Revert of Used native deinterleaved and float point format for the input streams. (patchset #5 of https://codereview.chromium.org/501823002/)

Reason for revert:
It broke the mac asan bot, http://build.chromium.org/p/chromium.memory/builders/Mac%20ASan%2064%20Tests%20%281%29/builds/1976

Original issue's description:
> Used native deinterleaved and float point format for the input streams.
>
> If we call GetProperty of kAudioUnitProperty_StreamFormat before setting the format, the device will report kAudioFormatFlagsNativeFloatPacked | kLinearPCMFormatFlagIsNonInterleaved as the native format of the device, which is the same as the output.
>
> This patch changes the format to use kAudioFormatFlagsNativeFloatPacked | kLinearPCMFormatFlagIsNonInterleaved to open the device, so that we will avoid format flipping back and forth. Hope this optimization will help increase the stability of the input audio on Mac.
>
>
> BUG=404884
> TEST=media_unittests && https://webrtc.googlecode.com/svn-history/r5497/trunk/samples/js/demos/html/pc1.html, https://www.google.com/intl/en/chrome/demos/speech.html
>
> Committed: https://chromium.googlesource.com/chromium/src/+/1a9ce977642c7f2cc2e30d83757c42264f052f0b

TBR=dalecurtis@chromium.org,eroman@chromium.org
NOTREECHECKS=true
NOTRY=true
BUG=404884

Review URL: https://codereview.chromium.org/514773002

Cr-Commit-Position: refs/heads/master@{#292176}

5 files changed, 138 insertions, 205 deletions

diff --git a/media/audio/mac/audio_low_latency_input_mac.cc b/media/audio/mac/audio_low_latency_input_mac.cc
index fd835d7..f1dbdf7 100644
--- a/media/audio/mac/audio_low_latency_input_mac.cc
+++ b/media/audio/mac/audio_low_latency_input_mac.cc
@@ -10,7 +10,6 @@
 #include "base/logging.h"
 #include "base/mac/mac_logging.h"
 #include "media/audio/mac/audio_manager_mac.h"
-#include "media/base/audio_block_fifo.h"
 #include "media/base/audio_bus.h"
 #include "media/base/data_buffer.h"
 
@@ -47,46 +46,44 @@ AUAudioInputStream::AUAudioInputStream(AudioManagerMac* manager,
       started_(false),
       hardware_latency_frames_(0),
       number_of_channels_in_frame_(0),
-      output_bus_(AudioBus::Create(input_params)) {
+      fifo_(input_params.channels(),
+            number_of_frames_,
+            kNumberOfBlocksBufferInFifo) {
   DCHECK(manager_);
 
   // Set up the desired (output) format specified by the client.
   format_.mSampleRate = input_params.sample_rate();
   format_.mFormatID = kAudioFormatLinearPCM;
-  format_.mFormatFlags =
-      kAudioFormatFlagsNativeFloatPacked | kLinearPCMFormatFlagIsNonInterleaved;
-  size_t bytes_per_sample = sizeof(Float32);
-  format_.mBitsPerChannel = bytes_per_sample * 8;
+  format_.mFormatFlags = kLinearPCMFormatFlagIsPacked |
+                         kLinearPCMFormatFlagIsSignedInteger;
+  format_.mBitsPerChannel = input_params.bits_per_sample();
   format_.mChannelsPerFrame = input_params.channels();
-  format_.mFramesPerPacket = 1;
-  format_.mBytesPerFrame = bytes_per_sample;
-  format_.mBytesPerPacket = format_.mBytesPerFrame * format_.mFramesPerPacket;
+  format_.mFramesPerPacket = 1;  // uncompressed audio
+  format_.mBytesPerPacket = (format_.mBitsPerChannel *
+                             input_params.channels()) / 8;
+  format_.mBytesPerFrame = format_.mBytesPerPacket;
   format_.mReserved = 0;
 
   DVLOG(1) << "Desired ouput format: " << format_;
 
