AudioStream.cpp
/* -*- 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 "prlog.h"
#include "prdtoa.h"
#include "AudioStream.h"
#include "VideoUtils.h"
#include "mozilla/Monitor.h"
#include "mozilla/Mutex.h"
#include <algorithm>
#include "mozilla/Preferences.h"
#include "mozilla/Telemetry.h"
#include "soundtouch/SoundTouch.h"
#include "Latency.h"
namespace mozilla {
#ifdef LOG
#undef LOG
#endif
#ifdef PR_LOGGING
PRLogModuleInfo* gAudioStreamLog = nullptr;
// For simple logs
#define LOG(x) PR_LOG(gAudioStreamLog, PR_LOG_DEBUG, x)
#else
#define LOG(x)
#endif
/**
* 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;
#define PREF_VOLUME_SCALE "media.volume_scale"
#define PREF_CUBEB_LATENCY "media.cubeb_latency_ms"
static const uint32_t CUBEB_NORMAL_LATENCY_MS = 100;
StaticMutex AudioStream::sMutex;
cubeb* AudioStream::sCubebContext;
uint32_t AudioStream::sPreferredSampleRate;
double AudioStream::sVolumeScale;
uint32_t AudioStream::sCubebLatency;
bool AudioStream::sCubebLatencyPrefSet;
/*static*/ void AudioStream::PrefChanged(const char* aPref, void* aClosure)
{
if (strcmp(aPref, PREF_VOLUME_SCALE) == 0) {
nsAdoptingString value = Preferences::GetString(aPref);
StaticMutexAutoLock lock(sMutex);
if (value.IsEmpty()) {
sVolumeScale = 1.0;
} else {
NS_ConvertUTF16toUTF8 utf8(value);
sVolumeScale = std::max<double>(0, PR_strtod(utf8.get(), nullptr));
}
} else if (strcmp(aPref, PREF_CUBEB_LATENCY) == 0) {
// Arbitrary default stream latency of 100ms. The higher this
// value, the longer stream volume changes will take to become
// audible.
sCubebLatencyPrefSet = Preferences::HasUserValue(aPref);
uint32_t value = Preferences::GetUint(aPref, CUBEB_NORMAL_LATENCY_MS);
StaticMutexAutoLock lock(sMutex);
sCubebLatency = std::min<uint32_t>(std::max<uint32_t>(value, 1), 1000);
}
}
/*static*/ bool AudioStream::GetFirstStream()
{
static bool sFirstStream = true;
StaticMutexAutoLock lock(sMutex);
bool result = sFirstStream;
sFirstStream = false;
return result;
}
/*static*/ double AudioStream::GetVolumeScale()
{
StaticMutexAutoLock lock(sMutex);
return sVolumeScale;
}
/*static*/ cubeb* AudioStream::GetCubebContext()
{
StaticMutexAutoLock lock(sMutex);
return GetCubebContextUnlocked();
}
/*static*/ void AudioStream::InitPreferredSampleRate()
{
StaticMutexAutoLock lock(sMutex);
if (sPreferredSampleRate == 0 &&
cubeb_get_preferred_sample_rate(GetCubebContextUnlocked(),
&sPreferredSampleRate) != CUBEB_OK) {
sPreferredSampleRate = 44100;
}
}
/*static*/ cubeb* AudioStream::GetCubebContextUnlocked()
{
sMutex.AssertCurrentThreadOwns();
if (sCubebContext ||
cubeb_init(&sCubebContext, "AudioStream") == CUBEB_OK) {
return sCubebContext;
}
NS_WARNING("cubeb_init failed");
return nullptr;
}
/*static*/ uint32_t AudioStream::GetCubebLatency()
{
StaticMutexAutoLock lock(sMutex);
return sCubebLatency;
}
/*static*/ bool AudioStream::CubebLatencyPrefSet()
{
StaticMutexAutoLock lock(sMutex);
return sCubebLatencyPrefSet;
}
