gecko-dev/content/media/AudioStream.cpp

1032 lines
28 KiB
C++

/* -*- 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 "soundtouch/SoundTouch.h"
#include "Latency.h"
#if defined(MOZ_CUBEB)
#include "nsAutoRef.h"
#include "cubeb/cubeb.h"
template <>
class nsAutoRefTraits<cubeb_stream> : public nsPointerRefTraits<cubeb_stream>
{
public:
static void Release(cubeb_stream* aStream) { cubeb_stream_destroy(aStream); }
};
#endif
namespace mozilla {
#ifdef PR_LOGGING
PRLogModuleInfo* gAudioStreamLog = nullptr;
#endif
#define PREF_VOLUME_SCALE "media.volume_scale"
#define PREF_CUBEB_LATENCY "media.cubeb_latency_ms"
static Mutex* gAudioPrefsLock = nullptr;
static double gVolumeScale;
static uint32_t gCubebLatency;
/**
* 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 nsBufferedAudioStream created, containing
* the audio for the stream including any skips due to underruns.
*/
static int gDumpedAudioCount = 0;
static int PrefChanged(const char* aPref, void* aClosure)
{
if (strcmp(aPref, PREF_VOLUME_SCALE) == 0) {
nsAdoptingString value = Preferences::GetString(aPref);
MutexAutoLock lock(*gAudioPrefsLock);
if (value.IsEmpty()) {
gVolumeScale = 1.0;
} else {
NS_ConvertUTF16toUTF8 utf8(value);
gVolumeScale = 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.
uint32_t value = Preferences::GetUint(aPref, 100);
MutexAutoLock lock(*gAudioPrefsLock);
gCubebLatency = std::min<uint32_t>(std::max<uint32_t>(value, 20), 1000);
}
return 0;
}
static double GetVolumeScale()
{
MutexAutoLock lock(*gAudioPrefsLock);
return gVolumeScale;
}
#if defined(MOZ_CUBEB)
static cubeb* gCubebContext;
static cubeb* GetCubebContext()
{
MutexAutoLock lock(*gAudioPrefsLock);
if (gCubebContext ||
cubeb_init(&gCubebContext, "AudioStream") == CUBEB_OK) {
return gCubebContext;
}
NS_WARNING("cubeb_init failed");
return nullptr;
}
static uint32_t GetCubebLatency()
{
MutexAutoLock lock(*gAudioPrefsLock);
return gCubebLatency;
}
#endif
#if defined(MOZ_CUBEB) && defined(__ANDROID__) && defined(MOZ_B2G)
static cubeb_stream_type ConvertChannelToCubebType(dom::AudioChannelType aType)
{
switch(aType) {
case dom::AUDIO_CHANNEL_NORMAL:
return CUBEB_STREAM_TYPE_SYSTEM;
case dom::AUDIO_CHANNEL_CONTENT:
return CUBEB_STREAM_TYPE_MUSIC;
case dom::AUDIO_CHANNEL_NOTIFICATION:
return CUBEB_STREAM_TYPE_NOTIFICATION;
case dom::AUDIO_CHANNEL_ALARM:
return CUBEB_STREAM_TYPE_ALARM;
case dom::AUDIO_CHANNEL_TELEPHONY:
return CUBEB_STREAM_TYPE_VOICE_CALL;
case dom::AUDIO_CHANNEL_RINGER:
return CUBEB_STREAM_TYPE_RING;
// Currently Android openSLES library doesn't support FORCE_AUDIBLE yet.