-  // Allocate AudioBufferList based on the number of channels.
-  audio_buffer_list_.reset(static_cast<AudioBufferList*>(
-      malloc(sizeof(UInt32) + input_params.channels() * sizeof(AudioBuffer))));
-  audio_buffer_list_->mNumberBuffers = input_params.channels();
+  // Derive size (in bytes) of the buffers that we will render to.
+  UInt32 data_byte_size = number_of_frames_ * format_.mBytesPerFrame;
+  DVLOG(1) << "Size of data buffer in bytes : " << data_byte_size;
 
   // Allocate AudioBuffers to be used as storage for the received audio.
   // The AudioBufferList structure works as a placeholder for the
   // AudioBuffer structure, which holds a pointer to the actual data buffer.
-  UInt32 data_byte_size = number_of_frames_ * format_.mBytesPerFrame;
-  CHECK_LE(static_cast<int>(data_byte_size * input_params.channels()),
-           media::AudioBus::CalculateMemorySize(input_params));
-  AudioBuffer* audio_buffer = audio_buffer_list_->mBuffers;
-  for (UInt32 i = 0; i < audio_buffer_list_->mNumberBuffers; ++i) {
-    audio_buffer[i].mNumberChannels = 1;
-    audio_buffer[i].mDataByteSize = data_byte_size;
-    audio_buffer[i].mData = output_bus_->channel(i);
-  }
-}
+  audio_data_buffer_.reset(new uint8[data_byte_size]);
+  audio_buffer_list_.mNumberBuffers = 1;
 
-AUAudioInputStream::~AUAudioInputStream() {
+  AudioBuffer* audio_buffer = audio_buffer_list_.mBuffers;
+  audio_buffer->mNumberChannels = input_params.channels();
+  audio_buffer->mDataByteSize = data_byte_size;
+  audio_buffer->mData = audio_data_buffer_.get();
 }
 
+AUAudioInputStream::~AUAudioInputStream() {}
+
 // Obtain and open the AUHAL AudioOutputUnit for recording.
 bool AUAudioInputStream::Open() {
   // Verify that we are not already opened.
@@ -168,6 +165,23 @@ bool AUAudioInputStream::Open() {
     return false;
   }
 
+  // Register the input procedure for the AUHAL.
+  // This procedure will be called when the AUHAL has received new data
+  // from the input device.
+  AURenderCallbackStruct callback;
+  callback.inputProc = InputProc;
+  callback.inputProcRefCon = this;
+  result = AudioUnitSetProperty(audio_unit_,
+                                kAudioOutputUnitProperty_SetInputCallback,
+                                kAudioUnitScope_Global,
+                                0,
+                                &callback,
+                                sizeof(callback));
+  if (result) {
+    HandleError(result);
+    return false;
+  }
+
   // Set up the the desired (output) format.
   // For obtaining input from a device, the device format is always expressed
   // on the output scope of the AUHAL's Element 1.
@@ -215,23 +229,6 @@ bool AUAudioInputStream::Open() {
     }
   }
 
-  // Register the input procedure for the AUHAL.
-  // This procedure will be called when the AUHAL has received new data
-  // from the input device.
-  AURenderCallbackStruct callback;
-  callback.inputProc = InputProc;
-  callback.inputProcRefCon = this;
-  result = AudioUnitSetProperty(audio_unit_,
-                                kAudioOutputUnitProperty_SetInputCallback,
-                                kAudioUnitScope_Global,
-                                0,
-                                &callback,
-                                sizeof(callback));
-  if (result) {
-    HandleError(result);
-    return false;
-  }
-
   // Finally, initialize the audio unit and ensure that it is ready to render.
   // Allocates memory according to the maximum number of audio frames
   // it can produce in response to a single render call.
@@ -345,9 +342,9 @@ void AUAudioInputStream::SetVolume(double volume) {
 
