audio_destination.cc | Explore in Territory

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 *     from this software without specific prior written permission.
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#include "third_party/blink/renderer/platform/audio/audio_destination.h"

#include <algorithm>
#include <memory>
#include <utility>

#include "base/feature_list.h"
#include "base/metrics/histogram.h"
#include "base/metrics/histogram_functions.h"
#include "base/numerics/safe_conversions.h"
#include "base/task/single_thread_task_runner.h"
#include "base/threading/thread_restrictions.h"
#include "base/trace_event/trace_event.h"
#include "media/base/audio_bus.h"
#include "media/base/audio_glitch_info.h"
#include "third_party/blink/public/common/features.h"
#include "third_party/blink/public/platform/modules/webrtc/webrtc_logging.h"
#include "third_party/blink/public/platform/platform.h"
#include "third_party/blink/public/platform/web_audio_latency_hint.h"
#include "third_party/blink/public/platform/web_audio_sink_descriptor.h"
#include "third_party/blink/renderer/platform/audio/audio_utilities.h"
#include "third_party/blink/renderer/platform/audio/push_pull_fifo.h"
#include "third_party/blink/renderer/platform/audio/vector_math.h"
#include "third_party/blink/renderer/platform/instrumentation/tracing/trace_event.h"
#include "third_party/blink/renderer/platform/scheduler/public/post_cross_thread_task.h"
#include "third_party/blink/renderer/platform/wtf/cross_thread_copier_base.h"
#include "third_party/blink/renderer/platform/wtf/cross_thread_copier_media.h"
#include "third_party/blink/renderer/platform/wtf/cross_thread_functional.h"

namespace blink {

namespace {

// This FIFO size of 16,384 was chosen based on the UMA data. It's the nearest
// multiple of 128 to 16,354 sample-frames, which represents 100% of the
// histogram from "WebAudio.AudioDestination.HardwareBufferSize".
// Although a buffer this big is atypical, some Android phones with a Bluetooth
// audio device report a large buffer size. This redundancy allows such device
// to play audio via Web Audio API.
constexpr uint32_t kFIFOSize = …;

const char* DeviceStateToString(AudioDestination::DeviceState state) { … }

}  // namespace

scoped_refptr<AudioDestination> AudioDestination::Create(
    AudioIOCallback& callback,
    const WebAudioSinkDescriptor& sink_descriptor,
    unsigned number_of_output_channels,
    const WebAudioLatencyHint& latency_hint,
    std::optional<float> context_sample_rate,
    unsigned render_quantum_frames) { … }

AudioDestination::~AudioDestination() { … }

int AudioDestination::Render(base::TimeDelta delay,
                             base::TimeTicks delay_timestamp,
                             const media::AudioGlitchInfo& glitch_info,
                             media::AudioBus* dest) { … }

void AudioDestination::OnRenderError() { … }

void AudioDestination::Start() { … }

void AudioDestination::Stop() { … }

void AudioDestination::Pause() { … }

void AudioDestination::Resume() { … }

void AudioDestination::SetWorkletTaskRunner(
    scoped_refptr<base::SingleThreadTaskRunner> worklet_task_runner) { … }

void AudioDestination::StartWithWorkletTaskRunner(
    scoped_refptr<base::SingleThreadTaskRunner> worklet_task_runner) { … }

bool AudioDestination::IsPlaying() { … }

double AudioDestination::SampleRate() const { … }

uint32_t AudioDestination::CallbackBufferSize() const { … }

int AudioDestination::FramesPerBuffer() const { … }

base::TimeDelta AudioDestination::GetPlatformBufferDuration() const { … }

uint32_t AudioDestination::MaxChannelCount() const { … }

void AudioDestination::SetDetectSilence(bool detect_silence) { … }

AudioDestination::AudioDestination(
    AudioIOCallback& callback,
    const WebAudioSinkDescriptor& sink_descriptor,
    unsigned number_of_output_channels,
    const WebAudioLatencyHint& latency_hint,
    std::optional<float> context_sample_rate,
    unsigned render_quantum_frames)
    : … { … }

void AudioDestination::SetDeviceState(DeviceState state) { … }

void AudioDestination::RequestRenderWait(
    size_t frames_requested,
    size_t frames_to_render,
    base::TimeDelta delay,
    base::TimeTicks delay_timestamp,
    const media::AudioGlitchInfo& glitch_info) { … }

void AudioDestination::RequestRender(
    size_t frames_requested,
    size_t frames_to_render,
    base::TimeDelta delay,
    base::TimeTicks delay_timestamp,
    const media::AudioGlitchInfo& glitch_info) { … }

void AudioDestination::ProvideResamplerInput(int resampler_frame_delay,
                                             AudioBus* dest) { … }

void AudioDestination::SendLogMessage(const String& message) const { … }

void AudioDestination::PullFromCallback(AudioBus* destination_bus,
                                        base::TimeDelta delay) { … }

media::OutputDeviceStatus AudioDestination::MaybeCreateSinkAndGetStatus() { … }

}  // namespace blink
chromium/third_party/blink/renderer/platform/audio/audio_destination.cc