case dom::AUDIO_CHANNEL_PUBLICNOTIFICATION:
default:
NS_ERROR("The value of AudioChannelType is invalid");
return CUBEB_STREAM_TYPE_MAX;
}
}
#endif
AudioStream::AudioStream()
: mInRate(0),
mOutRate(0),
mChannels(0),
mWritten(0),
mAudioClock(MOZ_THIS_IN_INITIALIZER_LIST())
{}
void AudioStream::InitLibrary()
{
#ifdef PR_LOGGING
gAudioStreamLog = PR_NewLogModule("AudioStream");
#endif
gAudioPrefsLock = new Mutex("AudioStream::gAudioPrefsLock");
PrefChanged(PREF_VOLUME_SCALE, nullptr);
Preferences::RegisterCallback(PrefChanged, PREF_VOLUME_SCALE);
#if defined(MOZ_CUBEB)
PrefChanged(PREF_CUBEB_LATENCY, nullptr);
Preferences::RegisterCallback(PrefChanged, PREF_CUBEB_LATENCY);
#endif
}
void AudioStream::ShutdownLibrary()
{
Preferences::UnregisterCallback(PrefChanged, PREF_VOLUME_SCALE);
#if defined(MOZ_CUBEB)
Preferences::UnregisterCallback(PrefChanged, PREF_CUBEB_LATENCY);
#endif
delete gAudioPrefsLock;
gAudioPrefsLock = nullptr;
#if defined(MOZ_CUBEB)
if (gCubebContext) {
cubeb_destroy(gCubebContext);
gCubebContext = nullptr;
}
#endif
}
AudioStream::~AudioStream()
{
}
nsresult AudioStream::EnsureTimeStretcherInitialized()
{
if (!mTimeStretcher) {
// SoundTouch does not support a number of channels > 2
if (mChannels > 2) {
return NS_ERROR_FAILURE;
}
mTimeStretcher = new soundtouch::SoundTouch();
mTimeStretcher->setSampleRate(mInRate);
mTimeStretcher->setChannels(mChannels);
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.
if (aPlaybackRate == mAudioClock.GetPlaybackRate()) {
return NS_OK;
}
if (EnsureTimeStretcherInitialized() != NS_OK) {
return NS_ERROR_FAILURE;
}
mAudioClock.SetPlaybackRate(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;
}
if (EnsureTimeStretcherInitialized() != 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;
}
#if defined(MOZ_CUBEB)
class nsCircularByteBuffer
{
public:
nsCircularByteBuffer()
: mBuffer(nullptr), mCapacity(0), mStart(0), mCount(0)
{}
// Set the capacity of the buffer in bytes. Must be called before any
// call to append or pop elements.
void SetCapacity(uint32_t aCapacity) {
NS_ABORT_IF_FALSE(!mBuffer, "Buffer allocated.");
mCapacity = aCapacity;
mBuffer = new uint8_t[mCapacity];
}
uint32_t Length() {
return mCount;
}
uint32_t Capacity() {
return mCapacity;
}
uint32_t Available() {
return Capacity() - Length();
}
// Append aLength bytes from aSrc to the buffer. Caller must check that
// sufficient space is available.
void AppendElements(const uint8_t* aSrc, uint32_t aLength) {
NS_ABORT_IF_FALSE(mBuffer && mCapacity, "Buffer not initialized.");
NS_ABORT_IF_FALSE(aLength <= Available(), "Buffer full.");
uint32_t end = (mStart + mCount) % mCapacity;
uint32_t toCopy = std::min(mCapacity - end, aLength);
memcpy(&mBuffer[end], aSrc, toCopy);
memcpy(&mBuffer[0], aSrc + toCopy, aLength - toCopy);
mCount += aLength;
}
// Remove aSize bytes from the buffer. Caller must check returned size in
// aSize{1,2} before using the pointer returned in aData{1,2}. Caller
// must not specify an aSize larger than Length().
void PopElements(uint32_t aSize, void** aData1, uint32_t* aSize1,
void** aData2, uint32_t* aSize2) {
NS_ABORT_IF_FALSE(mBuffer && mCapacity, "Buffer not initialized.");
NS_ABORT_IF_FALSE(aSize <= Length(), "Request too large.");
*aData1 = &mBuffer[mStart];
*aSize1 = std::min(mCapacity - mStart, aSize);
*aData2 = &mBuffer[0];
*aSize2 = aSize - *aSize1;
mCount -= *aSize1 + *aSize2;
mStart += *aSize1 + *aSize2;
mStart %= mCapacity;
}
private:
nsAutoArrayPtr<uint8_t> mBuffer;
uint32_t mCapacity;
uint32_t mStart;
uint32_t mCount;
};
class BufferedAudioStream : public AudioStream
{
public:
BufferedAudioStream();
~BufferedAudioStream();
nsresult Init(int32_t aNumChannels, int32_t aRate,
const dom::AudioChannelType aAudioChannelType);
void Shutdown();
nsresult Write(const AudioDataValue* aBuf, uint32_t aFrames);
uint32_t Available();
void SetVolume(double aVolume);
void Drain();
void Start();
void Pause();
void Resume();
int64_t GetPosition();
int64_t GetPositionInFrames();
int64_t GetPositionInFramesInternal();
int64_t GetLatencyInFrames();
bool IsPaused();
// This method acquires the monitor and forward the call to the base
// class, to prevent a race on |mTimeStretcher|, in
// |AudioStream::EnsureTimeStretcherInitialized|.
