a certain number of callbacks within
a given time interval. Hope you like the new approach.

+  aos->Stop();
+
+  aos->Close();
+
+  // 40 ms packet size.
+
+  aos = aosw.Create(4 * samples_per_packet_10ms);
+  EXPECT_TRUE(aos->Open());
+  bytes_per_packet = aosw.channels() * aosw.samples_per_packet() *
+                    (aosw.bits_per_sample() / 8);
+
+  EXPECT_CALL(source, OnMoreData(aos, NotNull(), bytes_per_packet,
+                                 HasValidDelay(state)))
+      .Times(Between(5, 10))
+      .WillRepeatedly(Return(bytes_per_packet));
+
+  aos->Start(&source);
+  base::PlatformThread::Sleep(4 * TestTimeouts::tiny_timeout_ms());
+  aos->Stop();
+
+  aos->Close();
+
+  // 50 ms packet size.
+
+  aos = aosw.Create(samples_per_packet_10ms * 5);
+  EXPECT_TRUE(aos->Open());
+  bytes_per_packet = aosw.channels() * aosw.samples_per_packet() *
+                    (aosw.bits_per_sample() / 8);
+
+  EXPECT_CALL(source, OnMoreData(aos, NotNull(), bytes_per_packet,
+                                 HasValidDelay(state)))
+      .Times(Between(5, 10))
+      .WillRepeatedly(Return(bytes_per_packet));
+
+  aos->Start(&source);
+  base::PlatformThread::Sleep(5 * TestTimeouts::tiny_timeout_ms());
+  aos->Stop();
+
+  aos->Close();
+
+  // 5 ms packet size.
+
+  aos = aosw.Create(samples_per_packet_10ms / 2);
+  EXPECT_TRUE(aos->Open());
+  bytes_per_packet = aosw.channels() * aosw.samples_per_packet() *
+                    (aosw.bits_per_sample() / 8);
+
+  EXPECT_CALL(source, OnMoreData(aos, NotNull(), bytes_per_packet,
+                                 HasValidDelay(state)))
+      .Times(Between(2 * 5, 2 * 10))
+      .WillRepeatedly(Return(bytes_per_packet));
+
+  aos->Start(&source);
+  base::PlatformThread::Sleep(TestTimeouts::tiny_timeout_ms());
+  aos->Stop();
+
+  aos->Close();
+
+  // 512 samples (independent of sample rate)
+
+  aos = aosw.Create(512);
+  EXPECT_TRUE(aos->Open());
+  bytes_per_packet = aosw.channels() * aosw.samples_per_packet() *
+                    (aosw.bits_per_sample() / 8);
+
+  EXPECT_CALL(source, OnMoreData(aos, NotNull(), bytes_per_packet,
+                                 HasValidDelay(state)))
+      .WillRepeatedly(Return(bytes_per_packet));
+
+  aos->Start(&source);
+  base::PlatformThread::Sleep(TestTimeouts::tiny_timeout_ms());
+  aos->Stop();
+
+  aos->Close();
+}
+
+TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestMonoStereo) {
+  if (!CanRunAudioTests())
+    return;
+
+  // CHANNEL_LAYOUT_MONO
+
+  // 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);
+
+  MockAudioSourceCallback source;
+
+  EXPECT_CALL(source, OnMoreData(aos, NotNull(), bytes_per_packet,
+                                 HasValidDelay(state)))
+      .Times(Between(5, 10))
+      .WillRepeatedly(Return(bytes_per_packet));
+
+  aos->Start(&source);
+  base::PlatformThread::Sleep(TestTimeouts::tiny_timeout_ms());
+  aos->Stop();
+  aos->Close();
+
+  // CHANNEL_LAYOUT_STEREO
+
+  // Create default WASAPI output stream which plays out in *stereo* using
+  // the shared mixing rate. The default buffer size is 10ms.
+  aos = aosw.Create(CHANNEL_LAYOUT_STEREO);
+  EXPECT_TRUE(aos->Open());
+
+  bytes_per_packet = aosw.channels() * aosw.samples_per_packet() *
+                    (aosw.bits_per_sample() / 8);
+
+  state.pending_bytes = 0;
+  state.hardware_delay_bytes = bytes_per_packet;
+
+  EXPECT_CALL(source, OnMoreData(aos, NotNull(), bytes_per_packet,
+                                 HasValidDelay(state)))
+      .Times(Between(5, 10))
+      .WillRepeatedly(Return(bytes_per_packet));
+
+  aos->Start(&source);
+  base::PlatformThread::Sleep(TestTimeouts::tiny_timeout_ms());
+  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) {
+  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 {
+    fprintf(stderr, "This test supports 44.1 and 48kHz only.\n");
+    return;
+  }
+
+  ReadFromFileAudioSource file_source(file_name);
+  fprintf(stderr, "               File name  : %s\n", file_name.c_str());
+  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(2 * TestTimeouts::action_timeout_ms());
+  aos->Stop();
+  fprintf(stderr, "               >> File playout has stopped.\n");
+  aos->Close();
+}
+
+}  // namespace media
Property changes on: media\audio\win\audio_low_latency_output_win_unittest.cc
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