#if defined(__ANDROID__) && defined(MOZ_B2G)
static cubeb_stream_type ConvertChannelToCubebType(dom::AudioChannel aChannel)
{
switch(aChannel) {
case dom::AudioChannel::Normal:
return CUBEB_STREAM_TYPE_SYSTEM;
case dom::AudioChannel::Content:
return CUBEB_STREAM_TYPE_MUSIC;
case dom::AudioChannel::Notification:
return CUBEB_STREAM_TYPE_NOTIFICATION;
case dom::AudioChannel::Alarm:
return CUBEB_STREAM_TYPE_ALARM;
case dom::AudioChannel::Telephony:
return CUBEB_STREAM_TYPE_VOICE_CALL;
case dom::AudioChannel::Ringer:
return CUBEB_STREAM_TYPE_RING;
case dom::AudioChannel::Publicnotification:
return CUBEB_STREAM_TYPE_SYSTEM_ENFORCED;
default:
NS_ERROR("The value of AudioChannel is invalid");
return CUBEB_STREAM_TYPE_MAX;
}
}
#endif
AudioStream::AudioStream()
: mMonitor("AudioStream")
, mInRate(0)
, mOutRate(0)
, mChannels(0)
, mOutChannels(0)
, mWritten(0)
, mAudioClock(MOZ_THIS_IN_INITIALIZER_LIST())
, mLatencyRequest(HighLatency)
, mReadPoint(0)
, mWrittenFramesPast(0)
, mLostFramesPast(0)
, mWrittenFramesLast(0)
, mLostFramesLast(0)
, mDumpFile(nullptr)
, mVolume(1.0)
, mBytesPerFrame(0)
, mState(INITIALIZED)
, mNeedsStart(false)
{
// keep a ref in case we shut down later than nsLayoutStatics
mLatencyLog = AsyncLatencyLogger::Get(true);
}
AudioStream::~AudioStream()
{
LOG(("AudioStream: delete %p, state %d", this, mState));
Shutdown();
if (mDumpFile) {
fclose(mDumpFile);
}
}
size_t
AudioStream::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
{
size_t amount = aMallocSizeOf(this);
// Possibly add in the future:
// - mTimeStretcher
// - mLatencyLog
// - mCubebStream
amount += mInserts.SizeOfExcludingThis(aMallocSizeOf);
amount += mBuffer.SizeOfExcludingThis(aMallocSizeOf);
return amount;
}
/*static*/ void AudioStream::InitLibrary()
{
#ifdef PR_LOGGING
gAudioStreamLog = PR_NewLogModule("AudioStream");
#endif
PrefChanged(PREF_VOLUME_SCALE, nullptr);
Preferences::RegisterCallback(PrefChanged, PREF_VOLUME_SCALE);
PrefChanged(PREF_CUBEB_LATENCY, nullptr);
Preferences::RegisterCallback(PrefChanged, PREF_CUBEB_LATENCY);
}
/*static*/ void AudioStream::ShutdownLibrary()
{
Preferences::UnregisterCallback(PrefChanged, PREF_VOLUME_SCALE);
Preferences::UnregisterCallback(PrefChanged, PREF_CUBEB_LATENCY);
StaticMutexAutoLock lock(sMutex);
if (sCubebContext) {
cubeb_destroy(sCubebContext);
sCubebContext = nullptr;
}
}
nsresult AudioStream::EnsureTimeStretcherInitializedUnlocked()
{
mMonitor.AssertCurrentThreadOwns();
if (!mTimeStretcher) {
mTimeStretcher = new soundtouch::SoundTouch();
mTimeStretcher->setSampleRate(mInRate);
mTimeStretcher->setChannels(mOutChannels);
mTimeStretcher->setPitch(1.0);
}
return NS_OK;
}
nsresult AudioStream::SetPlaybackRate(double aPlaybackRate)
{
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;
}
// 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);
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)
{
// Avoid instantiating the timestretcher instance if not needed.