nsresult EnsureTimeStretcherInitialized();
private:
static long DataCallback_S(cubeb_stream*, void* aThis, void* aBuffer, long aFrames)
{
return static_cast<BufferedAudioStream*>(aThis)->DataCallback(aBuffer, aFrames);
}
static void StateCallback_S(cubeb_stream*, void* aThis, cubeb_state aState)
{
static_cast<BufferedAudioStream*>(aThis)->StateCallback(aState);
}
long DataCallback(void* aBuffer, long aFrames);
void StateCallback(cubeb_state aState);
long GetUnprocessed(void* aBuffer, long aFrames);
long GetTimeStretched(void* aBuffer, long aFrames);
// Shared implementation of underflow adjusted position calculation.
// Caller must own the monitor.
int64_t GetPositionInFramesUnlocked();
void StartUnlocked();
// The monitor is held to protect all access to member variables. Write()
// waits while mBuffer is full; DataCallback() notifies as it consumes
// data from mBuffer. Drain() waits while mState is DRAINING;
// StateCallback() notifies when mState is DRAINED.
Monitor mMonitor;
// Sum of silent frames written when DataCallback requests more frames
// than are available in mBuffer.
uint64_t mLostFrames;
// Output file for dumping audio
FILE* mDumpFile;
// Temporary audio buffer. Filled by Write() and consumed by
// DataCallback(). Once mBuffer is full, Write() blocks until sufficient
// space becomes available in mBuffer. mBuffer is sized in bytes, not
// frames.
nsCircularByteBuffer mBuffer;
// Software volume level. Applied during the servicing of DataCallback().
double mVolume;
// Owning reference to a cubeb_stream. cubeb_stream_destroy is called by
// nsAutoRef's destructor.
nsAutoRef<cubeb_stream> mCubebStream;
uint32_t mBytesPerFrame;
uint32_t BytesToFrames(uint32_t aBytes) {
NS_ASSERTION(aBytes % mBytesPerFrame == 0,
"Byte count not aligned on frames size.");
return aBytes / mBytesPerFrame;
}
uint32_t FramesToBytes(uint32_t aFrames) {
return aFrames * mBytesPerFrame;
}
enum StreamState {
INITIALIZED, // Initialized, playback has not begun.
STARTED, // Started by a call to Write() (iff INITIALIZED) or Resume().
STOPPED, // Stopped by a call to Pause().
DRAINING, // Drain requested. DataCallback will indicate end of stream
// once the remaining contents of mBuffer are requested by
// cubeb, after which StateCallback will indicate drain
// completion.
DRAINED, // StateCallback has indicated that the drain is complete.
ERRORED // Stream disabled due to an internal error.
