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gecko-dev/dom/media/AudioStream.cpp

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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
* file, You can obtain one at http://mozilla.org/MPL/2.0/. */
#include <stdio.h>
#include <math.h>
#include <string.h>
#include "mozilla/Logging.h"
#include "prdtoa.h"
#include "AudioStream.h"
#include "VideoUtils.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 "nsPrintfCString.h"
#include "gfxPrefs.h"
#include "AudioConverter.h"
#if defined(XP_WIN)
# include "nsXULAppAPI.h"
#endif
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, int 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 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)
: mMonitor("AudioStream"),
mChannels(0),
mOutChannels(0),
mTimeStretcher(nullptr),
mDumpFile(nullptr),
mState(INITIALIZED),
mDataSource(aSource),
mPrefillQuirk(false) {
#if defined(XP_WIN)
if (XRE_IsContentProcess()) {
audio::AudioNotificationReceiver::Register(this);
}
#endif
}
AudioStream::~AudioStream() {
LOG("deleted, state %d", mState);
MOZ_ASSERT(mState == SHUTDOWN && !mCubebStream,
"Should've called Shutdown() before deleting an AudioStream");
if (mDumpFile) {
fclose(mDumpFile);
}
if (mTimeStretcher) {
soundtouch::destroySoundTouchObj(mTimeStretcher);
}
#if defined(XP_WIN)
if (XRE_IsContentProcess()) {
audio::AudioNotificationReceiver::Unregister(this);
}
#endif
}
size_t AudioStream::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const {
size_t amount = aMallocSizeOf(this);
// Possibly add in the future:
// - mTimeStretcher
// - mCubebStream
return amount;
}
nsresult AudioStream::EnsureTimeStretcherInitializedUnlocked() {
mMonitor.AssertCurrentThreadOwns();
if (!mTimeStretcher) {
mTimeStretcher = soundtouch::createSoundTouchObj();
mTimeStretcher->setSampleRate(mAudioClock.GetInputRate());
mTimeStretcher->setChannels(mOutChannels);
mTimeStretcher->setPitch(1.0);
}
return NS_OK;
}
nsresult AudioStream::SetPlaybackRate(double aPlaybackRate) {
// MUST lock since the rate transposer is used from the cubeb callback,
// and rate changes can cause the buffer to be reallocated
MonitorAutoLock mon(mMonitor);
NS_ASSERTION(
aPlaybackRate > 0.0,
"Can't handle negative or null playbackrate in the AudioStream.");
// Avoid instantiating the resampler if we are not changing the playback rate.
// GetPreservesPitch/SetPreservesPitch don't need locking before calling
if (aPlaybackRate == mAudioClock.GetPlaybackRate()) {
return NS_OK;
}
if (EnsureTimeStretcherInitializedUnlocked() != NS_OK) {
return NS_ERROR_FAILURE;
}
mAudioClock.SetPlaybackRate(aPlaybackRate);
if (mAudioClock.GetPreservesPitch()) {
mTimeStretcher->setTempo(aPlaybackRate);
mTimeStretcher->setRate(1.0f);
} else {
mTimeStretcher->setTempo(1.0f);
mTimeStretcher->setRate(aPlaybackRate);
}
return NS_OK;
}
nsresult AudioStream::SetPreservesPitch(bool aPreservesPitch) {
// MUST lock since the rate transposer is used from the cubeb callback,
// and rate changes can cause the buffer to be reallocated
MonitorAutoLock mon(mMonitor);
// Avoid instantiating the timestretcher instance if not needed.
if (aPreservesPitch == mAudioClock.GetPreservesPitch()) {
return NS_OK;
}
if (EnsureTimeStretcherInitializedUnlocked() != NS_OK) {
return NS_ERROR_FAILURE;
}
if (aPreservesPitch == true) {
mTimeStretcher->setTempo(mAudioClock.GetPlaybackRate());
mTimeStretcher->setRate(1.0f);
} else {
mTimeStretcher->setTempo(1.0f);
mTimeStretcher->setRate(mAudioClock.GetPlaybackRate());
}
mAudioClock.SetPreservesPitch(aPreservesPitch);
return NS_OK;
}
static void SetUint16LE(uint8_t* aDest, uint16_t aValue) {
aDest[0] = aValue & 0xFF;
aDest[1] = aValue >> 8;
}
static void SetUint32LE(uint8_t* aDest, uint32_t aValue) {
SetUint16LE(aDest, aValue & 0xFFFF);
SetUint16LE(aDest + 2, aValue >> 16);
}
static FILE* OpenDumpFile(uint32_t aChannels, uint32_t aRate) {
/**
* When MOZ_DUMP_AUDIO is set in the environment (to anything),
* we'll drop a series of files in the current working directory named
* dumped-audio-<nnn>.wav, one per AudioStream created, containing
* the audio for the stream including any skips due to underruns.