if (aPreservesPitch == mAudioClock.GetPreservesPitch()) {
return NS_OK;
}
// 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);
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()
{
return mWritten;
}
/*static*/ int AudioStream::MaxNumberOfChannels()
{
cubeb* cubebContext = GetCubebContext();
uint32_t maxNumberOfChannels;
if (cubebContext &&
cubeb_get_max_channel_count(cubebContext,
&maxNumberOfChannels) == CUBEB_OK) {
return static_cast<int>(maxNumberOfChannels);
}
return 0;
}
/*static*/ int AudioStream::PreferredSampleRate()
{
MOZ_ASSERT(sPreferredSampleRate,
"sPreferredSampleRate has not been initialized!");
return sPreferredSampleRate;
}
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];
sprintf(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);
}
// NOTE: this must not block a LowLatency stream for any significant amount
// of time, or it will block the entirety of MSG
nsresult
AudioStream::Init(int32_t aNumChannels, int32_t aRate,
const dom::AudioChannel aAudioChannel,
LatencyRequest aLatencyRequest)
{
mStartTime = TimeStamp::Now();
mIsFirst = GetFirstStream();
if (!GetCubebContext() || aNumChannels < 0 || aRate < 0) {
return NS_ERROR_FAILURE;
}
PR_LOG(gAudioStreamLog, PR_LOG_DEBUG,
("%s channels: %d, rate: %d for %p", __FUNCTION__, aNumChannels, aRate, this));
mInRate = mOutRate = aRate;
mChannels = aNumChannels;
mOutChannels = (aNumChannels > 2) ? 2 : aNumChannels;
mLatencyRequest = aLatencyRequest;
mDumpFile = OpenDumpFile(this);
cubeb_stream_params params;
params.rate = aRate;
params.channels = mOutChannels;
#if defined(__ANDROID__)
#if defined(MOZ_B2G)
params.stream_type = ConvertChannelToCubebType(aAudioChannel);
#else
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);
NS_ABORT_IF_FALSE(bufferLimit % mBytesPerFrame == 0, "Must buffer complete frames");
mBuffer.SetCapacity(bufferLimit);
if (aLatencyRequest == LowLatency) {
// Don't block this thread to initialize a cubeb stream.
// When this is done, it will start callbacks from Cubeb. Those will
// cause us to move from INITIALIZED to RUNNING. Until then, we
// can't access any cubeb functions.
// Use a RefPtr to avoid leaks if Dispatch fails
RefPtr<AudioInitTask> init = new AudioInitTask(this, aLatencyRequest, params);
init->Dispatch();
return NS_OK;
}
// High latency - open synchronously
nsresult rv = OpenCubeb(params, aLatencyRequest);
// See if we need to start() the stream, since we must do that from this
// thread for now (cubeb API issue)
CheckForStart();
return rv;
}
// 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,
LatencyRequest aLatencyRequest)
{
cubeb* cubebContext = GetCubebContext();
if (!cubebContext) {
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;
if (aLatencyRequest == LowLatency && !CubebLatencyPrefSet()) {
if (cubeb_get_min_latency(cubebContext, aParams, &latency) != CUBEB_OK) {
latency = GetCubebLatency();
}
} else {
latency = GetCubebLatency();
}
{
cubeb_stream* stream;
if (cubeb_stream_init(cubebContext, &stream, "AudioStream", aParams,
latency, DataCallback_S, StateCallback_S, this) == CUBEB_OK) {
MonitorAutoLock mon(mMonitor);
mCubebStream.own(stream);
// Make sure we weren't shut down while in flight!
