Add basic resample support to WASAPIAudioInputStream.

media/audio/win/audio_low_latency_input_win_unittest.cc

Index: media/audio/win/audio_low_latency_input_win_unittest.cc
diff --git a/media/audio/win/audio_low_latency_input_win_unittest.cc b/media/audio/win/audio_low_latency_input_win_unittest.cc
index 7e16bae23fcaf65fb470ce310b5c13109c78021c..2ecbd8b5d04a10bc2b4fe75b1756e98045f2068f 100644
--- a/media/audio/win/audio_low_latency_input_win_unittest.cc
+++ b/media/audio/win/audio_low_latency_input_win_unittest.cc
@@ -4,10 +4,10 @@
 
 #include "media/audio/win/audio_low_latency_input_win.h"
 
-#include <windows.h>
 #include <mmsystem.h>
 #include <stddef.h>
 #include <stdint.h>
+#include <windows.h>
 
 #include <memory>
 
@@ -64,9 +64,7 @@ class FakeAudioInputCallback : public AudioInputStream::AudioInputCallback {
   int num_received_audio_frames() const { return num_received_audio_frames_; }
 
   // Waits until OnData() is called on another thread.
-  void WaitForData() {
-    data_event_.Wait();
-  }
+  void WaitForData() { data_event_.Wait(); }
 
   void OnData(AudioInputStream* stream,
               const AudioBus* src,
@@ -78,9 +76,7 @@ class FakeAudioInputCallback : public AudioInputStream::AudioInputCallback {
     data_event_.Signal();
   }
 
-  void OnError(AudioInputStream* stream) override {
-    error_ = true;
-  }
+  void OnError(AudioInputStream* stream) override { error_ = true; }
 
  private:
   int num_received_audio_frames_;
@@ -177,12 +173,17 @@ class AudioInputStreamWrapper {
     frames_per_buffer_ = default_params_.frames_per_buffer();
   }
 
+  AudioInputStreamWrapper(AudioManager* audio_manager,
+                          const AudioParameters& default_params)
+      : audio_man_(audio_manager), default_params_(default_params) {
+    EXPECT_EQ(format(), AudioParameters::AUDIO_PCM_LOW_LATENCY);
+    frames_per_buffer_ = default_params_.frames_per_buffer();
+  }
+
   ~AudioInputStreamWrapper() {}
 
   // Creates AudioInputStream object using default parameters.
-  AudioInputStream* Create() {
-    return CreateInputStream();
-  }
+  AudioInputStream* Create() { return CreateInputStream(); }
 
   // Creates AudioInputStream object using non-default parameters where the
   // frame size is modified.
@@ -225,8 +226,7 @@ static AudioInputStream* CreateDefaultAudioInputStream(
 
 class ScopedAudioInputStream {
  public:
-  explicit ScopedAudioInputStream(AudioInputStream* stream)
-      : stream_(stream) {}
+  explicit ScopedAudioInputStream(AudioInputStream* stream) : stream_(stream) {}
 
   ~ScopedAudioInputStream() {
     if (stream_)
@@ -239,9 +239,7 @@ class ScopedAudioInputStream {
     stream_ = NULL;
   }
 
-  AudioInputStream* operator->() {
-    return stream_;
-  }
+  AudioInputStream* operator->() { return stream_; }
 
   AudioInputStream* get() const { return stream_; }
 
@@ -404,8 +402,8 @@ TEST_F(WinAudioInputTest, WASAPIAudioInputStreamTestPacketSizes) {
   count = 0;
   ais.Reset(aisw.Create(2 * frames_per_buffer_10ms));
   EXPECT_TRUE(ais->Open());
-  bytes_per_packet = aisw.channels() * aisw.frames_per_buffer() *
-      (aisw.bits_per_sample() / 8);
+  bytes_per_packet =
+      aisw.channels() * aisw.frames_per_buffer() * (aisw.bits_per_sample() / 8);
 
   {
     base::RunLoop run_loop;
@@ -425,8 +423,8 @@ TEST_F(WinAudioInputTest, WASAPIAudioInputStreamTestPacketSizes) {
   count = 0;
   ais.Reset(aisw.Create(frames_per_buffer_10ms / 2));
   EXPECT_TRUE(ais->Open());
-  bytes_per_packet = aisw.channels() * aisw.frames_per_buffer() *
-    (aisw.bits_per_sample() / 8);
+  bytes_per_packet =
+      aisw.channels() * aisw.frames_per_buffer() * (aisw.bits_per_sample() / 8);
 
   {
     base::RunLoop run_loop;
@@ -483,11 +481,11 @@ TEST_F(WinAudioInputTest, DISABLED_WASAPIAudioInputStreamRecordToFile) {
   // Name of the output PCM file containing captured data. The output file
   // will be stored in the directory containing 'media_unittests.exe'.
   // Example of full name: \src\build\Debug\out_stereo_10sec.pcm.
-  const char* file_name = "out_stereo_10sec.pcm";
+  const char* file_name = "out_10sec.pcm";
 
   AudioInputStreamWrapper aisw(audio_manager_.get());
   ScopedAudioInputStream ais(aisw.Create());
-  EXPECT_TRUE(ais->Open());
+  ASSERT_TRUE(ais->Open());
 
   VLOG(0) << ">> Sample rate: " << aisw.sample_rate() << " [Hz]";
   WriteToFileAudioSink file_sink(file_name, aisw.bits_per_sample());
@@ -499,4 +497,38 @@ TEST_F(WinAudioInputTest, DISABLED_WASAPIAudioInputStreamRecordToFile) {
   ais.Close();
 }
 
+TEST_F(WinAudioInputTest, DISABLED_WASAPIAudioInputStreamResampleToFile) {
+  ABORT_AUDIO_TEST_IF_NOT(HasCoreAudioAndInputDevices(audio_manager_.get()));
+
+  // This is basically the same test as WASAPIAudioInputStreamRecordToFile
+  // except it forces use of a different sample rate than is preferred by
+  // the hardware.  This functionality is offered while we still have code
+  // that doesn't ask the lower levels for what the preferred audio parameters
+  // are (and previously depended on the old Wave API to do this automatically).
+
+  AudioParameters params;
+  ASSERT_TRUE(SUCCEEDED(CoreAudioUtil::GetPreferredAudioParameters(
+      AudioDeviceDescription::kDefaultDeviceId, false, &params)));
+
+  VLOG(0) << ">> Hardware sample rate: " << params.sample_rate() << " [Hz]";
+
+  params.set_sample_rate(params.sample_rate() == 48000 ? 44100 : 48000);
+  params.set_frames_per_buffer(params.sample_rate() / 100);
+
+  static const char kFileName[] = "out_10sec_resampled.pcm";
+
+  AudioInputStreamWrapper aisw(audio_manager_.get(), params);
+  ScopedAudioInputStream ais(aisw.Create());
+  ASSERT_TRUE(ais->Open());
+
+  VLOG(0) << ">> Resampled rate will be: " << aisw.sample_rate() << " [Hz]";
+  WriteToFileAudioSink file_sink(kFileName, aisw.bits_per_sample());
+  VLOG(0) << ">> Speak into the default microphone while recording.";
+  ais->Start(&file_sink);
+  base::PlatformThread::Sleep(TestTimeouts::action_timeout());
+  ais->Stop();
+  VLOG(0) << ">> Recording has stopped.";
+  ais.Close();
+}
+
 }  // namespace media