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/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
/* vim:set ts=2 sw=2 sts=2 et cindent: */
/* This Source Code Form is subject to the terms of the Mozilla Public
* License, v. 2.0. If a copy of the MPL was not distributed with this
#include <stdio.h>
#include <math.h>
#include <string.h>
#include "mozilla/Logging.h"
#include "prdtoa.h"
#include "AudioStream.h"
#include "VideoUtils.h"
#include "mozilla/dom/AudioDeviceInfo.h"
#include "mozilla/Monitor.h"
#include "mozilla/Mutex.h"
#include "mozilla/Sprintf.h"
#include "mozilla/Unused.h"
#include <algorithm>
#include "mozilla/Telemetry.h"
#include "CubebUtils.h"
#include "nsNativeCharsetUtils.h"
#include "nsPrintfCString.h"
#include "AudioConverter.h"
#include "UnderrunHandler.h"
#if defined(XP_WIN)
# include "nsXULAppAPI.h"
#endif
#include "Tracing.h"
#include "webaudio/blink/DenormalDisabler.h"
#include "CallbackThreadRegistry.h"
#include "mozilla/StaticPrefs_media.h"
#include "RLBoxSoundTouch.h"
namespace mozilla {
#undef LOG
#undef LOGW
#undef LOGE
LazyLogModule gAudioStreamLog("AudioStream");
// For simple logs
#define LOG(x, ...) \
MOZ_LOG(gAudioStreamLog, mozilla::LogLevel::Debug, \
("%p " x, this, ##__VA_ARGS__))
#define LOGW(x, ...) \
MOZ_LOG(gAudioStreamLog, mozilla::LogLevel::Warning, \
("%p " x, this, ##__VA_ARGS__))
#define LOGE(x, ...) \
NS_DebugBreak(NS_DEBUG_WARNING, \
nsPrintfCString("%p " x, this, ##__VA_ARGS__).get(), nullptr, \
__FILE__, __LINE__)
/**
* Keep a list of frames sent to the audio engine in each DataCallback along
* with the playback rate at the moment. Since the playback rate and number of
* underrun frames can vary in each callback. We need to keep the whole history
* in order to calculate the playback position of the audio engine correctly.
*/
class FrameHistory {
struct Chunk {
uint32_t servicedFrames;
uint32_t totalFrames;
uint32_t rate;
};
template <typename T>
static T FramesToUs(uint32_t frames, uint32_t rate) {
return static_cast<T>(frames) * USECS_PER_S / rate;
}
public:
FrameHistory() : mBaseOffset(0), mBasePosition(0) {}
void Append(uint32_t aServiced, uint32_t aUnderrun, uint32_t aRate) {
/* In most case where playback rate stays the same and we don't underrun
* frames, we are able to merge chunks to avoid lose of precision to add up
* in compressing chunks into |mBaseOffset| and |mBasePosition|.
*/
if (!mChunks.IsEmpty()) {
Chunk& c = mChunks.LastElement();
// 2 chunks (c1 and c2) can be merged when rate is the same and
// adjacent frames are zero. That is, underrun frames in c1 are zero
// or serviced frames in c2 are zero.
if (c.rate == aRate &&
(c.servicedFrames == c.totalFrames || aServiced == 0)) {
c.servicedFrames += aServiced;
c.totalFrames += aServiced + aUnderrun;
return;
}
}
Chunk* p = mChunks.AppendElement();
p->servicedFrames = aServiced;
p->totalFrames = aServiced + aUnderrun;
p->rate = aRate;
}
/**
* @param frames The playback position in frames of the audio engine.
* @return The playback position in microseconds of the audio engine,
* adjusted by playback rate changes and underrun frames.
*/
int64_t GetPosition(int64_t frames) {
// playback position should not go backward.