if (mState == SHUTDOWN) {
mCubebStream.reset();
LOG(("AudioStream::OpenCubeb() %p Shutdown while opening cubeb", this));
return NS_ERROR_FAILURE;
}
// We can't cubeb_stream_start() the thread from a transient thread due to
// cubeb API requirements (init can be called from another thread, but
// not start/stop/destroy/etc)
} else {
MonitorAutoLock mon(mMonitor);
mState = ERRORED;
LOG(("AudioStream::OpenCubeb() %p failed to init cubeb", this));
return NS_ERROR_FAILURE;
}
}
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;
}
void
AudioStream::CheckForStart()
{
if (mState == INITIALIZED) {
// Start the stream right away when low latency has been requested. This means
// that the DataCallback will feed silence to cubeb, until the first frames
// are written to this AudioStream. Also start if a start has been queued.
if (mLatencyRequest == LowLatency || mNeedsStart) {
StartUnlocked(); // mState = STARTED or ERRORED
mNeedsStart = false;
PR_LOG(gAudioStreamLog, PR_LOG_WARNING,
("Started waiting %s-latency stream",
mLatencyRequest == LowLatency ? "low" : "high"));
} else {
// high latency, not full - OR Pause() was called before we got here
PR_LOG(gAudioStreamLog, PR_LOG_DEBUG,
("Not starting waiting %s-latency stream",
mLatencyRequest == LowLatency ? "low" : "high"));
}
}
}
NS_IMETHODIMP
AudioInitTask::Run()
{
MOZ_ASSERT(mThread);
if (NS_IsMainThread()) {
mThread->Shutdown(); // can't Shutdown from the thread itself, darn
// Don't null out mThread!
// See bug 999104. We must hold a ref to the thread across Dispatch()
// since the internal mThread ref could be released while processing
// the Dispatch(), and Dispatch/PutEvent itself doesn't hold a ref; it
// assumes the caller does.
return NS_OK;
}
nsresult rv = mAudioStream->OpenCubeb(mParams, mLatencyRequest);
// and now kill this thread
NS_DispatchToMainThread(this);
return rv;
}
// aTime is the time in ms the samples were inserted into MediaStreamGraph
nsresult
AudioStream::Write(const AudioDataValue* aBuf, uint32_t aFrames, TimeStamp *aTime)
{
MonitorAutoLock mon(mMonitor);
if (mState == ERRORED) {
return NS_ERROR_FAILURE;
}
NS_ASSERTION(mState == INITIALIZED || mState == STARTED || mState == RUNNING,
"Stream write in unexpected state.");
// See if we need to start() the stream, since we must do that from this thread
CheckForStart();
// Downmix to Stereo.
if (mChannels > 2 && mChannels <= 8) {
DownmixAudioToStereo(const_cast<AudioDataValue*> (aBuf), mChannels, aFrames);
}
else if (mChannels > 8) {
return NS_ERROR_FAILURE;
}
const uint8_t* src = reinterpret_cast<const uint8_t*>(aBuf);
uint32_t bytesToCopy = FramesToBytes(aFrames);
// XXX this will need to change if we want to enable this on-the-fly!
if (PR_LOG_TEST(GetLatencyLog(), PR_LOG_DEBUG)) {
// Record the position and time this data was inserted
int64_t timeMs;
if (aTime && !aTime->IsNull()) {
if (mStartTime.IsNull()) {
AsyncLatencyLogger::Get(true)->GetStartTime(mStartTime);
}
timeMs = (*aTime - mStartTime).ToMilliseconds();
} else {
timeMs = 0;
}
struct Inserts insert = { timeMs, aFrames};
mInserts.AppendElement(insert);
}
while (bytesToCopy > 0) {
uint32_t available = std::min(bytesToCopy, mBuffer.Available());
NS_ABORT_IF_FALSE(available % mBytesPerFrame == 0,
"Must copy complete frames.");
mBuffer.AppendElements(src, available);
src += available;
bytesToCopy -= available;
if (bytesToCopy > 0) {
// Careful - the CubebInit thread may not have gotten to STARTED yet