*/
static Atomic<int> gDumpedAudioCount(0);
if (!getenv("MOZ_DUMP_AUDIO")) return nullptr;
char buf[100];
SprintfLiteral(buf, "dumped-audio-%d.wav", ++gDumpedAudioCount);
FILE* f = fopen(buf, "wb");
if (!f) return nullptr;
uint8_t header[] = {
// RIFF header
0x52, 0x49, 0x46, 0x46, 0x00, 0x00, 0x00, 0x00, 0x57, 0x41, 0x56, 0x45,
// fmt chunk. We always write 16-bit samples.
0x66, 0x6d, 0x74, 0x20, 0x10, 0x00, 0x00, 0x00, 0x01, 0x00, 0xFF, 0xFF,
0xFF, 0xFF, 0xFF, 0xFF, 0x00, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x10, 0x00,
// data chunk
0x64, 0x61, 0x74, 0x61, 0xFE, 0xFF, 0xFF, 0x7F};
static const int CHANNEL_OFFSET = 22;
static const int SAMPLE_RATE_OFFSET = 24;
static const int BLOCK_ALIGN_OFFSET = 32;
SetUint16LE(header + CHANNEL_OFFSET, aChannels);
SetUint32LE(header + SAMPLE_RATE_OFFSET, aRate);
SetUint16LE(header + BLOCK_ALIGN_OFFSET, aChannels * 2);
Unused << fwrite(header, sizeof(header), 1, f);
return f;
}
template <typename T>
typename EnableIf<IsSame<T, int16_t>::value, void>::Type WriteDumpFileHelper(
T* aInput, size_t aSamples, FILE* aFile) {
Unused << fwrite(aInput, sizeof(T), aSamples, aFile);
}
template <typename T>
typename EnableIf<IsSame<T, float>::value, void>::Type WriteDumpFileHelper(
T* aInput, size_t aSamples, FILE* aFile) {
AutoTArray<uint8_t, 1024 * 2> buf;
buf.SetLength(aSamples * 2);
uint8_t* output = buf.Elements();
for (uint32_t i = 0; i < aSamples; ++i) {
SetUint16LE(output + i * 2, int16_t(aInput[i] * 32767.0f));
}
Unused << fwrite(output, 2, aSamples, aFile);
fflush(aFile);
}
static void WriteDumpFile(FILE* aDumpFile, AudioStream* aStream,
uint32_t aFrames, void* aBuffer) {
if (!aDumpFile) return;
uint32_t samples = aStream->GetOutChannels() * aFrames;
using SampleT = AudioSampleTraits<AUDIO_OUTPUT_FORMAT>::Type;
WriteDumpFileHelper(reinterpret_cast<SampleT*>(aBuffer), samples, aDumpFile);
}
template <AudioSampleFormat N>
struct ToCubebFormat {
static const cubeb_sample_format value = CUBEB_SAMPLE_FLOAT32NE;
};
template <>
struct ToCubebFormat<AUDIO_FORMAT_S16> {
static const cubeb_sample_format value = CUBEB_SAMPLE_S16NE;
};
template <typename Function, typename... Args>
int AudioStream::InvokeCubeb(Function aFunction, Args&&... aArgs) {
MonitorAutoUnlock mon(mMonitor);
return aFunction(mCubebStream.get(), std::forward<Args>(aArgs)...);
}
nsresult AudioStream::Init(uint32_t aNumChannels,
AudioConfig::ChannelLayout::ChannelMap aChannelMap,
uint32_t aRate, AudioDeviceInfo* aSinkInfo) {
auto startTime = TimeStamp::Now();
LOG("%s channels: %d, rate: %d", __FUNCTION__, aNumChannels, aRate);
mChannels = aNumChannels;
mOutChannels = aNumChannels;
mDumpFile = OpenDumpFile(aNumChannels, aRate);
mSinkInfo = aSinkInfo;
cubeb_stream_params params;
params.rate = aRate;
params.channels = mOutChannels;
params.layout = static_cast<uint32_t>(aChannelMap);
params.format = ToCubebFormat<AUDIO_OUTPUT_FORMAT>::value;
params.prefs = CubebUtils::GetDefaultStreamPrefs();
mAudioClock.Init(aRate);
cubeb* cubebContext = CubebUtils::GetCubebContext();
if (!cubebContext) {
LOGE("Can't get cubeb context!");
CubebUtils::ReportCubebStreamInitFailure(true);
return NS_ERROR_DOM_MEDIA_CUBEB_INITIALIZATION_ERR;
}
// cubeb's winmm backend prefills buffers on init rather than stream start.