   Float32 volume_float32 = static_cast<Float32>(volume);
   AudioObjectPropertyAddress property_address = {
-      kAudioDevicePropertyVolumeScalar,
-      kAudioDevicePropertyScopeInput,
-      kAudioObjectPropertyElementMaster
+    kAudioDevicePropertyVolumeScalar,
+    kAudioDevicePropertyScopeInput,
+    kAudioObjectPropertyElementMaster
   };
 
   // Try to set the volume for master volume channel.
@@ -393,15 +390,15 @@ void AUAudioInputStream::SetVolume(double volume) {
 
 double AUAudioInputStream::GetVolume() {
   // Verify that we have a valid device.
-  if (input_device_id_ == kAudioObjectUnknown) {
+  if (input_device_id_ == kAudioObjectUnknown){
     NOTREACHED() << "Device ID is unknown";
     return 0.0;
   }
 
   AudioObjectPropertyAddress property_address = {
-      kAudioDevicePropertyVolumeScalar,
-      kAudioDevicePropertyScopeInput,
-      kAudioObjectPropertyElementMaster
+    kAudioDevicePropertyVolumeScalar,
+    kAudioDevicePropertyScopeInput,
+    kAudioObjectPropertyElementMaster
   };
 
   if (AudioObjectHasProperty(input_device_id_, &property_address)) {
@@ -409,8 +406,12 @@ double AUAudioInputStream::GetVolume() {
     // master channel.
     Float32 volume_float32 = 0.0;
     UInt32 size = sizeof(volume_float32);
-    OSStatus result = AudioObjectGetPropertyData(
-        input_device_id_, &property_address, 0, NULL, &size, &volume_float32);
+    OSStatus result = AudioObjectGetPropertyData(input_device_id_,
+                                                 &property_address,
+                                                 0,
+                                                 NULL,
+                                                 &size,
+                                                 &volume_float32);
     if (result == noErr)
       return static_cast<double>(volume_float32);
   } else {
@@ -471,8 +472,9 @@ OSStatus AUAudioInputStream::InputProc(void* user_data,
     return result;
 
   // Deliver recorded data to the consumer as a callback.
-  return audio_input->Provide(
-      number_of_frames, audio_input->audio_buffer_list(), time_stamp);
+  return audio_input->Provide(number_of_frames,
+                              audio_input->audio_buffer_list(),
+                              time_stamp);
 }
 
 OSStatus AUAudioInputStream::Provide(UInt32 number_of_frames,
@@ -489,39 +491,22 @@ OSStatus AUAudioInputStream::Provide(UInt32 number_of_frames,
 
   AudioBuffer& buffer = io_data->mBuffers[0];
   uint8* audio_data = reinterpret_cast<uint8*>(buffer.mData);
-  uint32 capture_delay_bytes = static_cast<uint32>(
-      (capture_latency_frames + 0.5) * format_.mBytesPerFrame);
+  uint32 capture_delay_bytes = static_cast<uint32>
+      ((capture_latency_frames + 0.5) * format_.mBytesPerFrame);
   DCHECK(audio_data);
   if (!audio_data)
     return kAudioUnitErr_InvalidElement;
 
-  // If the stream parameters change for any reason, we need to insert a FIFO
-  // since the OnMoreData() pipeline can't handle frame size changes.
-  if (number_of_frames != number_of_frames_) {
-    // Create a FIFO on the fly to handle any discrepancies in callback rates.
-    if (!fifo_) {
-      fifo_.reset(new AudioBlockFifo(output_bus_->channels(),
-                                     number_of_frames_,
-                                     kNumberOfBlocksBufferInFifo));
-    }
-  }
+  // Copy captured (and interleaved) data into FIFO.
+  fifo_.Push(audio_data, number_of_frames, format_.mBitsPerChannel / 8);
 