if ((mState == INITIALIZED || mState == STARTED) && mLatencyRequest == LowLatency) {
// don't ever block MediaStreamGraph low-latency streams
uint32_t remains = 0; // we presume the buffer is full
if (mBuffer.Length() > bytesToCopy) {
remains = mBuffer.Length() - bytesToCopy; // Free up just enough space
}
// account for dropping samples
PR_LOG(gAudioStreamLog, PR_LOG_WARNING, ("Stream %p dropping %u bytes (%u frames)in Write()",
this, mBuffer.Length() - remains, BytesToFrames(mBuffer.Length() - remains)));
mReadPoint += BytesToFrames(mBuffer.Length() - remains);
mBuffer.ContractTo(remains);
} else { // RUNNING or high latency
// 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) {
PR_LOG(gAudioStreamLog, PR_LOG_WARNING, ("Starting stream %p in Write (%u waiting)",
this, bytesToCopy));
StartUnlocked();
if (mState == ERRORED) {
return NS_ERROR_FAILURE;
}
}
PR_LOG(gAudioStreamLog, PR_LOG_WARNING, ("Stream %p waiting in Write() (%u waiting)",
this, bytesToCopy));
mon.Wait();
}
}
}
mWritten += aFrames;
return NS_OK;
}
uint32_t
AudioStream::Available()
{
MonitorAutoLock mon(mMonitor);
NS_ABORT_IF_FALSE(mBuffer.Length() % mBytesPerFrame == 0, "Buffer invariant violated.");
return BytesToFrames(mBuffer.Available());
}
void
AudioStream::SetVolume(double aVolume)
{
MonitorAutoLock mon(mMonitor);
NS_ABORT_IF_FALSE(aVolume >= 0.0 && aVolume <= 1.0, "Invalid volume");
mVolume = aVolume;
}
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) {
mNeedsStart = true;
return;
}
MonitorAutoUnlock mon(mMonitor);
if (mState == INITIALIZED) {
int r = cubeb_stream_start(mCubebStream);
mState = r == CUBEB_OK ? STARTED : ERRORED;
LOG(("AudioStream: started %p, state %s", this, mState == STARTED ? "STARTED" : "ERRORED"));
}
}
void
AudioStream::Pause()
{
MonitorAutoLock mon(mMonitor);
if (!mCubebStream || (mState != STARTED && mState != RUNNING)) {
mNeedsStart = false;
mState = STOPPED; // which also tells async OpenCubeb not to start, just init
return;
}
int r;
{
MonitorAutoUnlock mon(mMonitor);
r = cubeb_stream_stop(mCubebStream);
}
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);
}
if (mState != ERRORED && r == CUBEB_OK) {
mState = STARTED;
}
}
void
AudioStream::Shutdown()
{
LOG(("AudioStream: Shutdown %p, state %d", this, mState));
{
MonitorAutoLock mon(mMonitor);
if (mState == STARTED || mState == RUNNING) {
MonitorAutoUnlock mon(mMonitor);
Pause();
}
MOZ_ASSERT(mState != STARTED && mState != RUNNING); // paranoia
mState = SHUTDOWN;
}
// Must not try to shut down cubeb from within the lock! wasapi may still
// call our callback after Pause()/stop()!?! Bug 996162
if (mCubebStream) {
mCubebStream.reset();
}
}
int64_t
AudioStream::GetPosition()
{
MonitorAutoLock mon(mMonitor);
return mAudioClock.GetPositionUnlocked();
}
// This function is miscompiled by PGO with MSVC 2010. See bug 768333.
#ifdef _MSC_VER
#pragma optimize("", off)
#endif
int64_t
AudioStream::GetPositionInFrames()
{
return mAudioClock.GetPositionInFrames();
}
#ifdef _MSC_VER
#pragma optimize("", on)
#endif
int64_t
AudioStream::GetPositionInFramesInternal()
{
MonitorAutoLock mon(mMonitor);
return GetPositionInFramesUnlocked();
}
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, &position) != CUBEB_OK) {
return -1;
}
}
// Adjust the reported position by the number of silent frames written
// during stream underruns.
// Since frames sent to DataCallback is not consumed by the backend immediately,
// it will be an over adjustment if we return |position - mLostFramesPast - mLostFramesLast|.