};
StreamState mState;
};
#endif
AudioStream* AudioStream::AllocateStream()
{
#if defined(MOZ_CUBEB)
return new BufferedAudioStream();
#endif
return nullptr;
}
int AudioStream::MaxNumberOfChannels()
{
uint32_t maxNumberOfChannels, rv;
rv = cubeb_get_max_channel_count(GetCubebContext(), &maxNumberOfChannels);
if (rv != CUBEB_OK) {
return 0;
}
return static_cast<int>(maxNumberOfChannels);
}
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->GetChannels()*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);
}
#if defined(MOZ_CUBEB)
BufferedAudioStream::BufferedAudioStream()
: mMonitor("BufferedAudioStream"), mLostFrames(0), mDumpFile(nullptr),
mVolume(1.0), mBytesPerFrame(0), mState(INITIALIZED)
{
AsyncLatencyLogger::Get(true)->AddRef();
}
BufferedAudioStream::~BufferedAudioStream()
{
Shutdown();
if (mDumpFile) {
fclose(mDumpFile);
}
AsyncLatencyLogger::Get()->Release();
}
nsresult
BufferedAudioStream::EnsureTimeStretcherInitialized()
{
MonitorAutoLock mon(mMonitor);
return AudioStream::EnsureTimeStretcherInitialized();
}
nsresult
BufferedAudioStream::Init(int32_t aNumChannels, int32_t aRate,
const dom::AudioChannelType aAudioChannelType)
{
cubeb* cubebContext = GetCubebContext();
if (!cubebContext || aNumChannels < 0 || aRate < 0) {
return NS_ERROR_FAILURE;
}
mInRate = mOutRate = aRate;
mChannels = aNumChannels;
mDumpFile = OpenDumpFile(this);
cubeb_stream_params params;
params.rate = aRate;
params.channels = aNumChannels;
#if defined(__ANDROID__)
#if defined(MOZ_B2G)
params.stream_type = ConvertChannelToCubebType(aAudioChannelType);
#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) * aNumChannels;
mAudioClock.Init();
{
cubeb_stream* stream;
if (cubeb_stream_init(cubebContext, &stream, "BufferedAudioStream", params,
GetCubebLatency(), DataCallback_S, StateCallback_S, this) == CUBEB_OK) {
mCubebStream.own(stream);
}
}
if (!mCubebStream) {
return NS_ERROR_FAILURE;
}
// 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);
return NS_OK;
}
void
BufferedAudioStream::Shutdown()
{
if (mState == STARTED) {
Pause();
}
if (mCubebStream) {
mCubebStream.reset();
}
}
nsresult
BufferedAudioStream::Write(const AudioDataValue* aBuf, uint32_t aFrames)
{
MonitorAutoLock mon(mMonitor);
if (!mCubebStream || mState == ERRORED) {
return NS_ERROR_FAILURE;
}
NS_ASSERTION(mState == INITIALIZED || mState == STARTED,
"Stream write in unexpected state.");
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());
NS_ABORT_IF_FALSE(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) {
StartUnlocked();
if (mState != STARTED) {
return NS_ERROR_FAILURE;
}
}
mon.Wait();
}
}
mWritten += aFrames;
return NS_OK;
}
uint32_t
BufferedAudioStream::Available()
{
MonitorAutoLock mon(mMonitor);
NS_ABORT_IF_FALSE(mBuffer.Length() % mBytesPerFrame == 0, "Buffer invariant violated.");
return BytesToFrames(mBuffer.Available());
}
void
BufferedAudioStream::SetVolume(double aVolume)
{
MonitorAutoLock mon(mMonitor);
NS_ABORT_IF_FALSE(aVolume >= 0.0 && aVolume <= 1.0, "Invalid volume");
mVolume = aVolume;
}
void
BufferedAudioStream::Drain()
{
MonitorAutoLock mon(mMonitor);
if (mState != STARTED) {
NS_ASSERTION(mBuffer.Available() == 0, "Draining with unplayed audio");
return;
}
mState = DRAINING;
while (mState == DRAINING) {
mon.Wait();
}
}
void
BufferedAudioStream::Start()
{
MonitorAutoLock mon(mMonitor);
StartUnlocked();
}
void
BufferedAudioStream::StartUnlocked()
{
mMonitor.AssertCurrentThreadOwns();
if (!mCubebStream || mState != INITIALIZED) {
return;
}
if (mState != STARTED) {
int r;
{
MonitorAutoUnlock mon(mMonitor);
r = cubeb_stream_start(mCubebStream);
}
if (mState != ERRORED) {
mState = r == CUBEB_OK ? STARTED : ERRORED;
}
}
}
void
BufferedAudioStream::Pause()
{
MonitorAutoLock mon(mMonitor);
if (!mCubebStream || mState != STARTED) {
return;
}
int r;
{
MonitorAutoUnlock mon(mMonitor);
r = cubeb_stream_stop(mCubebStream);
}
if (mState != ERRORED && r == CUBEB_OK) {
mState = STOPPED;
}
}
void
BufferedAudioStream::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;
}
}
int64_t
BufferedAudioStream::GetPosition()
{
return mAudioClock.GetPosition();
}
// This function is miscompiled by PGO with MSVC 2010. See bug 768333.
