Low-latency AudioInputStream implementation based on WASAPI for Windows.

media/audio/win/audio_low_latency_input_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/test/test_timeouts.h"
+#include "media/audio/audio_io.h"
+#include "media/audio/audio_manager.h"
+#include "media/audio/win/audio_low_latency_input_win.h"
+#include "media/base/seekable_buffer.h"
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+
+using ::testing::Gt;
+using ::testing::AnyNumber;
+using ::testing::Between;
+using ::testing::NotNull;
+
+class MockAudioInputCallback : public AudioInputStream::AudioInputCallback {
+ public:
+  MOCK_METHOD4(OnData, void(AudioInputStream* stream,
+    const uint8* src, uint32 size,

[tommi (sloooow) - chröme, 2011/10/14 14:31:04]

indent

[henrika (OOO until Aug 14), 2011/10/17 12:08:24]

Done.

+    uint32 hardware_delay_bytes));
+  MOCK_METHOD1(OnClose, void(AudioInputStream* stream));
+  MOCK_METHOD2(OnError, void(AudioInputStream* stream, int code));
+};
+
+// This audio sink implementation should be used for manual tests only since
+// the recorded data is stored on a raw binary data file.
+class WriteToFileAudioSink : public AudioInputStream::AudioInputCallback {
+ public:
+  // Allocate space for ~10 seconds of data @ 48kHz in stereo:
+  // 2 bytes per sample, 2 channels, 10ms @ 48kHz, 10 seconds <=> 1920000 bytes.
+  static const size_t kMaxBufferSize = 2 * 2 * 480 * 100 * 10;
+
+  explicit WriteToFileAudioSink(const char* file_name)
+    : buffer_(0, kMaxBufferSize),
+    file_(fopen(file_name, "wb")),
+    bytes_to_write_(0) {
+  }
+
+  virtual ~WriteToFileAudioSink() {
+    size_t bytes_written = 0;
+    while (bytes_written < bytes_to_write_) {
+      const uint8* chunk;
+      size_t chunk_size;
+
+      // Stop writing if no more data is available.
+      if (!buffer_.GetCurrentChunk(&chunk, &chunk_size))
+        break;
+
+      // Write recorded data chunk to the file and prepare for next chunk.
+      fwrite(chunk, 1, chunk_size, file_);
+      buffer_.Seek(chunk_size);
+      bytes_written += chunk_size;
+    }
+    fclose(file_);
+  }
+
+  // AudioInputStream::AudioInputCallback implementation.
+  virtual void OnData(AudioInputStream* stream,

[tommi (sloooow) - chröme, 2011/10/14 14:31:04]

strange indentation in this function

[henrika (OOO until Aug 14), 2011/10/17 12:08:24]

Done.