-  // When FIFO does not kick in, data will be directly passed to the callback.
-  if (!fifo_) {
-    CHECK_EQ(output_bus_->frames(), static_cast<int>(number_of_frames_));
-    sink_->OnData(
-        this, output_bus_.get(), capture_delay_bytes, normalized_volume);
-    return noErr;
-  }
-
-  // Compensate the audio delay caused by the FIFO.
-  capture_delay_bytes += fifo_->GetAvailableFrames() * format_.mBytesPerFrame;
-
-  fifo_->Push(output_bus_.get());
   // Consume and deliver the data when the FIFO has a block of available data.
-  while (fifo_->available_blocks()) {
-    const AudioBus* audio_bus = fifo_->Consume();
+  while (fifo_.available_blocks()) {
+    const AudioBus* audio_bus = fifo_.Consume();
     DCHECK_EQ(audio_bus->frames(), static_cast<int>(number_of_frames_));
+
+    // Compensate the audio delay caused by the FIFO.
+    capture_delay_bytes += fifo_.GetAvailableFrames() * format_.mBytesPerFrame;
     sink_->OnData(this, audio_bus, capture_delay_bytes, normalized_volume);
   }
 
@@ -534,9 +519,9 @@ int AUAudioInputStream::HardwareSampleRate() {
   UInt32 info_size = sizeof(device_id);
 
   AudioObjectPropertyAddress default_input_device_address = {
-      kAudioHardwarePropertyDefaultInputDevice,
-      kAudioObjectPropertyScopeGlobal,
-      kAudioObjectPropertyElementMaster
+    kAudioHardwarePropertyDefaultInputDevice,
+    kAudioObjectPropertyScopeGlobal,
+    kAudioObjectPropertyElementMaster
   };
   OSStatus result = AudioObjectGetPropertyData(kAudioObjectSystemObject,
                                                &default_input_device_address,
@@ -551,8 +536,10 @@ int AUAudioInputStream::HardwareSampleRate() {
   info_size = sizeof(nominal_sample_rate);
 
   AudioObjectPropertyAddress nominal_sample_rate_address = {
-      kAudioDevicePropertyNominalSampleRate, kAudioObjectPropertyScopeGlobal,
-      kAudioObjectPropertyElementMaster};
+    kAudioDevicePropertyNominalSampleRate,
+    kAudioObjectPropertyScopeGlobal,
+    kAudioObjectPropertyElementMaster
+  };
   result = AudioObjectGetPropertyData(device_id,
                                       &nominal_sample_rate_address,
                                       0,
@@ -585,9 +572,9 @@ double AUAudioInputStream::GetHardwareLatency() {
 
   // Get input audio device latency.
   AudioObjectPropertyAddress property_address = {
-      kAudioDevicePropertyLatency,
-      kAudioDevicePropertyScopeInput,
-      kAudioObjectPropertyElementMaster
+    kAudioDevicePropertyLatency,
+    kAudioDevicePropertyScopeInput,
+    kAudioObjectPropertyElementMaster
   };
   UInt32 device_latency_frames = 0;
   size = sizeof(device_latency_frames);
@@ -599,19 +586,19 @@ double AUAudioInputStream::GetHardwareLatency() {
                                       &device_latency_frames);
   DLOG_IF(WARNING, result != noErr) << "Could not get audio device latency.";
 
-  return static_cast<double>((audio_unit_latency_sec * format_.mSampleRate) +
-                             device_latency_frames);
+  return static_cast<double>((audio_unit_latency_sec *
+      format_.mSampleRate) + device_latency_frames);
 }
 
 double AUAudioInputStream::GetCaptureLatency(
     const AudioTimeStamp* input_time_stamp) {
   // Get the delay between between the actual recording instant and the time
   // when the data packet is provided as a callback.
-  UInt64 capture_time_ns =
-      AudioConvertHostTimeToNanos(input_time_stamp->mHostTime);
+  UInt64 capture_time_ns = AudioConvertHostTimeToNanos(
+      input_time_stamp->mHostTime);
   UInt64 now_ns = AudioConvertHostTimeToNanos(AudioGetCurrentHostTime());
-  double delay_frames = static_cast<double>(1e-9 * (now_ns - capture_time_ns) *
-                                            format_.mSampleRate);
+  double delay_frames = static_cast<double>
+      (1e-9 * (now_ns - capture_time_ns) * format_.mSampleRate);
 