// See https://github.com/kinetiknz/cubeb/issues/150
mPrefillQuirk = !strcmp(cubeb_get_backend_id(cubebContext), "winmm");
return OpenCubeb(cubebContext, params, startTime,
CubebUtils::GetFirstStream());
}
nsresult AudioStream::OpenCubeb(cubeb* aContext, cubeb_stream_params& aParams,
TimeStamp aStartTime, bool aIsFirst) {
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 (cubeb_stream_init(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());
Telemetry::Accumulate(aIsFirst ? Telemetry::AUDIOSTREAM_FIRST_OPEN_MS
: Telemetry::AUDIOSTREAM_LATER_OPEN_MS,
timeDelta.ToMilliseconds());
return NS_OK;
}
void AudioStream::SetVolume(double aVolume) {
MOZ_ASSERT(aVolume >= 0.0 && aVolume <= 1.0, "Invalid volume");
#ifdef DEBUG
{
MonitorAutoLock mon(mMonitor);
MOZ_ASSERT(mState != SHUTDOWN, "Don't set volume after shutdown.");
}
#endif
if (cubeb_stream_set_volume(mCubebStream.get(),
aVolume * CubebUtils::GetVolumeScale()) !=
CUBEB_OK) {
LOGE("Could not change volume on cubeb stream.");
}
}
nsresult AudioStream::Start() {
MonitorAutoLock mon(mMonitor);
MOZ_ASSERT(mState == INITIALIZED);
mState = STARTED;
auto r = InvokeCubeb(cubeb_stream_start);
if (r != CUBEB_OK) {
mState = ERRORED;
}
LOG("started, state %s", mState == STARTED
? "STARTED"
: mState == DRAINED ? "DRAINED" : "ERRORED");
if (mState == STARTED || mState == DRAINED) {
return NS_OK;
}
return NS_ERROR_FAILURE;
}
void AudioStream::Pause() {
MonitorAutoLock mon(mMonitor);
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;
}
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() {
MonitorAutoLock mon(mMonitor);
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;
}
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() {
MonitorAutoLock mon(mMonitor);
LOG("Shutdown, state %d", mState);
if (mCubebStream) {
MonitorAutoUnlock mon(mMonitor);
// Force stop to put the cubeb stream in a stable state before deletion.
cubeb_stream_stop(mCubebStream.get());
// Must not try to shut down cubeb from within the lock! wasapi may still
// call our callback after Pause()/stop()!?! Bug 996162
mCubebStream.reset();
}
mState = SHUTDOWN;
}
#if defined(XP_WIN)
void AudioStream::ResetDefaultDevice() {
MonitorAutoLock mon(mMonitor);
if (mState != STARTED && mState != STOPPED) {
return;
}
MOZ_ASSERT(mCubebStream);
auto r = InvokeCubeb(cubeb_stream_reset_default_device);
if (!(r == CUBEB_OK || r == CUBEB_ERROR_NOT_SUPPORTED)) {
mState = ERRORED;
}
}
#endif
int64_t AudioStream::GetPosition() {
MonitorAutoLock mon(mMonitor);
int64_t frames = GetPositionInFramesUnlocked();
return frames >= 0 ? mAudioClock.GetPosition(frames) : -1;
}
int64_t AudioStream::GetPositionInFrames() {
MonitorAutoLock mon(mMonitor);
int64_t frames = GetPositionInFramesUnlocked();
return frames >= 0 ? mAudioClock.GetPositionInFrames(frames) : -1;
}
int64_t AudioStream::GetPositionInFramesUnlocked() {
mMonitor.AssertCurrentThreadOwns();
if (mState == ERRORED) {
return -1;
}
uint64_t position = 0;
if (InvokeCubeb(cubeb_stream_get_position, &position) != CUBEB_OK) {
return -1;
}
return 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;
}
if (aChunk->Channels() > 8) {
return false;
}
return true;
}
void AudioStream::GetUnprocessed(AudioBufferWriter& aWriter) {
mMonitor.AssertCurrentThreadOwns();
// Flush the timestretcher pipeline, if we were playing using a playback rate
// other than 1.0.
if (mTimeStretcher && mTimeStretcher->numSamples()) {
auto 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.