MOZ_ASSERT(frames >= mBaseOffset);
while (true) {
if (mChunks.IsEmpty()) {
return static_cast<int64_t>(mBasePosition);
}
const Chunk& c = mChunks[0];
if (frames <= mBaseOffset + c.totalFrames) {
uint32_t delta = frames - mBaseOffset;
delta = std::min(delta, c.servicedFrames);
return static_cast<int64_t>(mBasePosition) +
FramesToUs<int64_t>(delta, c.rate);
}
// Since the playback position of the audio engine will not go backward,
// we are able to compress chunks so that |mChunks| won't grow
// unlimitedly. Note that we lose precision in converting integers into
// floats and inaccuracy will accumulate over time. However, for a 24hr
// long, sample rate = 44.1k file, the error will be less than 1
// microsecond after playing 24 hours. So we are fine with that.
mBaseOffset += c.totalFrames;
mBasePosition += FramesToUs<double>(c.servicedFrames, c.rate);
mChunks.RemoveElementAt(0);
}
}
private:
AutoTArray<Chunk, 7> mChunks;
int64_t mBaseOffset;
double mBasePosition;
};
AudioStream::AudioStream(DataSource& aSource, uint32_t aInRate,
uint32_t aOutputChannels,
AudioConfig::ChannelLayout::ChannelMap aChannelMap)
: mTimeStretcher(nullptr),
mAudioClock(aInRate),
mChannelMap(aChannelMap),
mMonitor("AudioStream"),
mOutChannels(aOutputChannels),
mState(INITIALIZED),
mDataSource(aSource),
mAudioThreadId(ProfilerThreadId{}),
mSandboxed(CubebUtils::SandboxEnabled()),
mPlaybackComplete(false),
mPlaybackRate(1.0f),
mPreservesPitch(true),
mCallbacksStarted(false) {}
AudioStream::~AudioStream() {
LOG("deleted, state %d", mState.load());
MOZ_ASSERT(mState == SHUTDOWN && !mCubebStream,
"Should've called ShutDown() before deleting an AudioStream");
}
size_t AudioStream::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const {
size_t amount = aMallocSizeOf(this);
// Possibly add in the future:
// - mTimeStretcher
// - mCubebStream
return amount;
}
nsresult AudioStream::EnsureTimeStretcherInitialized() {
AssertIsOnAudioThread();
if (!mTimeStretcher) {
mTimeStretcher = new RLBoxSoundTouch();
mTimeStretcher->setSampleRate(mAudioClock.GetInputRate());
mTimeStretcher->setChannels(mOutChannels);
mTimeStretcher->setPitch(1.0);
// SoundTouch v2.1.2 uses automatic time-stretch settings with the following
// values:
// Tempo 0.5: 90ms sequence, 20ms seekwindow, 8ms overlap
// Tempo 2.0: 40ms sequence, 15ms seekwindow, 8ms overlap
// We are going to use a smaller 10ms sequence size to improve speech
// clarity, giving more resolution at high tempo and less reverb at low
// tempo. Maintain 15ms seekwindow and 8ms overlap for smoothness.