#ifdef _MSC_VER
#pragma optimize("", off)
#endif
int64_t
BufferedAudioStream::GetPositionInFrames()
{
return mAudioClock.GetPositionInFrames();
}
#ifdef _MSC_VER
#pragma optimize("", on)
#endif
int64_t
BufferedAudioStream::GetPositionInFramesInternal()
{
MonitorAutoLock mon(mMonitor);
return GetPositionInFramesUnlocked();
}
int64_t
BufferedAudioStream::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.
uint64_t adjustedPosition = 0;
if (position >= mLostFrames) {
adjustedPosition = position - mLostFrames;
}
return std::min<uint64_t>(adjustedPosition, INT64_MAX);
}
int64_t
BufferedAudioStream::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
BufferedAudioStream::IsPaused()
{
MonitorAutoLock mon(mMonitor);
return mState == STOPPED;
}
long
BufferedAudioStream::GetUnprocessed(void* aBuffer, long aFrames)
{
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
BufferedAudioStream::GetTimeStretched(void* aBuffer, long aFrames)
{
long processedFrames = 0;
// We need to call the non-locking version, because we already have the lock.
if (AudioStream::EnsureTimeStretcherInitialized() != 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
BufferedAudioStream::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");
uint32_t underrunFrames = 0;
uint32_t servicedFrames = 0;
if (available) {
AudioDataValue* output = reinterpret_cast<AudioDataValue*>(aBuffer);
if (mInRate == mOutRate) {
servicedFrames = GetUnprocessed(output, aFrames);
} else {
servicedFrames = GetTimeStretched(output, aFrames);
}
float scaled_volume = float(GetVolumeScale() * mVolume);
ScaleAudioSamples(output, aFrames * mChannels, 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();
}
underrunFrames = aFrames - servicedFrames;
if (mState != DRAINING) {
uint8_t* rpos = static_cast<uint8_t*>(aBuffer) + FramesToBytes(aFrames - underrunFrames);
memset(rpos, 0, FramesToBytes(underrunFrames));
#ifdef PR_LOGGING
if (underrunFrames) {
PR_LOG(gAudioStreamLog, PR_LOG_WARNING,
("AudioStream %p lost %d frames", this, underrunFrames));
}
#endif
mLostFrames += underrunFrames;
servicedFrames += underrunFrames;
}
WriteDumpFile(mDumpFile, this, aFrames, aBuffer);
if (PR_LOG_TEST(GetLatencyLog(), PR_LOG_DEBUG)) {
uint32_t latency = UINT32_MAX;
if (cubeb_stream_get_latency(mCubebStream, &latency)) {
NS_WARNING("Could not get latency from cubeb.");
}
LogLatency(AsyncLatencyLogger::AudioStream, 0, (mBuffer.Length() * 1000) / mOutRate);
LogLatency(AsyncLatencyLogger::Cubeb, 0, (latency * 1000) / mOutRate);
}
mAudioClock.UpdateWritePosition(servicedFrames);
return servicedFrames;
}
void
BufferedAudioStream::StateCallback(cubeb_state aState)
{
MonitorAutoLock mon(mMonitor);
if (aState == CUBEB_STATE_DRAINED) {
mState = DRAINED;
} else if (aState == CUBEB_STATE_ERROR) {
mState = ERRORED;
}
mon.NotifyAll();
}
#endif
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::GetPosition()
{
int64_t position = mAudioStream->GetPositionInFramesInternal();
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 (GetPosition() * mOutRate) / USECS_PER_S;
}
void AudioClock::SetPlaybackRate(double aPlaybackRate)
{
int64_t position = mAudioStream->GetPositionInFramesInternal();
if (position > mPlaybackRateChangeOffset) {
mOldBasePosition = mBasePosition;
mBasePosition = GetPosition();
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 = GetPosition();
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