NS_WARNING_ASSERTION(mTimeStretcher->numUnprocessedSamples() == 0,
"no samples");
}
while (aWriter.Available() > 0) {
UniquePtr<Chunk> c = mDataSource.PopFrames(aWriter.Available());
if (c->Frames() == 0) {
break;
}
MOZ_ASSERT(c->Frames() <= aWriter.Available());
if (IsValidAudioFormat(c.get())) {
aWriter.Write(c->Data(), c->Frames());
} else {
// Write silence if invalid format.
aWriter.WriteZeros(c->Frames());
}
}
}
void AudioStream::GetTimeStretched(AudioBufferWriter& aWriter) {
mMonitor.AssertCurrentThreadOwns();
// We need to call the non-locking version, because we already have the lock.
if (EnsureTimeStretcherInitializedUnlocked() != NS_OK) {
return;
}
uint32_t toPopFrames =
ceil(aWriter.Available() * mAudioClock.GetPlaybackRate());
while (mTimeStretcher->numSamples() < aWriter.Available()) {
UniquePtr<Chunk> c = mDataSource.PopFrames(toPopFrames);
if (c->Frames() == 0) {
break;
}
MOZ_ASSERT(c->Frames() <= toPopFrames);
if (IsValidAudioFormat(c.get())) {
mTimeStretcher->putSamples(c->Data(), c->Frames());
} else {
// Write silence if invalid format.
AutoTArray<AudioDataValue, 1000> buf;
auto size = CheckedUint32(mOutChannels) * c->Frames();
if (!size.isValid()) {
// The overflow should not happen in normal case.
LOGW("Invalid member data: %d channels, %d frames", mOutChannels,
c->Frames());
return;
}
buf.SetLength(size.value());
size = size * sizeof(AudioDataValue);
if (!size.isValid()) {
LOGW("The required memory size is too large.");
return;
}
memset(buf.Elements(), 0, size.value());
mTimeStretcher->putSamples(buf.Elements(), c->Frames());
}
}
auto timeStretcher = mTimeStretcher;
aWriter.Write(
[timeStretcher](AudioDataValue* aPtr, uint32_t aFrames) {
return timeStretcher->receiveSamples(aPtr, aFrames);
},
aWriter.Available());
}
long AudioStream::DataCallback(void* aBuffer, long aFrames) {
MonitorAutoLock mon(mMonitor);
MOZ_ASSERT(mState != SHUTDOWN, "No data callback after shutdown");
auto writer = AudioBufferWriter(reinterpret_cast<AudioDataValue*>(aBuffer),
mOutChannels, aFrames);
if (mPrefillQuirk) {
// Don't consume audio data until Start() is called.
// Expected only with cubeb winmm backend.
if (mState == INITIALIZED) {
NS_WARNING("data callback fires before cubeb_stream_start() is called");
mAudioClock.UpdateFrameHistory(0, aFrames);
return writer.WriteZeros(aFrames);
}
} else {
MOZ_ASSERT(mState != INITIALIZED);
}
// NOTE: wasapi (others?) can call us back *after* stop()/Shutdown() (mState
// == SHUTDOWN) Bug 996162
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()) {
mAudioClock.UpdateFrameHistory(aFrames - writer.Available(),
writer.Available());
if (writer.Available() > 0) {
LOGW("lost %d frames", writer.Available());
writer.WriteZeros(writer.Available());
}
} else {
// No more new data in the data source. Don't send silent frames so the
// cubeb stream can start draining.
mAudioClock.UpdateFrameHistory(aFrames - writer.Available(), 0);
}
WriteDumpFile(mDumpFile, this, aFrames, aBuffer);
return aFrames - writer.Available();
}
void AudioStream::StateCallback(cubeb_state aState) {
MonitorAutoLock mon(mMonitor);
MOZ_ASSERT(mState != SHUTDOWN, "No state callback after shutdown");
LOG("StateCallback, mState=%d cubeb_state=%d", mState, aState);
if (aState == CUBEB_STATE_DRAINED) {
mState = DRAINED;
mDataSource.Drained();
} else if (aState == CUBEB_STATE_ERROR) {
LOGE("StateCallback() state %d cubeb error", mState);
mState = ERRORED;
}
}
AudioClock::AudioClock()
: mOutRate(0),
mInRate(0),
mPreservesPitch(true),
mFrameHistory(new FrameHistory()) {}
void AudioClock::Init(uint32_t aRate) {
mOutRate = aRate;
mInRate = aRate;
}
void AudioClock::UpdateFrameHistory(uint32_t aServiced, uint32_t aUnderrun) {
mFrameHistory->Append(aServiced, aUnderrun, mOutRate);
}
int64_t AudioClock::GetPositionInFrames(int64_t aFrames) const {
CheckedInt64 v = UsecsToFrames(GetPosition(aFrames), mInRate);
return v.isValid() ? v.value() : -1;
}
int64_t AudioClock::GetPosition(int64_t frames) const {
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
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