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/*
* Copyright (C) 2010 Google Inc. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
* 3. Neither the name of Apple Computer, Inc. ("Apple") nor the names of
* its contributors may be used to endorse or promote products derived
* from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
* EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
* WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
* DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
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* (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
* LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
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* THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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#ifndef THIRD_PARTY_BLINK_RENDERER_PLATFORM_AUDIO_AUDIO_DESTINATION_H_
#define THIRD_PARTY_BLINK_RENDERER_PLATFORM_AUDIO_AUDIO_DESTINATION_H_
#include <memory>
#include "base/memory/scoped_refptr.h"
#include "base/single_thread_task_runner.h"
#include "third_party/blink/public/platform/web_audio_device.h"
#include "third_party/blink/public/platform/web_vector.h"
#include "third_party/blink/renderer/platform/audio/audio_bus.h"
#include "third_party/blink/renderer/platform/audio/audio_io_callback.h"
#include "third_party/blink/renderer/platform/audio/media_multi_channel_resampler.h"
#include "third_party/blink/renderer/platform/scheduler/public/thread.h"
#include "third_party/blink/renderer/platform/wtf/text/wtf_string.h"
#include "third_party/blink/renderer/platform/wtf/thread_safe_ref_counted.h"
namespace blink {
class PushPullFIFO;
class WebAudioLatencyHint;
// The AudioDestination class is an audio sink interface between the media
// renderer and the Blink's WebAudio module. It has a FIFO to adapt the
// different processing block sizes of WebAudio renderer and actual hardware
// audio callback.
//
// Currently AudioDestination supports two types of threading models:
// - Single-thread (default): process the entire WebAudio render call chain by
// AudioDeviceThread.
// - Dual-thread (experimental): Use WebThread for the WebAudio rendering with
// AudioWorkletThread.
class PLATFORM_EXPORT AudioDestination
: public ThreadSafeRefCounted<AudioDestination>,
public WebAudioDevice::RenderCallback {
USING_FAST_MALLOC(AudioDestination);
public:
AudioDestination(AudioIOCallback&,
unsigned number_of_output_channels,
const WebAudioLatencyHint&,
base::Optional<float> context_sample_rate);
~AudioDestination() override;
static scoped_refptr<AudioDestination> Create(
AudioIOCallback&,
unsigned number_of_output_channels,
const WebAudioLatencyHint&,
base::Optional<float> context_sample_rate);
// The actual render function (WebAudioDevice::RenderCallback) isochronously
// invoked by the media renderer. This is never called after Stop() is called.
void Render(const WebVector<float*>& destination_data,
size_t number_of_frames,
double delay,
double delay_timestamp,
size_t prior_frames_skipped) override;
// The actual render request to the WebAudio destination node. This method
// can be invoked on both AudioDeviceThread (single-thread rendering) and
// AudioWorkletThread (dual-thread rendering).
void RequestRender(size_t frames_requested,
size_t frames_to_render,
double delay,
double delay_timestamp,
size_t prior_frames_skipped);
virtual void Start();
virtual void Stop();
virtual void Pause();
virtual void Resume();
// Starts the destination with the AudioWorklet support.
void StartWithWorkletTaskRunner(
scoped_refptr<base::SingleThreadTaskRunner> worklet_task_runner);
// Getters must be accessed from the main thread.
uint32_t CallbackBufferSize() const;
bool IsPlaying();
// This is the context sample rate, not the device one.
double SampleRate() const { return context_sample_rate_; }
// Returns the audio buffer size in frames used by the underlying audio
// hardware.
int FramesPerBuffer() const;
// The information from the actual audio hardware. (via Platform::current)
static float HardwareSampleRate();
static uint32_t MaxChannelCount();
private:
// Provide input to the resampler (if used).
void ProvideResamplerInput(int resampler_frame_delay, AudioBus* dest);
enum class PlayState { kStopped, kPlaying, kPaused };
// Check if the buffer size chosen by the WebAudioDevice is too large.
bool CheckBufferSize();
size_t HardwareBufferSize();
// Accessed by the main thread.
std::unique_ptr<WebAudioDevice> web_audio_device_;
const unsigned number_of_output_channels_;
uint32_t callback_buffer_size_;
PlayState play_state_;
// The task runner for AudioWorklet operation. This is only valid when
// the AudioWorklet is activated.
scoped_refptr<base::SingleThreadTaskRunner> worklet_task_runner_;
// Can be accessed by both threads: resolves the buffer size mismatch between
// the WebAudio engine and the callback function from the actual audio device.
std::unique_ptr<PushPullFIFO> fifo_;
// Accessed by device thread: to pass the data from FIFO to the device.
scoped_refptr<AudioBus> output_bus_;
// Accessed by rendering thread: to push the rendered result from WebAudio
// graph into the FIFO.
scoped_refptr<AudioBus> render_bus_;
// Accessed by rendering thread: the render callback function of WebAudio
// engine. (i.e. DestinationNode)
AudioIOCallback& callback_;
// When the last callback function from the device is called.
base::TimeTicks previous_callback_request_;
// The time duration spent on rendering previous render quanta per callback.
base::TimeDelta previous_render_duration_;
// Accessed by rendering thread.
size_t frames_elapsed_;
// The sample rate used for rendering the Web Audio graph.
float context_sample_rate_;
// Used for resampling if the Web Audio sample rate differs from the platform
// one.
std::unique_ptr<MediaMultiChannelResampler> resampler_;
std::unique_ptr<media::AudioBus> resampler_bus_;
// Required for RequestRender and also in the resampling callback (if used).
AudioIOPosition output_position_;
AudioIOCallbackMetric metric_;
DISALLOW_COPY_AND_ASSIGN(AudioDestination);
};
} // namespace blink
#endif // THIRD_PARTY_BLINK_RENDERER_PLATFORM_AUDIO_AUDIO_DESTINATION_H_