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tenfourfox/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/Snprintf.h"
#include <algorithm>
#include "mozilla/Telemetry.h"
#include "CubebUtils.h"
#include "nsPrintfCString.h"
#include "gfxPrefs.h"
namespace mozilla {
#ifdef LOG
#undef LOG
#endif
LazyLogModule gAudioStreamLog("AudioStream");
// For simple logs
#define LOG(x) MOZ_LOG(gAudioStreamLog, mozilla::LogLevel::Debug, x)
/**
* 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 int gDumpedAudioCount = 0;
/**
* 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;
int 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, int 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:
nsAutoTArray<Chunk, 7> mChunks;
int64_t mBaseOffset;
double mBasePosition;
};
AudioStream::AudioStream()
: mMonitor("AudioStream")
, mInRate(0)
, mOutRate(0)
, mChannels(0)
, mOutChannels(0)
, mWritten(0)
, mAudioClock(this)
, mTimeStretcher(nullptr)
, mDumpFile(nullptr)
, mBytesPerFrame(0)
, mState(INITIALIZED)
, mIsMonoAudioEnabled(gfxPrefs::MonoAudio())
{
}
AudioStream::~AudioStream()
{
LOG(("AudioStream: delete %p, state %d", this, mState));
MOZ_ASSERT(mState == SHUTDOWN && !mCubebStream,
"Should've called Shutdown() before deleting an AudioStream");
if (mDumpFile) {
fclose(mDumpFile);
}
if (mTimeStretcher) {
soundtouch::destroySoundTouchObj(mTimeStretcher);
}
}
size_t
AudioStream::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
{
size_t amount = aMallocSizeOf(this);
// Possibly add in the future:
// - mTimeStretcher
// - mCubebStream
amount += mBuffer.SizeOfExcludingThis(aMallocSizeOf);
return amount;
}
nsresult AudioStream::EnsureTimeStretcherInitializedUnlocked()
{
mMonitor.AssertCurrentThreadOwns();
if (!mTimeStretcher) {
mTimeStretcher = soundtouch::createSoundTouchObj();
mTimeStretcher->setSampleRate(mInRate);
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.SetPlaybackRateUnlocked(aPlaybackRate);
mOutRate = mInRate / 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;
}
int64_t AudioStream::GetWritten()
{
MonitorAutoLock mon(mMonitor);
return mWritten;
}
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(AudioStream* aStream)
{
if (!getenv("MOZ_DUMP_AUDIO"))
return nullptr;
char buf[100];
snprintf_literal(buf, "dumped-audio-%d.wav", gDumpedAudioCount);
FILE* f = fopen(buf, "wb");
if (!f)
return nullptr;
++gDumpedAudioCount;
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, aStream->GetChannels());
SetUint32LE(header + SAMPLE_RATE_OFFSET, aStream->GetRate());
SetUint16LE(header + BLOCK_ALIGN_OFFSET, aStream->GetChannels()*2);
fwrite(header, sizeof(header), 1, f);
return f;
}
static void
WriteDumpFile(FILE* aDumpFile, AudioStream* aStream, uint32_t aFrames,
void* aBuffer)
{
if (!aDumpFile)
return;
uint32_t samples = aStream->GetOutChannels()*aFrames;
if (AUDIO_OUTPUT_FORMAT == AUDIO_FORMAT_S16) {
fwrite(aBuffer, 2, samples, aDumpFile);
return;
}
NS_ASSERTION(AUDIO_OUTPUT_FORMAT == AUDIO_FORMAT_FLOAT32, "bad format");
nsAutoTArray<uint8_t, 1024*2> buf;
buf.SetLength(samples*2);
float* input = static_cast<float*>(aBuffer);
uint8_t* output = buf.Elements();
for (uint32_t i = 0; i < samples; ++i) {
SetUint16LE(output + i*2, int16_t(input[i]*32767.0f));
}
fwrite(output, 2, samples, aDumpFile);
fflush(aDumpFile);
}
nsresult