mTimeStretcher->setSetting(
SETTING_SEQUENCE_MS,
StaticPrefs::media_audio_playbackrate_soundtouch_sequence_ms());
mTimeStretcher->setSetting(
SETTING_SEEKWINDOW_MS,
StaticPrefs::media_audio_playbackrate_soundtouch_seekwindow_ms());
mTimeStretcher->setSetting(
SETTING_OVERLAP_MS,
StaticPrefs::media_audio_playbackrate_soundtouch_overlap_ms());
}
return NS_OK;
}
nsresult AudioStream::SetPlaybackRate(double aPlaybackRate) {
TRACE_COMMENT("AudioStream::SetPlaybackRate", "%f", aPlaybackRate);
NS_ASSERTION(
aPlaybackRate > 0.0,
"Can't handle negative or null playbackrate in the AudioStream.");
if (aPlaybackRate == mPlaybackRate) {
return NS_OK;
}
mPlaybackRate = static_cast<float>(aPlaybackRate);
return NS_OK;
}
nsresult AudioStream::SetPreservesPitch(bool aPreservesPitch) {
TRACE_COMMENT("AudioStream::SetPreservesPitch", "%d", aPreservesPitch);
if (aPreservesPitch == mPreservesPitch) {
return NS_OK;
}
mPreservesPitch = aPreservesPitch;
return NS_OK;
}
template <typename Function, typename... Args>
int AudioStream::InvokeCubeb(Function aFunction, Args&&... aArgs) {
mMonitor.AssertCurrentThreadOwns();
MonitorAutoUnlock mon(mMonitor);
return aFunction(mCubebStream.get(), std::forward<Args>(aArgs)...);
}
nsresult AudioStream::Init(AudioDeviceInfo* aSinkInfo)
MOZ_NO_THREAD_SAFETY_ANALYSIS {
auto startTime = TimeStamp::Now();
TRACE("AudioStream::Init");
LOG("%s channels: %d, rate: %d", __FUNCTION__, mOutChannels,
mAudioClock.GetInputRate());
mSinkInfo = aSinkInfo;
cubeb_stream_params params;
params.rate = mAudioClock.GetInputRate();
params.channels = mOutChannels;
params.layout = static_cast<uint32_t>(mChannelMap);
params.format = CubebUtils::ToCubebFormat<AUDIO_OUTPUT_FORMAT>::value;
params.prefs = CubebUtils::GetDefaultStreamPrefs(CUBEB_DEVICE_TYPE_OUTPUT);
// This is noop if MOZ_DUMP_AUDIO is not set.
mDumpFile.Open("AudioStream", mOutChannels, mAudioClock.GetInputRate());
RefPtr<CubebUtils::CubebHandle> handle = CubebUtils::GetCubeb();
if (!handle) {
LOGE("Can't get cubeb context!");
CubebUtils::ReportCubebStreamInitFailure(true);
return NS_ERROR_DOM_MEDIA_CUBEB_INITIALIZATION_ERR;
}
mCubeb = handle;
return OpenCubeb(handle->Context(), params, startTime,
CubebUtils::GetFirstStream());
}
nsresult AudioStream::OpenCubeb(cubeb* aContext, cubeb_stream_params& aParams,
TimeStamp aStartTime, bool aIsFirst) {
TRACE("AudioStream::OpenCubeb");
MOZ_ASSERT(aContext);
cubeb_stream* stream = nullptr;
/* Convert from milliseconds to frames. */
uint32_t latency_frames =
CubebUtils::GetCubebPlaybackLatencyInMilliseconds() * aParams.rate / 1000;
cubeb_devid deviceID = nullptr;
if (mSinkInfo && mSinkInfo->DeviceID()) {
deviceID = mSinkInfo->DeviceID();
}
if (CubebUtils::CubebStreamInit(aContext, &stream, "AudioStream", nullptr,
nullptr, deviceID, &aParams, latency_frames,
DataCallback_S, StateCallback_S,
this) == CUBEB_OK) {
mCubebStream.reset(stream);
CubebUtils::ReportCubebBackendUsed();
} else {
LOGE("OpenCubeb() failed to init cubeb");
CubebUtils::ReportCubebStreamInitFailure(aIsFirst);
return NS_ERROR_FAILURE;
}
TimeDuration timeDelta = TimeStamp::Now() - aStartTime;
LOG("creation time %sfirst: %u ms", aIsFirst ? "" : "not ",
(uint32_t)timeDelta.ToMilliseconds());
return NS_OK;
}
void AudioStream::SetVolume(double aVolume) {
TRACE_COMMENT("AudioStream::SetVolume", "%f", aVolume);
MOZ_ASSERT(aVolume >= 0.0 && aVolume <= 1.0, "Invalid volume");
MOZ_ASSERT(mState != SHUTDOWN, "Don't set volume after shutdown.");
if (mState == ERRORED) {
return;
}
MonitorAutoLock mon(mMonitor);
if (InvokeCubeb(cubeb_stream_set_volume,
aVolume * CubebUtils::GetVolumeScale()) != CUBEB_OK) {
LOGE("Could not change volume on cubeb stream.");
}
}
void AudioStream::SetStreamName(const nsAString& aStreamName) {
TRACE("AudioStream::SetStreamName");
nsAutoCString aRawStreamName;
nsresult rv = NS_CopyUnicodeToNative(aStreamName, aRawStreamName);
if (NS_FAILED(rv) || aStreamName.IsEmpty()) {
return;
}
MonitorAutoLock mon(mMonitor);
int r = InvokeCubeb(cubeb_stream_set_name, aRawStreamName.get());
if (r && r != CUBEB_ERROR_NOT_SUPPORTED) {
LOGE("Could not set cubeb stream name.");
}
}
RefPtr<MediaSink::EndedPromise> AudioStream::Start() {
TRACE("AudioStream::Start");
MOZ_ASSERT(mState == INITIALIZED);
mState = STARTED;
RefPtr<MediaSink::EndedPromise> promise;
{
MonitorAutoLock mon(mMonitor);
// As cubeb might call audio stream's state callback very soon after we
// start cubeb, we have to create the promise beforehand in order to handle
// the case where we immediately get `drained`.