+    const uint8* src, uint32 size,
+    uint32 hardware_delay_bytes) {
+      // Store data data in a temporary buffer to avoid making blocking
+      // fwrite() calls in the audio callback. The complete buffer will be
+      // written to file in the destructor.
+      if (buffer_.Append(src, size)) {
+        bytes_to_write_ += size;
+      }
+  }
+
+  virtual void OnClose(AudioInputStream* stream) {}
+  virtual void OnError(AudioInputStream* stream, int code) {}
+
+ private:
+  media::SeekableBuffer buffer_;
+  FILE* file_;
+  size_t bytes_to_write_;
+};
+
+// Convenience method which ensures that we are not running on the build
+// bots and that at least one valid input 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 input devices.
+  return audio_man->HasAudioInputDevices();
+}
+
+// Convenience method which creates a default AudioInputStream object but
+// also allows the user to modify the default settings.
+class AudioInputStreamWrapper {
+ public:
+  AudioInputStreamWrapper()
+      : 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>(
+        WASAPIAudioInputStream::HardwareSampleRate(eConsole));
+    samples_per_packet_ = sample_rate_ / 100;
+  }
+
+  ~AudioInputStreamWrapper() { }
+
+  // Creates AudioInputStream object using default parameters.
+  AudioInputStream* Create() {
+    return CreateInputStream();
+  }
+
+  // Creates AudioInputStream object using non-default parameters where the
+  // frame size is modified.
+  AudioInputStream* Create(int samples_per_packet) {
+    samples_per_packet_ = samples_per_packet;
+    return CreateInputStream();
+  }
+
+  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:
+  AudioInputStream* CreateInputStream() {
+    AudioInputStream* ais = audio_man_->MakeAudioInputStream(
+        AudioParameters(format_, channel_layout_, sample_rate_,
+                        bits_per_sample_, samples_per_packet_));
+    EXPECT_TRUE(ais);
+    return ais;
+  }
+
+  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 AudioInputStream object.
+static AudioInputStream* CreateDefaultAudioInputStream() {
+  AudioInputStreamWrapper aisw;
+  AudioInputStream* ais = aisw.Create();
+  return ais;
+}
+
+// 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 suport full device enumeration.
+TEST(WinAudioInputTest, WASAPIAudioInputStreamHardwareSampleRate) {
+  if (!CanRunAudioTests())
+    return;
+
+  // Default device intended for games, system notification sounds,
+  // and voice commands.
+  int fs = static_cast<int>(
+      WASAPIAudioInputStream::HardwareSampleRate(eConsole));
+  EXPECT_GE(fs, 0);
+
+  // Default communication device intended for e.g. VoIP communication.
+  fs = static_cast<int>(
+      WASAPIAudioInputStream::HardwareSampleRate(eCommunications));
+  EXPECT_GE(fs, 0);
+
+  // Multimedia device for music, movies and live music recording.
+  fs = static_cast<int>(
+      WASAPIAudioInputStream::HardwareSampleRate(eMultimedia));
+  EXPECT_GE(fs, 0);
+}
+
+// Test Create(), Close() calling sequence.
+TEST(WinAudioInputTest, WASAPIAudioInputStreamCreateAndClose) {
+  if (!CanRunAudioTests())
+    return;
+  AudioInputStream* ais = CreateDefaultAudioInputStream();
+  ais->Close();
+}
+
+// Test Open(), Close() calling sequence.
+TEST(WinAudioInputTest, WASAPIAudioInputStreamOpenAndClose) {
+  if (!CanRunAudioTests())
+    return;
+  AudioInputStream* ais = CreateDefaultAudioInputStream();
+  EXPECT_TRUE(ais->Open());
+  ais->Close();
+}
+
+// Test Open(), Start(), Close() calling sequence.
+TEST(WinAudioInputTest, WASAPIAudioInputStreamOpenStartAndClose) {
+  if (!CanRunAudioTests())
+    return;
+  AudioInputStream* ais = CreateDefaultAudioInputStream();
+  EXPECT_TRUE(ais->Open());
+  MockAudioInputCallback sink;
+  ais->Start(&sink);
+  EXPECT_CALL(sink, OnClose(ais))
+      .Times(1);
+  ais->Close();
+}
+
+// Test Open(), Start(), Stop(), Close() calling sequence.
+TEST(WinAudioInputTest, WASAPIAudioInputStreamOpenStartStopAndClose) {
+  if (!CanRunAudioTests())
+    return;
+  AudioInputStream* ais = CreateDefaultAudioInputStream();
+  EXPECT_TRUE(ais->Open());
+  MockAudioInputCallback sink;
+  ais->Start(&sink);
+  ais->Stop();
+  EXPECT_CALL(sink, OnClose(ais))
+      .Times(1);
+  ais->Close();
+}
+
+// Test some additional calling sequences.
+TEST(MacAudioInputTest, WASAPIAudioInputStreamMiscCallingSequences) {
+  if (!CanRunAudioTests())
+    return;
+  AudioInputStream* ais = CreateDefaultAudioInputStream();
+  WASAPIAudioInputStream* wais = static_cast<WASAPIAudioInputStream*>(ais);
+
+  // Open(), Open() should fail the second time.
+  EXPECT_TRUE(ais->Open());
+  EXPECT_FALSE(ais->Open());
+
+  MockAudioInputCallback sink;
+
+  // Start(), Start() is a valid calling sequence (second call does nothing).
+  ais->Start(&sink);
+  EXPECT_TRUE(wais->started());
+  ais->Start(&sink);
+  EXPECT_TRUE(wais->started());
+
+  // Stop(), Stop() is a valid calling sequence (second call does nothing).
+  ais->Stop();
+  EXPECT_FALSE(wais->started());
+  ais->Stop();
+  EXPECT_FALSE(wais->started());
+
+  EXPECT_CALL(sink, OnClose(ais))
+    .Times(1);
+  ais->Close();
+}
+
+TEST(WinAudioInputTest, WASAPIAudioInputStreamTestPacketSizes) {
+  if (!CanRunAudioTests())
+    return;
+
+  // 10 ms packet size.
+
+  // Create default WASAPI input stream which records in stereo using
+  // the shared mixing rate. The default buffer size is 10ms.
+  AudioInputStreamWrapper aisw;
+  AudioInputStream* ais = aisw.Create();
+  EXPECT_TRUE(ais->Open());
+
+  MockAudioInputCallback sink;
+
+  // Derive the expected size in bytes of each recorded packet.
+  uint32 bytes_per_packet = aisw.channels() * aisw.samples_per_packet() *
+    (aisw.bits_per_sample() / 8);