AudioStream::Init(int32_t aNumChannels, int32_t aRate,
const dom::AudioChannel aAudioChannel)
{
mStartTime = TimeStamp::Now();
mIsFirst = CubebUtils::GetFirstStream();
if (!CubebUtils::GetCubebContext() || aNumChannels < 0 || aRate < 0) {
return NS_ERROR_FAILURE;
}
MOZ_LOG(gAudioStreamLog, LogLevel::Debug,
("%s channels: %d, rate: %d for %p", __FUNCTION__, aNumChannels, aRate, this));
mInRate = mOutRate = aRate;
mChannels = aNumChannels;
mOutChannels = (aNumChannels > 2) ? 2 : aNumChannels;
mDumpFile = OpenDumpFile(this);
cubeb_stream_params params;
params.rate = aRate;
params.channels = mOutChannels;
#if defined(__ANDROID__)
#if defined(MOZ_B2G)
mAudioChannel = aAudioChannel;
params.stream_type = CubebUtils::ConvertChannelToCubebType(aAudioChannel);
#else
mAudioChannel = dom::AudioChannel::Content;
params.stream_type = CUBEB_STREAM_TYPE_MUSIC;
#endif
if (params.stream_type == CUBEB_STREAM_TYPE_MAX) {
return NS_ERROR_INVALID_ARG;
}
#endif
if (AUDIO_OUTPUT_FORMAT == AUDIO_FORMAT_S16) {
params.format = CUBEB_SAMPLE_S16NE;
} else {
params.format = CUBEB_SAMPLE_FLOAT32NE;
}
mBytesPerFrame = sizeof(AudioDataValue) * mOutChannels;
mAudioClock.Init();
// Size mBuffer for one second of audio. This value is arbitrary, and was
// selected based on the observed behaviour of the existing AudioStream
// implementations.
uint32_t bufferLimit = FramesToBytes(aRate);
MOZ_ASSERT(bufferLimit % mBytesPerFrame == 0, "Must buffer complete frames");
mBuffer.SetCapacity(bufferLimit);
return OpenCubeb(params);
}
// This code used to live inside AudioStream::Init(), but on Mac (others?)
// it has been known to take 300-800 (or even 8500) ms to execute(!)
nsresult
AudioStream::OpenCubeb(cubeb_stream_params &aParams)
{
cubeb* cubebContext = CubebUtils::GetCubebContext();
if (!cubebContext) {
NS_WARNING("Can't get cubeb context!");
MonitorAutoLock mon(mMonitor);
mState = AudioStream::ERRORED;
return NS_ERROR_FAILURE;
}
// If the latency pref is set, use it. Otherwise, if this stream is intended
// for low latency playback, try to get the lowest latency possible.
// Otherwise, for normal streams, use 100ms.
uint32_t latency = CubebUtils::GetCubebLatency();
{
cubeb_stream* stream;
if (cubeb_stream_init(cubebContext, &stream, "AudioStream", aParams,
latency, DataCallback_S, StateCallback_S, this) == CUBEB_OK) {
MonitorAutoLock mon(mMonitor);
MOZ_ASSERT(mState != SHUTDOWN);
mCubebStream.reset(stream);
} else {
MonitorAutoLock mon(mMonitor);
mState = ERRORED;
NS_WARNING(nsPrintfCString("AudioStream::OpenCubeb() %p failed to init cubeb", this).get());
return NS_ERROR_FAILURE;
}
}
mState = INITIALIZED;
if (!mStartTime.IsNull()) {
TimeDuration timeDelta = TimeStamp::Now() - mStartTime;
LOG(("AudioStream creation time %sfirst: %u ms", mIsFirst ? "" : "not ",
(uint32_t) timeDelta.ToMilliseconds()));
Telemetry::Accumulate(mIsFirst ? Telemetry::AUDIOSTREAM_FIRST_OPEN_MS :
Telemetry::AUDIOSTREAM_LATER_OPEN_MS, timeDelta.ToMilliseconds());
}
return NS_OK;
}
// aTime is the time in ms the samples were inserted into MediaStreamGraph
nsresult
AudioStream::Write(const AudioDataValue* aBuf, uint32_t aFrames)
{
MonitorAutoLock mon(mMonitor);
if (mState == ERRORED) {
return NS_ERROR_FAILURE;
}
NS_ASSERTION(mState == INITIALIZED || mState == STARTED || mState == RUNNING,
"Stream write in unexpected state.");
// Downmix to Stereo.