// On the other hand, we need to keep the whole history of frames sent to DataCallback
// in order to adjust position correctly which will require more storage.
// We choose a simple way to store the history where |mWrittenFramesPast| and
// |mLostFramesPast| are the sum of frames from 1th to |N-1|th callbacks, and
// |mWrittenFramesLast| and |mLostFramesLast| represent the frames sent in last callback.
// When |position| lies in
// [mWrittenFramesPast+mLostFramesPast, mWrittenFramesPast+mLostFramesPast+mWrittenFramesLast+mLostFramesLast],
// we will be able to adjust position precisely which should be the major case.
// If |position| falls in [0, mWrittenFramesPast+mLostFramesPast), there will be an
// error in the adjustment. However that is fine as long as we can ensure the
// adjusted position is mono-increasing to avoid audio clock going backward.
uint64_t adjustedPosition = 0;
if (position <= mWrittenFramesPast) {
adjustedPosition = position;
} else if (position <= mWrittenFramesPast + mLostFramesPast) {
adjustedPosition = mWrittenFramesPast;
} else if (position <= mWrittenFramesPast + mLostFramesPast + mWrittenFramesLast) {
adjustedPosition = position - mLostFramesPast;
} else {
adjustedPosition = mWrittenFramesPast + mWrittenFramesLast;
}
return std::min<uint64_t>(adjustedPosition, INT64_MAX);
}
int64_t
AudioStream::GetLatencyInFrames()
{
uint32_t latency;
if (cubeb_stream_get_latency(mCubebStream, &latency)) {
NS_WARNING("Could not get cubeb latency.");
return 0;
}
return static_cast<int64_t>(latency);
}
bool
AudioStream::IsPaused()
{
MonitorAutoLock mon(mMonitor);
return mState == STOPPED;
}
void
AudioStream::GetBufferInsertTime(int64_t &aTimeMs)
{
if (mInserts.Length() > 0) {
// Find the right block, but don't leave the array empty
while (mInserts.Length() > 1 && mReadPoint >= mInserts[0].mFrames) {
mReadPoint -= mInserts[0].mFrames;
mInserts.RemoveElementAt(0);
}
// offset for amount already read
// XXX Note: could misreport if we couldn't find a block in the right timeframe
aTimeMs = mInserts[0].mTimeMs + ((mReadPoint * 1000) / mOutRate);
} else {
aTimeMs = INT64_MAX;
}
}
long
AudioStream::GetUnprocessed(void* aBuffer, long aFrames, int64_t &aTimeMs)
{
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]);
// First time block now has our first returned sample
mReadPoint += BytesToFrames(available);
GetBufferInsertTime(aTimeMs);
return BytesToFrames(available) + flushedFrames;
}
// Get unprocessed samples, and pad the beginning of the buffer with silence if
// there is not enough data.
long
AudioStream::GetUnprocessedWithSilencePadding(void* aBuffer, long aFrames, int64_t& aTimeMs)
{
uint32_t toPopBytes = FramesToBytes(aFrames);
uint32_t available = std::min(toPopBytes, mBuffer.Length());
uint32_t silenceOffset = toPopBytes - available;
uint8_t* wpos = reinterpret_cast<uint8_t*>(aBuffer);
memset(wpos, 0, silenceOffset);
wpos += silenceOffset;
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]);
GetBufferInsertTime(aTimeMs);
return aFrames;
}
long
AudioStream::GetTimeStretched(void* aBuffer, long aFrames, int64_t &aTimeMs)
{
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]);
mReadPoint += BytesToFrames(available);
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);
GetBufferInsertTime(aTimeMs);
return processedFrames;
}
long
AudioStream::DataCallback(void* aBuffer, long aFrames)
{
MonitorAutoLock mon(mMonitor);
uint32_t available = std::min(static_cast<uint32_t>(FramesToBytes(aFrames)), mBuffer.Length());
NS_ABORT_IF_FALSE(available % mBytesPerFrame == 0, "Must copy complete frames");
AudioDataValue* output = reinterpret_cast<AudioDataValue*>(aBuffer);
uint32_t underrunFrames = 0;
uint32_t servicedFrames = 0;
int64_t insertTime;
mWrittenFramesPast += mWrittenFramesLast;
mLostFramesPast += mLostFramesLast;
// NOTE: wasapi (others?) can call us back *after* stop()/Shutdown() (mState == SHUTDOWN)
// Bug 996162
// callback tells us cubeb succeeded initializing
if (mState == STARTED) {
// For low-latency streams, we want to minimize any built-up data when
// we start getting callbacks.