promise = mEndedPromise.Ensure(__func__);
mPlaybackComplete = false;
if (InvokeCubeb(cubeb_stream_start) != CUBEB_OK) {
mState = ERRORED;
mEndedPromise.RejectIfExists(NS_ERROR_FAILURE, __func__);
}
LOG("started, state %s", mState == STARTED ? "STARTED"
: mState == DRAINED ? "DRAINED"
: "ERRORED");
}
return promise;
}
void AudioStream::Pause() {
TRACE("AudioStream::Pause");
MOZ_ASSERT(mState != INITIALIZED, "Must be Start()ed.");
MOZ_ASSERT(mState != STOPPED, "Already Pause()ed.");
MOZ_ASSERT(mState != SHUTDOWN, "Already ShutDown()ed.");
// Do nothing if we are already drained or errored.
if (mState == DRAINED || mState == ERRORED) {
return;
}
MonitorAutoLock mon(mMonitor);
if (InvokeCubeb(cubeb_stream_stop) != CUBEB_OK) {
mState = ERRORED;
} else if (mState != DRAINED && mState != ERRORED) {
// Don't transition to other states if we are already
// drained or errored.
mState = STOPPED;
}
}
void AudioStream::Resume() {
TRACE("AudioStream::Resume");
MOZ_ASSERT(mState != INITIALIZED, "Must be Start()ed.");
MOZ_ASSERT(mState != STARTED, "Already Start()ed.");
MOZ_ASSERT(mState != SHUTDOWN, "Already ShutDown()ed.");
// Do nothing if we are already drained or errored.
if (mState == DRAINED || mState == ERRORED) {
return;
}
MonitorAutoLock mon(mMonitor);
if (InvokeCubeb(cubeb_stream_start) != CUBEB_OK) {
mState = ERRORED;
} else if (mState != DRAINED && mState != ERRORED) {
// Don't transition to other states if we are already
// drained or errored.
mState = STARTED;
}
}
void AudioStream::ShutDown() {
TRACE("AudioStream::ShutDown");
LOG("ShutDown, state %d", mState.load());
MonitorAutoLock mon(mMonitor);
if (mCubebStream) {
// Force stop to put the cubeb stream in a stable state before deletion.
InvokeCubeb(cubeb_stream_stop);
// Must not try to shut down cubeb from within the lock! wasapi may still
cubeb_stream* cubeb = mCubebStream.release();
MonitorAutoUnlock unlock(mMonitor);
cubeb_stream_destroy(cubeb);
}
// After `cubeb_stream_stop` has been called, there is no audio thread
// anymore. We can delete the time stretcher.