   // Total latency is composed by the dynamic latency and the fixed
   // hardware latency.
@@ -621,14 +608,18 @@ double AUAudioInputStream::GetCaptureLatency(
 int AUAudioInputStream::GetNumberOfChannelsFromStream() {
   // Get the stream format, to be able to read the number of channels.
   AudioObjectPropertyAddress property_address = {
-      kAudioDevicePropertyStreamFormat,
-      kAudioDevicePropertyScopeInput,
-      kAudioObjectPropertyElementMaster
+    kAudioDevicePropertyStreamFormat,
+    kAudioDevicePropertyScopeInput,
+    kAudioObjectPropertyElementMaster
   };
   AudioStreamBasicDescription stream_format;
   UInt32 size = sizeof(stream_format);
-  OSStatus result = AudioObjectGetPropertyData(
-      input_device_id_, &property_address, 0, NULL, &size, &stream_format);
+  OSStatus result = AudioObjectGetPropertyData(input_device_id_,
+                                               &property_address,
+                                               0,
+                                               NULL,
+                                               &size,
+                                               &stream_format);
   if (result != noErr) {
     DLOG(WARNING) << "Could not get stream format";
     return 0;
@@ -638,8 +629,8 @@ int AUAudioInputStream::GetNumberOfChannelsFromStream() {
 }
 
 void AUAudioInputStream::HandleError(OSStatus err) {
-  NOTREACHED() << "error " << GetMacOSStatusErrorString(err) << " (" << err
-               << ")";
+  NOTREACHED() << "error " << GetMacOSStatusErrorString(err)
+               << " (" << err << ")";
   if (sink_)
     sink_->OnError(this);
 }
@@ -647,12 +638,13 @@ void AUAudioInputStream::HandleError(OSStatus err) {
 bool AUAudioInputStream::IsVolumeSettableOnChannel(int channel) {
   Boolean is_settable = false;
   AudioObjectPropertyAddress property_address = {
-      kAudioDevicePropertyVolumeScalar,
-      kAudioDevicePropertyScopeInput,
-      static_cast<UInt32>(channel)
+    kAudioDevicePropertyVolumeScalar,
+    kAudioDevicePropertyScopeInput,
+    static_cast<UInt32>(channel)
   };
-  OSStatus result = AudioObjectIsPropertySettable(
-      input_device_id_, &property_address, &is_settable);
+  OSStatus result = AudioObjectIsPropertySettable(input_device_id_,
+                                                  &property_address,
+                                                  &is_settable);
   return (result == noErr) ? is_settable : false;
 }
 
diff --git a/media/audio/mac/audio_low_latency_input_mac.h b/media/audio/mac/audio_low_latency_input_mac.h
index db444aa..c8e43fa 100644
--- a/media/audio/mac/audio_low_latency_input_mac.h
+++ b/media/audio/mac/audio_low_latency_input_mac.h
@@ -45,10 +45,10 @@
 #include "media/audio/agc_audio_stream.h"
 #include "media/audio/audio_io.h"
 #include "media/audio/audio_parameters.h"
+#include "media/base/audio_block_fifo.h"
 
 namespace media {
 
-class AudioBlockFifo;
 class AudioBus;
 class AudioManagerMac;
 class DataBuffer;
@@ -78,7 +78,7 @@ class AUAudioInputStream : public AgcAudioStream<AudioInputStream> {
 
   bool started() const { return started_; }
   AudioUnit audio_unit() { return audio_unit_; }
-  AudioBufferList* audio_buffer_list() { return audio_buffer_list_.get(); }
+  AudioBufferList* audio_buffer_list() { return &audio_buffer_list_; }
 
  private:
   // AudioOutputUnit callback.
@@ -90,8 +90,7 @@ class AUAudioInputStream : public AgcAudioStream<AudioInputStream> {
                             AudioBufferList* io_data);
 
   // Pushes recorded data to consumer of the input audio stream.
-  OSStatus Provide(UInt32 number_of_frames,
-                   AudioBufferList* io_data,
+  OSStatus Provide(UInt32 number_of_frames, AudioBufferList* io_data,
                    const AudioTimeStamp* time_stamp);
 