if (mChannels > 2 && mChannels <= 8) {
DownmixAudioToStereo(const_cast<AudioDataValue*> (aBuf), mChannels, aFrames);
} else if (mChannels > 8) {
return NS_ERROR_FAILURE;
}
if (mChannels >= 2 && mIsMonoAudioEnabled) {
DownmixStereoToMono(const_cast<AudioDataValue*> (aBuf), aFrames);
}
const uint8_t* src = reinterpret_cast<const uint8_t*>(aBuf);
uint32_t bytesToCopy = FramesToBytes(aFrames);
while (bytesToCopy > 0) {
uint32_t available = std::min(bytesToCopy, mBuffer.Available());
MOZ_ASSERT(available % mBytesPerFrame == 0,
"Must copy complete frames.");
mBuffer.AppendElements(src, available);
src += available;
bytesToCopy -= available;
if (bytesToCopy > 0) {
// If we are not playing, but our buffer is full, start playing to make
// room for soon-to-be-decoded data.
if (mState != STARTED && mState != RUNNING) {
MOZ_LOG(gAudioStreamLog, LogLevel::Warning, ("Starting stream %p in Write (%u waiting)",
this, bytesToCopy));
StartUnlocked();
if (mState == ERRORED) {
return NS_ERROR_FAILURE;
}
}
MOZ_LOG(gAudioStreamLog, LogLevel::Warning, ("Stream %p waiting in Write() (%u waiting)",
this, bytesToCopy));
mon.Wait();
}
}
mWritten += aFrames;
return NS_OK;
}
uint32_t
AudioStream::Available()
{
MonitorAutoLock mon(mMonitor);
MOZ_ASSERT(mBuffer.Length() % mBytesPerFrame == 0, "Buffer invariant violated.");
return BytesToFrames(mBuffer.Available());
}
void
AudioStream::SetVolume(double aVolume)
{
MOZ_ASSERT(aVolume >= 0.0 && aVolume <= 1.0, "Invalid volume");
if (cubeb_stream_set_volume(mCubebStream.get(), aVolume * CubebUtils::GetVolumeScale()) != CUBEB_OK) {
NS_WARNING("Could not change volume on cubeb stream.");
}
}
void
AudioStream::Cancel()
{
MonitorAutoLock mon(mMonitor);
mState = ERRORED;
mon.NotifyAll();
}
void
AudioStream::Drain()
{
MonitorAutoLock mon(mMonitor);
LOG(("AudioStream::Drain() for %p, state %d, avail %u", this, mState, mBuffer.Available()));
if (mState != STARTED && mState != RUNNING) {
NS_ASSERTION(mState == ERRORED || mBuffer.Available() == 0, "Draining without full buffer of unplayed audio");
return;
}
mState = DRAINING;
while (mState == DRAINING) {
mon.Wait();
}
}
void
AudioStream::Start()
{
MonitorAutoLock mon(mMonitor);
StartUnlocked();
}
void
AudioStream::StartUnlocked()
{
mMonitor.AssertCurrentThreadOwns();
if (!mCubebStream) {
return;
}
if (mState == INITIALIZED) {
int r;
{
MonitorAutoUnlock mon(mMonitor);
r = cubeb_stream_start(mCubebStream.get());
}
mState = r == CUBEB_OK ? STARTED : ERRORED;
LOG(("AudioStream: started %p, state %s", this, mState == STARTED ? "STARTED" : "ERRORED"));
}
}
void
AudioStream::Pause()
{
MonitorAutoLock mon(mMonitor);
if (mState == ERRORED) {
return;
}
if (!mCubebStream || (mState != STARTED && mState != RUNNING)) {
mState = STOPPED; // which also tells async OpenCubeb not to start, just init
