Reland 153623004: Remove the unified IO code on the browser

media/audio/win/audio_unified_win.h

Index: media/audio/win/audio_unified_win.h
diff --git a/media/audio/win/audio_unified_win.h b/media/audio/win/audio_unified_win.h
deleted file mode 100644
index 76c53297b51a6766bd7f69a5c5a213be941c68ae..0000000000000000000000000000000000000000
--- a/media/audio/win/audio_unified_win.h
+++ /dev/null
@@ -1,352 +0,0 @@
-// Copyright (c) 2012 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.
-
-#ifndef MEDIA_AUDIO_WIN_AUDIO_UNIFIED_WIN_H_
-#define MEDIA_AUDIO_WIN_AUDIO_UNIFIED_WIN_H_
-
-#include <Audioclient.h>
-#include <MMDeviceAPI.h>
-
-#include <string>
-
-#include "base/compiler_specific.h"
-#include "base/gtest_prod_util.h"
-#include "base/threading/platform_thread.h"
-#include "base/threading/simple_thread.h"
-#include "base/win/scoped_co_mem.h"
-#include "base/win/scoped_comptr.h"
-#include "base/win/scoped_handle.h"
-#include "media/audio/audio_io.h"
-#include "media/audio/audio_parameters.h"
-#include "media/base/audio_fifo.h"
-#include "media/base/channel_mixer.h"
-#include "media/base/media_export.h"
-#include "media/base/multi_channel_resampler.h"
-
-namespace media {
-
-class AudioManagerWin;
-
-// Implementation of AudioOutputStream for Windows using the Core Audio API
-// where both capturing and rendering takes place on the same thread to enable
-// audio I/O. This class allows arbitrary combinations of input and output
-// devices running off different clocks and using different drivers, with
-// potentially differing sample-rates.
-//
-// It is required to first acquire the native sample rate of the selected
-// output device and then use the same rate when creating this object.
-// The inner operation depends on the input sample rate which is determined
-// during construction. Three different main modes are supported:
-//
-//  1)  input rate == output rate => input side drives output side directly.
-//  2)  input rate != output rate => both sides are driven independently by
-//      events and a FIFO plus a resampling unit is used to compensate for
-//      differences in sample rates between the two sides.
-//  3)  input rate == output rate but native buffer sizes are not identical =>
-//      same inner functionality as in (2) to compensate for the differences
-//      in buffer sizes and also compensate for any potential clock drift
-//      between the two devices.
-//
-// Mode detection is is done at construction and using mode (1) will lead to
-// best performance (lower delay and no "varispeed distortion"), i.e., it is
-// recommended to use same sample rates for input and output. Mode (2) uses a
-// resampler which supports rate adjustments to fine tune for things like
-// clock drift and differences in sample rates between different devices.
-// Mode (2) - which uses a FIFO and a adjustable multi-channel resampler -
-// is also called the varispeed mode and it is used for case (3) as well to
-// compensate for the difference in buffer sizes mainly.
-// Mode (3) can happen if two different audio devices are used.
-// As an example: some devices needs a buffer size of 441 @ 44.1kHz and others
-// 448 @ 44.1kHz. This is a rare case and will only happen for sample rates
-// which are even multiples of 11025 Hz (11025, 22050, 44100, 88200 etc.).
-//
-// Implementation notes:
-//
-//  - Open() can fail if the input and output parameters do not fulfill
-//    certain conditions. See source for Open() for more details.
-//  - Channel mixing will be performed if the clients asks for a larger
-//    number of channels than the native audio layer provides.
-//    Example: client wants stereo but audio layer provides mono. In this case
-//    upmixing from mono to stereo (1->2) will be done.
-//
-// TODO(henrika):
-//
-//  - Add support for exclusive mode.
-//  - Add support for KSDATAFORMAT_SUBTYPE_IEEE_FLOAT, i.e., 32-bit float
-//    as internal sample-value representation.
-//  - Perform fine-tuning for non-matching sample rates to reduce latency.
-//
-class MEDIA_EXPORT WASAPIUnifiedStream
-    : public AudioOutputStream,
-      public base::DelegateSimpleThread::Delegate {
- public:
-  // The ctor takes all the usual parameters, plus |manager| which is the
-  // the audio manager who is creating this object.
-  WASAPIUnifiedStream(AudioManagerWin* manager,
-                      const AudioParameters& params,
-                      const std::string& input_device_id);
-
-  // The dtor is typically called by the AudioManager only and it is usually
-  // triggered by calling AudioOutputStream::Close().
-  virtual ~WASAPIUnifiedStream();
-
-  // Implementation of AudioOutputStream.
-  virtual bool Open() OVERRIDE;
-  virtual void Start(AudioSourceCallback* callback) OVERRIDE;
-  virtual void Stop() OVERRIDE;
-  virtual void Close() OVERRIDE;
-  virtual void SetVolume(double volume) OVERRIDE;
-  virtual void GetVolume(double* volume) OVERRIDE;
-
-  bool started() const {
-    return audio_io_thread_.get() != NULL;
-  }
-
-  // Returns true if input sample rate differs from the output sample rate.