   // Gets the fixed capture hardware latency and store it during initialization.
@@ -133,7 +132,11 @@ class AUAudioInputStream : public AgcAudioStream<AudioInputStream> {
   AudioDeviceID input_device_id_;
 
   // Provides a mechanism for encapsulating one or more buffers of audio data.
-  scoped_ptr<AudioBufferList, base::FreeDeleter> audio_buffer_list_;
+  AudioBufferList audio_buffer_list_;
+
+  // Temporary storage for recorded data. The InputProc() renders into this
+  // array as soon as a frame of the desired buffer size has been recorded.
+  scoped_ptr<uint8[]> audio_data_buffer_;
 
   // True after successfull Start(), false after successful Stop().
   bool started_;
@@ -145,12 +148,8 @@ class AUAudioInputStream : public AgcAudioStream<AudioInputStream> {
   // when querying the volume of each channel.
   int number_of_channels_in_frame_;
 
-  // Dynamically allocated FIFO used when CoreAudio asks for unexpected frame
-  // sizes.
-  scoped_ptr<AudioBlockFifo> fifo_;
-
-  // AudioBus for delievering data via AudioSourceCallback::OnData().
-  scoped_ptr<AudioBus> output_bus_;
+  // FIFO used to accumulates recorded data.
+  media::AudioBlockFifo fifo_;
 
   // Used to defer Start() to workaround http://crbug.com/160920.
   base::CancelableClosure deferred_start_cb_;
diff --git a/media/base/audio_block_fifo.cc b/media/base/audio_block_fifo.cc
index 9634574..5e8c9b3 100644
--- a/media/base/audio_block_fifo.cc
+++ b/media/base/audio_block_fifo.cc
@@ -22,8 +22,7 @@ AudioBlockFifo::AudioBlockFifo(int channels, int frames, int blocks)
   }
 }
 
-AudioBlockFifo::~AudioBlockFifo() {
-}
+AudioBlockFifo::~AudioBlockFifo() {}
 
 void AudioBlockFifo::Push(const void* source,
                           int frames,
@@ -47,39 +46,20 @@ void AudioBlockFifo::Push(const void* source,
     // Deinterleave the content to the FIFO and update the |write_pos_|.
     current_block->FromInterleavedPartial(
         source_ptr, write_pos_, push_frames, bytes_per_sample);
+    write_pos_ = (write_pos_ + push_frames) % block_frames_;
+    if (!write_pos_) {
+      // The current block is completely filled, increment |write_block_| and
+      // |available_blocks_|.
+      write_block_ = (write_block_ + 1) % audio_blocks_.size();
+      ++available_blocks_;
+    }
 
-    UpdatePosition(push_frames);
     source_ptr += push_frames * bytes_per_sample * current_block->channels();
     frames_to_push -= push_frames;
     DCHECK_GE(frames_to_push, 0);
   }
 }
 
-void AudioBlockFifo::Push(const AudioBus* source) {
-  DCHECK(source);
-  DCHECK_LT(available_blocks_, static_cast<int>(audio_blocks_.size()));
-
-  int source_start_frame = 0;
-  while (source_start_frame < source->frames()) {
-    // Get the current write block.
-    AudioBus* current_block = audio_blocks_[write_block_];
-    DCHECK_EQ(source->channels(), current_block->channels());
-
-    // Figure out what segment sizes we need when adding the new content to
-    // the FIFO.
-    const int push_frames = std::min(block_frames_ - write_pos_,
-                                     source->frames() - source_start_frame);
-
-    // Copy the data to FIFO.
-    source->CopyPartialFramesTo(
-        source_start_frame, push_frames, write_pos_, current_block);
-
-    UpdatePosition(push_frames);
-    source_start_frame += push_frames;
-    DCHECK_LE(source_start_frame, source->frames());
-  }
-}
-
 const AudioBus* AudioBlockFifo::Consume() {
   DCHECK(available_blocks_);
   AudioBus* audio_bus = audio_blocks_[read_block_];
@@ -106,14 +86,4 @@ int AudioBlockFifo::GetUnfilledFrames() const {
   return unfilled_frames;
 }
 