return;
}
int r;
{
MonitorAutoUnlock mon(mMonitor);
r = cubeb_stream_stop(mCubebStream.get());
}
if (mState != ERRORED && r == CUBEB_OK) {
mState = STOPPED;
}
}
void
AudioStream::Resume()
{
MonitorAutoLock mon(mMonitor);
if (!mCubebStream || mState != STOPPED) {
return;
}
int r;
{
MonitorAutoUnlock mon(mMonitor);
r = cubeb_stream_start(mCubebStream.get());
}
if (mState != ERRORED && r == CUBEB_OK) {
mState = STARTED;
}
}
void
AudioStream::Shutdown()
{
MonitorAutoLock mon(mMonitor);
LOG(("AudioStream: Shutdown %p, state %d", this, 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;
}
int64_t
AudioStream::GetPosition()
{
MonitorAutoLock mon(mMonitor);
return mAudioClock.GetPositionUnlocked();
}
int64_t
AudioStream::GetPositionInFrames()
{
MonitorAutoLock mon(mMonitor);
return mAudioClock.GetPositionInFrames();
}
int64_t
AudioStream::GetPositionInFramesUnlocked()
{
mMonitor.AssertCurrentThreadOwns();
if (!mCubebStream || mState == ERRORED) {
return -1;
}
uint64_t position = 0;
{
MonitorAutoUnlock mon(mMonitor);
if (cubeb_stream_get_position(mCubebStream.get(), &position) != CUBEB_OK) {
return -1;
}
}
return std::min<uint64_t>(position, INT64_MAX);
}
bool
AudioStream::IsPaused()
{
MonitorAutoLock mon(mMonitor);
return mState == STOPPED;
}
long
AudioStream::GetUnprocessed(void* aBuffer, long aFrames)
{
mMonitor.AssertCurrentThreadOwns();
uint8_t* wpos = reinterpret_cast<uint8_t*>(aBuffer);
// Flush the timestretcher pipeline, if we were playing using a playback rate
// other than 1.0.
uint32_t flushedFrames = 0;
if (mTimeStretcher && mTimeStretcher->numSamples()) {
flushedFrames = mTimeStretcher->receiveSamples(reinterpret_cast<AudioDataValue*>(wpos), aFrames);
wpos += FramesToBytes(flushedFrames);
}
uint32_t toPopBytes = FramesToBytes(aFrames - flushedFrames);
uint32_t available = std::min(toPopBytes, mBuffer.Length());
void* input[2];
uint32_t input_size[2];
mBuffer.PopElements(available, &input[0], &input_size[0], &input[1], &input_size[1]);
memcpy(wpos, input[0], input_size[0]);
wpos += input_size[0];
memcpy(wpos, input[1], input_size[1]);
return BytesToFrames(available) + flushedFrames;
}
long
AudioStream::GetTimeStretched(void* aBuffer, long aFrames)
{
mMonitor.AssertCurrentThreadOwns();
long processedFrames = 0;
// We need to call the non-locking version, because we already have the lock.
if (EnsureTimeStretcherInitializedUnlocked() != NS_OK) {
return 0;
}
uint8_t* wpos = reinterpret_cast<uint8_t*>(aBuffer);
double playbackRate = static_cast<double>(mInRate) / mOutRate;
uint32_t toPopBytes = FramesToBytes(ceil(aFrames * playbackRate));
uint32_t available = 0;
bool lowOnBufferedData = false;
do {
// Check if we already have enough data in the time stretcher pipeline.