if (mTimeStretcher) {
delete mTimeStretcher;
mTimeStretcher = nullptr;
}
mState = SHUTDOWN;
mEndedPromise.ResolveIfExists(true, __func__);
}
int64_t AudioStream::GetPosition() {
TRACE("AudioStream::GetPosition");
#ifndef XP_MACOSX
MonitorAutoLock mon(mMonitor);
#endif
int64_t frames = GetPositionInFramesUnlocked();
return frames >= 0 ? mAudioClock.GetPosition(frames) : -1;
}
int64_t AudioStream::GetPositionInFrames() {
TRACE("AudioStream::GetPositionInFrames");
#ifndef XP_MACOSX
MonitorAutoLock mon(mMonitor);
#endif
int64_t frames = GetPositionInFramesUnlocked();
return frames >= 0 ? mAudioClock.GetPositionInFrames(frames) : -1;
}
int64_t AudioStream::GetPositionInFramesUnlocked() {
TRACE("AudioStream::GetPositionInFramesUnlocked");
#ifndef XP_MACOSX
mMonitor.AssertCurrentThreadOwns();
#endif
if (mState == ERRORED) {
return -1;
}
uint64_t position = 0;
int rv;
#ifndef XP_MACOSX
rv = InvokeCubeb(cubeb_stream_get_position, &position);
#else
rv = cubeb_stream_get_position(mCubebStream.get(), &position);
#endif
if (rv != CUBEB_OK) {
return -1;
}
return static_cast<int64_t>(std::min<uint64_t>(position, INT64_MAX));
}
bool AudioStream::IsValidAudioFormat(Chunk* aChunk) {
if (aChunk->Rate() != mAudioClock.GetInputRate()) {
LOGW("mismatched sample %u, mInRate=%u", aChunk->Rate(),
mAudioClock.GetInputRate());
return false;
}
return aChunk->Channels() <= 8;
}
void AudioStream::GetUnprocessed(AudioBufferWriter& aWriter) {
TRACE("AudioStream::GetUnprocessed");
AssertIsOnAudioThread();
// Flush the timestretcher pipeline, if we were playing using a playback rate
// other than 1.0.
if (mTimeStretcher) {
// Get number of samples and based on this either receive samples or write
// silence. At worst, the attacker can supply weird sound samples or
// result in us writing silence.
auto numSamples = mTimeStretcher->numSamples().unverified_safe_because(
"We only use this to decide whether to receive samples or write "
"silence.");
if (numSamples) {
RLBoxSoundTouch* timeStretcher = mTimeStretcher;
aWriter.Write(
[timeStretcher](AudioDataValue* aPtr, uint32_t aFrames) {
return timeStretcher->receiveSamples(aPtr, aFrames);
},
aWriter.Available());
// TODO: There might be still unprocessed samples in the stretcher.
// We should either remove or flush them so they won't be in the output
// next time we switch a playback rate other than 1.0.
mTimeStretcher->numUnprocessedSamples().copy_and_verify([](auto samples) {
NS_WARNING_ASSERTION(samples == 0, "no samples");
});
} else {
// Don't need it anymore: playbackRate is 1.0, and the time stretcher has
// been flushed.
delete mTimeStretcher;
mTimeStretcher = nullptr;
}
}
while (aWriter.Available() > 0) {
uint32_t count = mDataSource.PopFrames(aWriter.Ptr(), aWriter.Available(),
mAudioThreadChanged);
if (count == 0) {
break;
}
aWriter.Advance(count);
}
}
void AudioStream::GetTimeStretched(AudioBufferWriter& aWriter) {
TRACE("AudioStream::GetTimeStretched");
AssertIsOnAudioThread();
if (EnsureTimeStretcherInitialized() != NS_OK) {
return;
}
uint32_t toPopFrames =
ceil(aWriter.Available() * mAudioClock.GetPlaybackRate());
// At each iteration, get number of samples and (based on this) write from
// the data source or silence. At worst, if the number of samples is a lie
// (i.e., under attacker control) we'll either not write anything or keep
// writing noise. This is safe because all the memory operations within the
// loop (and after) are checked.
while (mTimeStretcher->numSamples().unverified_safe_because(
"Only used to decide whether to put samples.") <
aWriter.Available()) {
// pop into a temp buffer, and put into the stretcher.