-void AudioBlockFifo::UpdatePosition(int push_frames) {
-  write_pos_ = (write_pos_ + push_frames) % block_frames_;
-  if (!write_pos_) {
-    // The current block is completely filled, increment |write_block_| and
-    // |available_blocks_|.
-    write_block_ = (write_block_ + 1) % audio_blocks_.size();
-    ++available_blocks_;
-  }
-}
-
 }  // namespace media
diff --git a/media/base/audio_block_fifo.h b/media/base/audio_block_fifo.h
index 56a2fe0..fdb5cef 100644
--- a/media/base/audio_block_fifo.h
+++ b/media/base/audio_block_fifo.h
@@ -28,10 +28,6 @@ class MEDIA_EXPORT AudioBlockFifo {
   // Push() will crash if the allocated space is insufficient.
   void Push(const void* source, int frames, int bytes_per_sample);
 
-  // Pushes the audio data from |source| to the FIFO.
-  // Push() will crash if the allocated space is insufficient.
-  void Push(const AudioBus* source);
-
   // Consumes a block of audio from the FIFO.  Returns an AudioBus which
   // contains the consumed audio data to avoid copying.
   // Consume() will crash if the FIFO does not contain a block of data.
@@ -50,9 +46,6 @@ class MEDIA_EXPORT AudioBlockFifo {
   int GetUnfilledFrames() const;
 
  private:
-  // Helper method to update the indexes in Push methods.
-  void UpdatePosition(int push_frames);
-
   // The actual FIFO is a vector of audio buses.
   ScopedVector<AudioBus> audio_blocks_;
 
diff --git a/media/base/audio_block_fifo_unittest.cc b/media/base/audio_block_fifo_unittest.cc
index f1ed228..8e8b5e0 100644
--- a/media/base/audio_block_fifo_unittest.cc
+++ b/media/base/audio_block_fifo_unittest.cc
@@ -8,48 +8,29 @@
 namespace media {
 
 class AudioBlockFifoTest : public testing::Test {
- protected:
+ public:
   AudioBlockFifoTest() {}
   virtual ~AudioBlockFifoTest() {}
 
- private:
-  DISALLOW_COPY_AND_ASSIGN(AudioBlockFifoTest);
-};
-
-class AudioBlockFifoFormatTest : public AudioBlockFifoTest,
-                                 public testing::WithParamInterface<bool> {
- protected:
-  void PushAndVerify(AudioBlockFifo* fifo,
-                     int frames_to_push,
-                     int channels,
-                     int block_frames,
-                     int max_frames) {
+  void PushAndVerify(AudioBlockFifo* fifo, int frames_to_push,
+                     int channels, int block_frames, int max_frames) {
     const int bytes_per_sample = 2;
     const int data_byte_size = bytes_per_sample * channels * frames_to_push;
-    if (GetParam()) {
-      scoped_ptr<media::AudioBus> data =
-          AudioBus::Create(channels, frames_to_push);
-      for (int filled_frames = max_frames - fifo->GetUnfilledFrames();
-           filled_frames + frames_to_push <= max_frames;) {
-        fifo->Push(data.get());
-        filled_frames += frames_to_push;
-        EXPECT_EQ(max_frames - filled_frames, fifo->GetUnfilledFrames());
-        EXPECT_EQ(static_cast<int>(filled_frames / block_frames),
-                  fifo->available_blocks());
-      }
-    } else {
-      scoped_ptr<uint8[]> data(new uint8[data_byte_size]);
-      memset(data.get(), 0, data_byte_size);
-      for (int filled_frames = max_frames - fifo->GetUnfilledFrames();
-           filled_frames + frames_to_push <= max_frames;) {
-        fifo->Push(data.get(), frames_to_push, bytes_per_sample);
-        filled_frames += frames_to_push;
-        EXPECT_EQ(max_frames - filled_frames, fifo->GetUnfilledFrames());
-        EXPECT_EQ(static_cast<int>(filled_frames / block_frames),
-                  fifo->available_blocks());
-      }
+    scoped_ptr<uint8[]> data(new uint8[data_byte_size]);
+    memset(data.get(), 0, data_byte_size);
+
+    for (int filled_frames = max_frames - fifo->GetUnfilledFrames();
+         filled_frames + frames_to_push <= max_frames;) {
+      fifo->Push(data.get(), frames_to_push, bytes_per_sample);
+      filled_frames += frames_to_push;
+      EXPECT_EQ(max_frames - filled_frames, fifo->GetUnfilledFrames());
+      EXPECT_EQ(static_cast<int>(filled_frames / block_frames),
+                fifo->available_blocks());
     }
   }
+
+ private:
+  DISALLOW_COPY_AND_ASSIGN(AudioBlockFifoTest);
 };
 