// Simple version - contract on first callback only.
if (mLatencyRequest == LowLatency) {
#ifdef PR_LOGGING
uint32_t old_len = mBuffer.Length();
#endif
available = mBuffer.ContractTo(FramesToBytes(aFrames));
#ifdef PR_LOGGING
TimeStamp now = TimeStamp::Now();
if (!mStartTime.IsNull()) {
int64_t timeMs = (now - mStartTime).ToMilliseconds();
PR_LOG(gAudioStreamLog, PR_LOG_WARNING,
("Stream took %lldms to start after first Write() @ %u", timeMs, mOutRate));
} else {
PR_LOG(gAudioStreamLog, PR_LOG_WARNING,
("Stream started before Write() @ %u", mOutRate));
}
if (old_len != available) {
// Note that we may have dropped samples in Write() as well!
PR_LOG(gAudioStreamLog, PR_LOG_WARNING,
("AudioStream %p dropped %u + %u initial frames @ %u", this,
mReadPoint, BytesToFrames(old_len - available), mOutRate));
mReadPoint += BytesToFrames(old_len - available);
}
#endif
}
mState = RUNNING;
}
if (available) {
// When we are playing a low latency stream, and it is the first time we are
// getting data from the buffer, we prefer to add the silence for an
// underrun at the beginning of the buffer, so the first buffer is not cut
// in half by the silence inserted to compensate for the underrun.
if (mInRate == mOutRate) {
if (mLatencyRequest == LowLatency && !mWritten) {
servicedFrames = GetUnprocessedWithSilencePadding(output, aFrames, insertTime);
} else {
servicedFrames = GetUnprocessed(output, aFrames, insertTime);
}
} else {
servicedFrames = GetTimeStretched(output, aFrames, insertTime);
}
float scaled_volume = float(GetVolumeScale() * mVolume);
ScaleAudioSamples(output, aFrames * mOutChannels, scaled_volume);
NS_ABORT_IF_FALSE(mBuffer.Length() % mBytesPerFrame == 0, "Must copy complete frames");
// Notify any blocked Write() call that more space is available in mBuffer.
mon.NotifyAll();
} else {
GetBufferInsertTime(insertTime);
}
underrunFrames = aFrames - servicedFrames;
mWrittenFramesLast = servicedFrames;
mLostFramesLast = underrunFrames;
if (mState != DRAINING) {
uint8_t* rpos = static_cast<uint8_t*>(aBuffer) + FramesToBytes(aFrames - underrunFrames);
memset(rpos, 0, FramesToBytes(underrunFrames));
if (underrunFrames) {
PR_LOG(gAudioStreamLog, PR_LOG_WARNING,
("AudioStream %p lost %d frames", this, underrunFrames));
}
servicedFrames += underrunFrames;
}
WriteDumpFile(mDumpFile, this, aFrames, aBuffer);
// Don't log if we're not interested or if the stream is inactive
if (PR_LOG_TEST(GetLatencyLog(), PR_LOG_DEBUG) &&
mState != SHUTDOWN &&
insertTime != INT64_MAX && servicedFrames > underrunFrames) {
uint32_t latency = UINT32_MAX;
if (cubeb_stream_get_latency(mCubebStream, &latency)) {
NS_WARNING("Could not get latency from cubeb.");
}
TimeStamp now = TimeStamp::Now();
mLatencyLog->Log(AsyncLatencyLogger::AudioStream, reinterpret_cast<uint64_t>(this),
insertTime, now);
mLatencyLog->Log(AsyncLatencyLogger::Cubeb, reinterpret_cast<uint64_t>(mCubebStream.get()),
(latency * 1000) / mOutRate, now);
}