AutoTArray<AudioDataValue, 1000> buf;
auto size = CheckedUint32(mOutChannels) * toPopFrames;
if (!size.isValid()) {
// The overflow should not happen in normal case.
LOGW("Invalid member data: %d channels, %d frames", mOutChannels,
toPopFrames);
return;
}
buf.SetLength(size.value());
// ensure no variable channel count or something like that
uint32_t count =
mDataSource.PopFrames(buf.Elements(), toPopFrames, mAudioThreadChanged);
if (count == 0) {
break;
}
mTimeStretcher->putSamples(buf.Elements(), count);
}
auto* timeStretcher = mTimeStretcher;
aWriter.Write(
[timeStretcher](AudioDataValue* aPtr, uint32_t aFrames) {
return timeStretcher->receiveSamples(aPtr, aFrames);
},
aWriter.Available());
}
bool AudioStream::CheckThreadIdChanged() {
ProfilerThreadId id = profiler_current_thread_id();
if (id != mAudioThreadId) {
mAudioThreadId = id;
mAudioThreadChanged = true;
return true;
}
mAudioThreadChanged = false;
return false;
}
void AudioStream::AssertIsOnAudioThread() const {
// This can be called right after CheckThreadIdChanged, because the audio
// thread can change when not sandboxed.
MOZ_ASSERT(mAudioThreadId.load() == profiler_current_thread_id());
}
void AudioStream::UpdatePlaybackRateIfNeeded() {
AssertIsOnAudioThread();
if (mAudioClock.GetPreservesPitch() == mPreservesPitch &&
mAudioClock.GetPlaybackRate() == mPlaybackRate) {
return;
}
EnsureTimeStretcherInitialized();
mAudioClock.SetPlaybackRate(mPlaybackRate);
mAudioClock.SetPreservesPitch(mPreservesPitch);
if (mPreservesPitch) {
mTimeStretcher->setTempo(mPlaybackRate);
mTimeStretcher->setRate(1.0f);
} else {
mTimeStretcher->setTempo(1.0f);
mTimeStretcher->setRate(mPlaybackRate);
}
}
long AudioStream::DataCallback(void* aBuffer, long aFrames) {
if (CheckThreadIdChanged() && !mSandboxed) {
CallbackThreadRegistry::Get()->Register(mAudioThreadId,
"NativeAudioCallback");
}
WebCore::DenormalDisabler disabler;
if (!mCallbacksStarted) {
mCallbacksStarted = true;
}
TRACE_AUDIO_CALLBACK_BUDGET("AudioStream real-time budget", aFrames,
mAudioClock.GetInputRate());
TRACE("AudioStream::DataCallback");
MOZ_ASSERT(mState != SHUTDOWN, "No data callback after shutdown");
if (SoftRealTimeLimitReached()) {
DemoteThreadFromRealTime();
}
UpdatePlaybackRateIfNeeded();
auto writer = AudioBufferWriter(
Span<AudioDataValue>(reinterpret_cast<AudioDataValue*>(aBuffer),
mOutChannels * aFrames),
mOutChannels, aFrames);
if (mAudioClock.GetInputRate() == mAudioClock.GetOutputRate()) {
GetUnprocessed(writer);
} else {
GetTimeStretched(writer);
}
// Always send audible frames first, and silent frames later.
// Otherwise it will break the assumption of FrameHistory.
if (!mDataSource.Ended()) {
#ifndef XP_MACOSX
MonitorAutoLock mon(mMonitor);
#endif
mAudioClock.UpdateFrameHistory(aFrames - writer.Available(),
writer.Available(), mAudioThreadChanged);
if (writer.Available() > 0) {
TRACE_COMMENT("AudioStream::DataCallback", "Underrun: %d frames missing",
writer.Available());
LOGW("lost %d frames", writer.Available());
writer.WriteZeros(writer.Available());
}
} else {
// No more new data in the data source, and the drain has completed. We
// don't need the time stretcher anymore at this point.