 // Verify that construction works as intended.
@@ -63,7 +44,7 @@ TEST_F(AudioBlockFifoTest, Construct) {
 }
 
 // Pushes audio bus objects to/from a FIFO up to different degrees.
-TEST_P(AudioBlockFifoFormatTest, Push) {
+TEST_F(AudioBlockFifoTest, Push) {
   const int channels = 2;
   const int frames = 128;
   const int blocks = 2;
@@ -84,7 +65,7 @@ TEST_P(AudioBlockFifoFormatTest, Push) {
 
 // Perform a sequence of Push/Consume calls to different degrees, and verify
 // things are correct.
-TEST_P(AudioBlockFifoFormatTest, PushAndConsume) {
+TEST_F(AudioBlockFifoTest, PushAndConsume) {
   const int channels = 2;
   const int frames = 441;
   const int blocks = 4;
@@ -119,9 +100,10 @@ TEST_P(AudioBlockFifoFormatTest, PushAndConsume) {
   fifo.Clear();
   int new_push_frames = 128;
   // Change the input frame and try to fill up the FIFO.
-  PushAndVerify(&fifo, new_push_frames, channels, frames, frames * blocks);
+  PushAndVerify(&fifo, new_push_frames, channels, frames,
+                frames * blocks);
   EXPECT_TRUE(fifo.GetUnfilledFrames() != 0);
-  EXPECT_TRUE(fifo.available_blocks() == blocks - 1);
+  EXPECT_TRUE(fifo.available_blocks() == blocks -1);
 
   // Consume all the existing filled blocks of data.
   while (fifo.available_blocks()) {
@@ -140,13 +122,14 @@ TEST_P(AudioBlockFifoFormatTest, PushAndConsume) {
 
   // Completely fill up the buffer again.
   new_push_frames = frames * blocks - remain_frames;
-  PushAndVerify(&fifo, new_push_frames, channels, frames, frames * blocks);
+  PushAndVerify(&fifo, new_push_frames, channels, frames,
+                frames * blocks);
   EXPECT_TRUE(fifo.GetUnfilledFrames() == 0);
   EXPECT_TRUE(fifo.available_blocks() == blocks);
 }
 
 // Perform a sequence of Push/Consume calls to a 1 block FIFO.
-TEST_P(AudioBlockFifoFormatTest, PushAndConsumeOneBlockFifo) {
+TEST_F(AudioBlockFifoTest, PushAndConsumeOneBlockFifo) {
   static const int channels = 2;
   static const int frames = 441;
   static const int blocks = 1;
@@ -163,8 +146,4 @@ TEST_P(AudioBlockFifoFormatTest, PushAndConsumeOneBlockFifo) {
   EXPECT_TRUE(fifo.GetUnfilledFrames() == frames);
 }
 
-INSTANTIATE_TEST_CASE_P(AudioBlockFifoTests,
-                        AudioBlockFifoFormatTest,
-                        ::testing::Values(false, true));
-
 }  // namespace media