[tommi (sloooow) - chröme, 2011/10/14 14:31:04]

indent

[henrika (OOO until Aug 14), 2011/10/17 12:08:24]

Done.

+
+  // 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 contain valid packets of
+  // the same size and a valid delay estimate.
+  EXPECT_CALL(sink, OnData(
+      ais, NotNull(), bytes_per_packet, Gt(bytes_per_packet)))
+      .Times(Between(5, 10));
+
+  ais->Start(&sink);
+  base::PlatformThread::Sleep(TestTimeouts::tiny_timeout_ms());
+  ais->Stop();
+
+  // Store current packet size (to be used in the subsequent tests).
+  int samples_per_packet_10ms = aisw.samples_per_packet();
+
+  EXPECT_CALL(sink, OnClose(ais))
+      .Times(1);
+  ais->Close();
+
+  // 20 ms packet size.
+
+  ais = aisw.Create(2 * samples_per_packet_10ms);
+  EXPECT_TRUE(ais->Open());
+  bytes_per_packet = aisw.channels() * aisw.samples_per_packet() *
+      (aisw.bits_per_sample() / 8);
+
+  EXPECT_CALL(sink, OnData(
+      ais, NotNull(), bytes_per_packet, Gt(bytes_per_packet)))
+      .Times(Between(5, 10));
+  ais->Start(&sink);
+  base::PlatformThread::Sleep(2 * TestTimeouts::tiny_timeout_ms());
+  ais->Stop();
+
+  EXPECT_CALL(sink, OnClose(ais))
+      .Times(1);
+  ais->Close();
+
+  // 5 ms packet size.
+
+  ais = aisw.Create(samples_per_packet_10ms / 2);
+  EXPECT_TRUE(ais->Open());
+  bytes_per_packet = aisw.channels() * aisw.samples_per_packet() *
+    (aisw.bits_per_sample() / 8);
+
+  EXPECT_CALL(sink, OnData(
+    ais, NotNull(), bytes_per_packet, Gt(bytes_per_packet)))
+    .Times(Between(2 * 5, 2 * 10));
+  ais->Start(&sink);
+  base::PlatformThread::Sleep(TestTimeouts::tiny_timeout_ms());
+  ais->Stop();
+
+  EXPECT_CALL(sink, OnClose(ais))
+    .Times(1);

[tommi (sloooow) - chröme, 2011/10/14 14:31:04]

indent

[henrika (OOO until Aug 14), 2011/10/17 12:08:24]

Done.

+  ais->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.
+TEST(WinAudioInputTest, DISABLED_WASAPIAudioInputStreamRecordToFile) {
+  if (!CanRunAudioTests())
+    return;
+
+  const char* file_name = "out_stereo_10sec.pcm";
+
+  AudioInputStreamWrapper aisw;
+  AudioInputStream* ais = aisw.Create();
+  EXPECT_TRUE(ais->Open());
+
+  fprintf(stderr, "               File name  : %s\n", file_name);
+  fprintf(stderr, "               Sample rate: %d\n", aisw.sample_rate());
+  WriteToFileAudioSink file_sink(file_name);
+  fprintf(stderr, "               >> Speak into the mic while recording...\n");
+  ais->Start(&file_sink);
+  base::PlatformThread::Sleep(TestTimeouts::action_timeout_ms());
+  ais->Stop();
+  fprintf(stderr, "               >> Recording has stopped.\n");
+  ais->Close();
+}