if (mTimeStretcher && writer.Available()) {
delete mTimeStretcher;
mTimeStretcher = nullptr;
}
#ifndef XP_MACOSX
MonitorAutoLock mon(mMonitor);
#endif
mAudioClock.UpdateFrameHistory(aFrames - writer.Available(), 0,
mAudioThreadChanged);
}
mDumpFile.Write(static_cast<const AudioDataValue*>(aBuffer),
aFrames * mOutChannels);
if (!mSandboxed && writer.Available() != 0) {
CallbackThreadRegistry::Get()->Unregister(mAudioThreadId);
}
return aFrames - writer.Available();
}
void AudioStream::StateCallback(cubeb_state aState) {
MOZ_ASSERT(mState != SHUTDOWN, "No state callback after shutdown");
LOG("StateCallback, mState=%d cubeb_state=%d", mState.load(), aState);
MonitorAutoLock mon(mMonitor);
if (aState == CUBEB_STATE_DRAINED) {
LOG("Drained");
mState = DRAINED;
mPlaybackComplete = true;
mEndedPromise.ResolveIfExists(true, __func__);
} else if (aState == CUBEB_STATE_ERROR) {
LOGE("StateCallback() state %d cubeb error", mState.load());
mState = ERRORED;
mPlaybackComplete = true;
mEndedPromise.RejectIfExists(NS_ERROR_FAILURE, __func__);
}
}
bool AudioStream::IsPlaybackCompleted() const { return mPlaybackComplete; }
AudioClock::AudioClock(uint32_t aInRate)
: mOutRate(aInRate),
mInRate(aInRate),
mPreservesPitch(true),
mFrameHistory(new FrameHistory()) {}
// Audio thread only
void AudioClock::UpdateFrameHistory(uint32_t aServiced, uint32_t aUnderrun,
bool aAudioThreadChanged) {
#ifdef XP_MACOSX
if (aAudioThreadChanged) {
mCallbackInfoQueue.ResetProducerThreadId();
}
// Flush the local items, if any, and then attempt to enqueue the current
// item. This is only a fallback mechanism, under non-critical load this is
// just going to enqueue an item in the queue.
while (!mAudioThreadCallbackInfo.IsEmpty()) {
CallbackInfo& info = mAudioThreadCallbackInfo[0];
// If still full, keep it audio-thread side for now.
if (mCallbackInfoQueue.Enqueue(info) != 1) {
break;
}
mAudioThreadCallbackInfo.RemoveElementAt(0);
}
CallbackInfo info(aServiced, aUnderrun, mOutRate);
if (mCallbackInfoQueue.Enqueue(info) != 1) {
NS_WARNING(
"mCallbackInfoQueue full, storing the values in the audio thread.");
mAudioThreadCallbackInfo.AppendElement(info);
}
#else
MutexAutoLock lock(mMutex);
mFrameHistory->Append(aServiced, aUnderrun, mOutRate);
#endif
}
int64_t AudioClock::GetPositionInFrames(int64_t aFrames) {
CheckedInt64 v = UsecsToFrames(GetPosition(aFrames), mInRate);
return v.isValid() ? v.value() : -1;
}
int64_t AudioClock::GetPosition(int64_t frames) {
#ifdef XP_MACOSX
// Dequeue all history info, and apply them before returning the position
// based on frame history.
CallbackInfo info;
while (mCallbackInfoQueue.Dequeue(&info, 1)) {
mFrameHistory->Append(info.mServiced, info.mUnderrun, info.mOutputRate);
}
#else
MutexAutoLock lock(mMutex);
#endif
return mFrameHistory->GetPosition(frames);
}
void AudioClock::SetPlaybackRate(double aPlaybackRate) {
mOutRate = static_cast<uint32_t>(mInRate / aPlaybackRate);
}
double AudioClock::GetPlaybackRate() const {
return static_cast<double>(mInRate) / mOutRate;
}
void AudioClock::SetPreservesPitch(bool aPreservesPitch) {
mPreservesPitch = aPreservesPitch;
}
bool AudioClock::GetPreservesPitch() const { return mPreservesPitch; }
} // namespace mozilla