if (mTimeStretcher->numSamples() <= static_cast<uint32_t>(aFrames)) {
void* input[2];
uint32_t input_size[2];
available = std::min(mBuffer.Length(), toPopBytes);
if (available != toPopBytes) {
lowOnBufferedData = true;
}
mBuffer.PopElements(available, &input[0], &input_size[0],
&input[1], &input_size[1]);
for(uint32_t i = 0; i < 2; i++) {
mTimeStretcher->putSamples(reinterpret_cast<AudioDataValue*>(input[i]), BytesToFrames(input_size[i]));
}
}
uint32_t receivedFrames = mTimeStretcher->receiveSamples(reinterpret_cast<AudioDataValue*>(wpos), aFrames - processedFrames);
wpos += FramesToBytes(receivedFrames);
processedFrames += receivedFrames;
} while (processedFrames < aFrames && !lowOnBufferedData);
return processedFrames;
}
long
AudioStream::DataCallback(void* aBuffer, long aFrames)
{
MonitorAutoLock mon(mMonitor);
MOZ_ASSERT(mState != SHUTDOWN, "No data callback after shutdown");
uint32_t available = std::min(static_cast<uint32_t>(FramesToBytes(aFrames)), mBuffer.Length());
MOZ_ASSERT(available % mBytesPerFrame == 0, "Must copy complete frames");
AudioDataValue* output = reinterpret_cast<AudioDataValue*>(aBuffer);
uint32_t underrunFrames = 0;
uint32_t servicedFrames = 0;
// NOTE: wasapi (others?) can call us back *after* stop()/Shutdown() (mState == SHUTDOWN)
// Bug 996162
// callback tells us cubeb succeeded initializing
if (mState == STARTED) {
mState = RUNNING;
}
if (available) {
if (mInRate == mOutRate) {
servicedFrames = GetUnprocessed(output, aFrames);
} else {
servicedFrames = GetTimeStretched(output, aFrames);
}
MOZ_ASSERT(mBuffer.Length() % mBytesPerFrame == 0, "Must copy complete frames");
// Notify any blocked Write() call that more space is available in mBuffer.
mon.NotifyAll();
}
underrunFrames = aFrames - servicedFrames;
// Always send audible frames first, and silent frames later.
// Otherwise it will break the assumption of FrameHistory.
if (mState != DRAINING) {
mAudioClock.UpdateFrameHistory(servicedFrames, underrunFrames);
uint8_t* rpos = static_cast<uint8_t*>(aBuffer) + FramesToBytes(aFrames - underrunFrames);
memset(rpos, 0, FramesToBytes(underrunFrames));
if (underrunFrames) {
MOZ_LOG(gAudioStreamLog, LogLevel::Warning,
("AudioStream %p lost %d frames", this, underrunFrames));
}
servicedFrames += underrunFrames;
} else {
mAudioClock.UpdateFrameHistory(servicedFrames, 0);
}
WriteDumpFile(mDumpFile, this, aFrames, aBuffer);
return servicedFrames;
}
void
AudioStream::StateCallback(cubeb_state aState)
{
MonitorAutoLock mon(mMonitor);
MOZ_ASSERT(mState != SHUTDOWN, "No state callback after shutdown");
LOG(("AudioStream: StateCallback %p, mState=%d cubeb_state=%d", this, mState, aState));
if (aState == CUBEB_STATE_DRAINED) {
mState = DRAINED;
} else if (aState == CUBEB_STATE_ERROR) {
LOG(("AudioStream::StateCallback() state %d cubeb error", mState));
mState = ERRORED;
}
mon.NotifyAll();
}
AudioClock::AudioClock(AudioStream* aStream)
:mAudioStream(aStream),
mOutRate(0),
mInRate(0),
mPreservesPitch(true),
mFrameHistory(new FrameHistory())
{}
void AudioClock::Init()
{
mOutRate = mAudioStream->GetRate();
mInRate = mAudioStream->GetRate();
}
void AudioClock::UpdateFrameHistory(uint32_t aServiced, uint32_t aUnderrun)
{
mFrameHistory->Append(aServiced, aUnderrun, mOutRate);
}
int64_t AudioClock::GetPositionUnlocked() const
{
// GetPositionInFramesUnlocked() asserts it owns the monitor
int64_t frames = mAudioStream->GetPositionInFramesUnlocked();
NS_ASSERTION(frames < 0 || (mInRate != 0 && mOutRate != 0), "AudioClock not initialized.");
return frames >= 0 ? mFrameHistory->GetPosition(frames) : -1;
}
int64_t AudioClock::GetPositionInFrames() const
{
return (GetPositionUnlocked() * mInRate) / USECS_PER_S;
}
void AudioClock::SetPlaybackRateUnlocked(double aPlaybackRate)
{
mOutRate = static_cast<int>(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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