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Diffstat (limited to 'chromium/media/audio/win/audio_unified_win.h')
| -rw-r--r-- | chromium/media/audio/win/audio_unified_win.h | 352 |
1 files changed, 0 insertions, 352 deletions
diff --git a/chromium/media/audio/win/audio_unified_win.h b/chromium/media/audio/win/audio_unified_win.h deleted file mode 100644 index 76c53297b51..00000000000 --- a/chromium/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_ |