-  // A FIFO and a adjustable multi-channel resampler are utilized in this mode.
-  bool VarispeedMode() const { return (fifo_ && resampler_); }
-
- private:
-  enum {
-    // Time in milliseconds between two successive delay measurements.
-    // We save resources by not updating the delay estimates for each capture
-    // event (typically 100Hz rate).
-    kTimeDiffInMillisecondsBetweenDelayMeasurements = 1000,
-
-    // Max possible FIFO size.
-    kFifoSize = 16384,
-
-    // This value was determined empirically for minimum latency while still
-    // guarding against FIFO under-runs. The actual target size will be equal
-    // to kTargetFifoSafetyFactor * (native input buffer size).
-    // TODO(henrika): tune this value for lowest possible latency for all
-    // possible sample rate combinations.
-    kTargetFifoSafetyFactor = 2
-  };
-
-  // Additional initialization required when input and output sample rate
-  // differs. Allocates resources for |fifo_|, |resampler_|, |render_event_|,
-  // and the |capture_bus_| and configures the |input_format_| structure
-  // given the provided input and output audio parameters.
-  void DoVarispeedInitialization(const AudioParameters& input_params,
-                                 const AudioParameters& output_params);
-
-  // Clears varispeed related components such as the FIFO and the resampler.
-  void ResetVarispeed();
-
-  // Builds WAVEFORMATEX structures for input and output based on input and
-  // output audio parameters.
-  void SetIOFormats(const AudioParameters& input_params,
-                    const AudioParameters& output_params);
-
-  // DelegateSimpleThread::Delegate implementation.
-  virtual void Run() OVERRIDE;
-
-  // MultiChannelResampler::MultiChannelAudioSourceProvider implementation.
-  // Callback for providing more data into the resampler.
-  // Only used in varispeed mode, i.e., when input rate != output rate.
-  virtual void ProvideInput(int frame_delay, AudioBus* audio_bus);
-
-  // Issues the OnError() callback to the |source_|.
-  void HandleError(HRESULT err);
-
-  // Stops and joins the audio thread in case of an error.
-  void StopAndJoinThread(HRESULT err);
-
-  // Converts unique endpoint ID to user-friendly device name.
-  std::string GetDeviceName(LPCWSTR device_id) const;
-
-  // Called on the audio IO thread for each capture event.
-  // Buffers captured audio into a FIFO if varispeed is used or into an audio
-  // bus if input and output sample rates are identical.
-  void ProcessInputAudio();
-
-  // Called on the audio IO thread for each render event when varispeed is
-  // active or for each capture event when varispeed is not used.
-  // In varispeed mode, it triggers a resampling callback, which reads from the
-  // FIFO, and calls AudioSourceCallback::OnMoreIOData using the resampled
-  // input signal and at the same time asks for data to play out.
-  // If input and output rates are the same - instead of reading from the FIFO
-  // and do resampling - we read directly from the audio bus used to store
-  // captured data in ProcessInputAudio.
-  void ProcessOutputAudio(IAudioClock* audio_output_clock);
-
-  // Contains the thread ID of the creating thread.
-  base::PlatformThreadId creating_thread_id_;
-
-  // Our creator, the audio manager needs to be notified when we close.
-  AudioManagerWin* manager_;
-
-  // Contains the audio parameter structure provided at construction.
-  AudioParameters params_;
-  // For convenience, same as in params_.
-  int input_channels_;
-  int output_channels_;
-
-  // Unique ID of the input device to be opened.
-  const std::string input_device_id_;
-
-  // The sharing mode for the streams.
-  // Valid values are AUDCLNT_SHAREMODE_SHARED and AUDCLNT_SHAREMODE_EXCLUSIVE
-  // where AUDCLNT_SHAREMODE_SHARED is the default.
-  AUDCLNT_SHAREMODE share_mode_;
-
-  // Rendering and capturing is driven by this thread (no message loop).
-  // All OnMoreIOData() callbacks will be called from this thread.
-  scoped_ptr<base::DelegateSimpleThread> audio_io_thread_;
-
-  // Contains the desired audio output format which is set up at construction.
-  // It is required to first acquire the native sample rate of the selected
-  // output device and then use the same rate when creating this object.
-  WAVEFORMATPCMEX output_format_;
-
-  // Contains the native audio input format which is set up at construction
-  // if varispeed mode is utilized.
-  WAVEFORMATPCMEX input_format_;
-
-  // True when successfully opened.
-  bool opened_;
-
-  // Volume level from 0 to 1 used for output scaling.
-  double volume_;
-
-  // Size in audio frames of each audio packet where an audio packet
-  // is defined as the block of data which the destination is expected to
-  // receive in each OnMoreIOData() callback.
-  size_t output_buffer_size_frames_;
-
-  // Size in audio frames of each audio packet where an audio packet
-  // is defined as the block of data which the source is expected to
-  // deliver in each OnMoreIOData() callback.
-  size_t input_buffer_size_frames_;
-
-  // Length of the audio endpoint buffer.