mAudioClock.UpdateWritePosition(servicedFrames);
return servicedFrames;
}
void
AudioStream::StateCallback(cubeb_state aState)
{
MonitorAutoLock mon(mMonitor);
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),
mOldOutRate(0),
mBasePosition(0),
mBaseOffset(0),
mOldBaseOffset(0),
mOldBasePosition(0),
mPlaybackRateChangeOffset(0),
mPreviousPosition(0),
mWritten(0),
mOutRate(0),
mInRate(0),
mPreservesPitch(true),
mCompensatingLatency(false)
{}
void AudioClock::Init()
{
mOutRate = mAudioStream->GetRate();
mInRate = mAudioStream->GetRate();
mOldOutRate = mOutRate;
}
void AudioClock::UpdateWritePosition(uint32_t aCount)
{
mWritten += aCount;
}
uint64_t AudioClock::GetPositionUnlocked()
{
// GetPositionInFramesUnlocked() asserts it owns the monitor
int64_t position = mAudioStream->GetPositionInFramesUnlocked();
int64_t diffOffset;
NS_ASSERTION(position < 0 || (mInRate != 0 && mOutRate != 0), "AudioClock not initialized.");
if (position >= 0) {
if (position < mPlaybackRateChangeOffset) {
// See if we are still playing frames pushed with the old playback rate in
// the backend. If we are, use the old output rate to compute the
// position.
mCompensatingLatency = true;
diffOffset = position - mOldBaseOffset;
position = static_cast<uint64_t>(mOldBasePosition +
static_cast<float>(USECS_PER_S * diffOffset) / mOldOutRate);
mPreviousPosition = position;
return position;
}
if (mCompensatingLatency) {
diffOffset = position - mPlaybackRateChangeOffset;
mCompensatingLatency = false;
mBasePosition = mPreviousPosition;
} else {
diffOffset = position - mPlaybackRateChangeOffset;
}
position = static_cast<uint64_t>(mBasePosition +
(static_cast<float>(USECS_PER_S * diffOffset) / mOutRate));
return position;
}
return UINT64_MAX;
}
uint64_t AudioClock::GetPositionInFrames()
{
return (GetPositionUnlocked() * mOutRate) / USECS_PER_S;
}
void AudioClock::SetPlaybackRateUnlocked(double aPlaybackRate)
{
// GetPositionInFramesUnlocked() asserts it owns the monitor
int64_t position = mAudioStream->GetPositionInFramesUnlocked();
if (position > mPlaybackRateChangeOffset) {
mOldBasePosition = mBasePosition;
mBasePosition = GetPositionUnlocked();
mOldBaseOffset = mPlaybackRateChangeOffset;
mBaseOffset = position;
mPlaybackRateChangeOffset = mWritten;
mOldOutRate = mOutRate;
mOutRate = static_cast<int>(mInRate / aPlaybackRate);
} else {
// The playbackRate has been changed before the end of the latency
// compensation phase. We don't update the mOld* variable. That way, the
// last playbackRate set is taken into account.
mBasePosition = GetPositionUnlocked();
mBaseOffset = position;
mPlaybackRateChangeOffset = mWritten;
mOutRate = static_cast<int>(mInRate / aPlaybackRate);
}
}
double AudioClock::GetPlaybackRate()
{
return static_cast<double>(mInRate) / mOutRate;
}
void AudioClock::SetPreservesPitch(bool aPreservesPitch)
{
mPreservesPitch = aPreservesPitch;
}
bool AudioClock::GetPreservesPitch()
{
return mPreservesPitch;
}
} // namespace mozilla