-  uint32 endpoint_render_buffer_size_frames_;
-  uint32 endpoint_capture_buffer_size_frames_;
-
-  // Counts the number of audio frames written to the endpoint buffer.
-  uint64 num_written_frames_;
-
-  // Time stamp for last delay measurement.
-  base::TimeTicks last_delay_sample_time_;
-
-  // Contains the total (sum of render and capture) delay in milliseconds.
-  double total_delay_ms_;
-
-  // Contains the total (sum of render and capture and possibly FIFO) delay
-  // in bytes. The update frequency is set by a constant called
-  // |kTimeDiffInMillisecondsBetweenDelayMeasurements|.
-  int total_delay_bytes_;
-
-  // Pointer to the client that will deliver audio samples to be played out.
-  AudioSourceCallback* source_;
-
-  // IMMDevice interfaces which represents audio endpoint devices.
-  base::win::ScopedComPtr<IMMDevice> endpoint_render_device_;
-  base::win::ScopedComPtr<IMMDevice> endpoint_capture_device_;
-
-  // IAudioClient interfaces which enables a client to create and initialize
-  // an audio stream between an audio application and the audio engine.
-  base::win::ScopedComPtr<IAudioClient> audio_output_client_;
-  base::win::ScopedComPtr<IAudioClient> audio_input_client_;
-
-  // IAudioRenderClient interfaces enables a client to write output
-  // data to a rendering endpoint buffer.
-  base::win::ScopedComPtr<IAudioRenderClient> audio_render_client_;
-
-  // IAudioCaptureClient interfaces enables a client to read input
-  // data from a capturing endpoint buffer.
-  base::win::ScopedComPtr<IAudioCaptureClient> audio_capture_client_;
-
-  // The audio engine will signal this event each time a buffer has been
-  // recorded.
-  base::win::ScopedHandle capture_event_;
-
-  // The audio engine will signal this event each time it needs a new
-  // audio buffer to play out.
-  // Only utilized in varispeed mode.
-  base::win::ScopedHandle render_event_;
-
-  // This event will be signaled when streaming shall stop.
-  base::win::ScopedHandle stop_streaming_event_;
-
-  // Container for retrieving data from AudioSourceCallback::OnMoreIOData().
-  scoped_ptr<AudioBus> output_bus_;
-
-  // Container for sending data to AudioSourceCallback::OnMoreIOData().
-  scoped_ptr<AudioBus> input_bus_;
-
-  // Container for storing output from the channel mixer.
-  scoped_ptr<AudioBus> channel_bus_;
-
-  // All members below are only allocated, or used, in varispeed mode:
-
-  // Temporary storage of resampled input audio data.
-  scoped_ptr<AudioBus> resampled_bus_;
-
-  // Set to true first time a capture event has been received in varispeed
-  // mode.
-  bool input_callback_received_;
-
-  // MultiChannelResampler is a multi channel wrapper for SincResampler;
-  // allowing high quality sample rate conversion of multiple channels at once.
-  scoped_ptr<MultiChannelResampler> resampler_;
-
-  // Resampler I/O ratio.
-  double io_sample_rate_ratio_;
-
-  // Used for input to output buffering.
-  scoped_ptr<AudioFifo> fifo_;
-
-  // The channel mixer is only created and utilized if number of input channels
-  // is larger than the native number of input channels (e.g client wants
-  // stereo but the audio device only supports mono).
-  scoped_ptr<ChannelMixer> channel_mixer_;
-
-  // The optimal number of frames we'd like to keep in the FIFO at all times.
-  int target_fifo_frames_;
-
-  // A running average of the measured delta between actual number of frames
-  // in the FIFO versus |target_fifo_frames_|.
-  double average_delta_;
-
-  // A varispeed rate scalar which is calculated based on FIFO drift.
-  double fifo_rate_compensation_;
-
-  // Set to true when input side signals output side that a new delay
-  // estimate is needed.
-  bool update_output_delay_;
-
-  // Capture side stores its delay estimate so the sum can be derived in
-  // the render side.
-  double capture_delay_ms_;
-
-  // TODO(henrika): possibly remove these members once the performance is
-  // properly tuned. Only used for off-line debugging.
-#ifndef NDEBUG
-  enum LogElementNames {
-    INPUT_TIME_STAMP,
-    NUM_FRAMES_IN_FIFO,
-    RESAMPLER_MARGIN,
-    RATE_COMPENSATION
-  };
-
-  scoped_ptr<int64[]> input_time_stamps_;
-  scoped_ptr<int[]> num_frames_in_fifo_;
-  scoped_ptr<int[]> resampler_margin_;
-  scoped_ptr<double[]> fifo_rate_comps_;
-  scoped_ptr<int[]> num_elements_;
-  scoped_ptr<int[]> input_params_;
-  scoped_ptr<int[]> output_params_;
-
-  FILE* data_file_;
-  FILE* param_file_;
-#endif
-
-  DISALLOW_COPY_AND_ASSIGN(WASAPIUnifiedStream);
-};
-
-}  // namespace media
-
-#endif  // MEDIA_AUDIO_WIN_AUDIO_UNIFIED_WIN_H_