Audio: Remove all traces of rubberband
We will not be using librubberband for in place pitch shifting. RMXP "shifts" PCM based audio by just playing it back slower/faster, which OpenAL takes care of for us. A native midi backend will be able to effortlessly pitch shift by multiplying note pitches, should we ever get one. I have been chasing this ghost for way too long.
This commit is contained in:
Jonas Kulla
parent
e26d0366f1
commit
a817b31e7c
1 changed files with 33 additions and 154 deletions
187
src/audio.cpp
187
src/audio.cpp
|
|
@ -48,81 +48,6 @@
|
|||
#define SE_SOURCES 6
|
||||
#define SE_CACHE_MEM (10*1024*1024) // 10 MB
|
||||
|
||||
static void genFloatSamples(int sampleCount, int sdlFormat, const void *in, float *out)
|
||||
{
|
||||
(void) genFloatSamples;
|
||||
|
||||
int i = sampleCount;
|
||||
|
||||
/* Convert from the possible fixed point
|
||||
* formats to [-1;1] floats */
|
||||
switch (sdlFormat)
|
||||
{
|
||||
case AUDIO_U8 :
|
||||
{
|
||||
const uint8_t *_in = static_cast<const uint8_t*>(in);
|
||||
|
||||
while (i--)
|
||||
out[i] = (((float) _in[i] / 255.f) - 0.5) * 2;
|
||||
|
||||
break;
|
||||
}
|
||||
case AUDIO_S8 :
|
||||
{
|
||||
const int8_t *_in = static_cast<const int8_t*>(in);
|
||||
|
||||
while (i--)
|
||||
out[i] = (float) _in[i] / 128.f; // ???
|
||||
|
||||
break;
|
||||
}
|
||||
case AUDIO_U16LSB :
|
||||
{
|
||||
const uint16_t *_in = static_cast<const uint16_t*>(in);
|
||||
|
||||
while (i--)
|
||||
out[i] = (((float) _in[i] / 65536.f) - 0.5) * 2;
|
||||
|
||||
break;
|
||||
}
|
||||
case AUDIO_U16MSB :
|
||||
{
|
||||
const int16_t *_in = static_cast<const int16_t*>(in);
|
||||
|
||||
while (i--)
|
||||
{
|
||||
int16_t swp = SDL_Swap16(_in[i]);
|
||||
out[i] = (((float) swp / 65536.f) - 0.5) * 2;
|
||||
}
|
||||
|
||||
break;
|
||||
}
|
||||
case AUDIO_S16LSB :
|
||||
{
|
||||
const int16_t *_in = static_cast<const int16_t*>(in);
|
||||
|
||||
while (i--)
|
||||
out[i] = (float) _in[i] / 32768.f; // ???
|
||||
|
||||
break;
|
||||
}
|
||||
case AUDIO_S16MSB :
|
||||
{
|
||||
const int16_t *_in = static_cast<const int16_t*>(in);
|
||||
|
||||
while (i--)
|
||||
{
|
||||
int16_t swp = SDL_Swap16(_in[i]);
|
||||
out[i] = (float) swp / 32768.f;
|
||||
}
|
||||
|
||||
break;
|
||||
}
|
||||
qDebug() << "Unhandled sample format";
|
||||
default: Q_ASSERT(0); // XXX
|
||||
}
|
||||
}
|
||||
|
||||
static uint8_t formatSampleSize(int sdlFormat)
|
||||
{
|
||||
switch (sdlFormat)
|
||||
|
|
@ -183,10 +108,8 @@ static const int streamBufSize = 32768;
|
|||
|
||||
struct SoundBuffer
|
||||
{
|
||||
/* Filename, pitch.
|
||||
* Uniquely identifies this or equal buffer */
|
||||
typedef QPair<QByteArray, int> Key;
|
||||
Key key;
|
||||
/* Uniquely identifies this or equal buffer */
|
||||
QByteArray key;
|
||||
|
||||
AL::Buffer::ID alBuffer;
|
||||
|
||||
|
|
@ -229,7 +152,7 @@ struct SoundBuffer
|
|||
struct SoundEmitter
|
||||
{
|
||||
IntruList<SoundBuffer> buffers;
|
||||
QHash<SoundBuffer::Key, SoundBuffer*> bufferHash;
|
||||
QHash<QByteArray, SoundBuffer*> bufferHash;
|
||||
|
||||
/* Byte count sum of all cached / playing buffers */
|
||||
uint32_t bufferBytes;
|
||||
|
|
@ -239,10 +162,6 @@ struct SoundEmitter
|
|||
/* Index of next source to be used */
|
||||
int srcIndex;
|
||||
|
||||
#ifdef RUBBERBAND
|
||||
std::vector<float> floatBuf;
|
||||
#endif
|
||||
|
||||
SoundEmitter()
|
||||
: bufferBytes(0),
|
||||
srcIndex(0)
|
||||
|
|
@ -252,9 +171,6 @@ struct SoundEmitter
|
|||
alSrcs[i] = AL::Source::gen();
|
||||
atchBufs[i] = 0;
|
||||
}
|
||||
#ifdef RUBBERBAND
|
||||
floatBuf.resize(streamBufSize/4);
|
||||
#endif
|
||||
}
|
||||
|
||||
~SoundEmitter()
|
||||
|
|
@ -268,7 +184,7 @@ struct SoundEmitter
|
|||
SoundBuffer::deref(atchBufs[i]);
|
||||
}
|
||||
|
||||
QHash<SoundBuffer::Key, SoundBuffer*>::iterator iter;
|
||||
QHash<QByteArray, SoundBuffer*>::iterator iter;
|
||||
for (iter = bufferHash.begin(); iter != bufferHash.end(); ++iter)
|
||||
delete iter.value();
|
||||
}
|
||||
|
|
@ -278,8 +194,9 @@ struct SoundEmitter
|
|||
int pitch)
|
||||
{
|
||||
float _volume = clamp<int>(volume, 0, 100) / 100.f;
|
||||
float _pitch = clamp<int>(pitch, 50, 150) / 100.f;
|
||||
|
||||
SoundBuffer *buffer = allocateBuffer(filename, pitch);
|
||||
SoundBuffer *buffer = allocateBuffer(filename);
|
||||
|
||||
int soundIndex = srcIndex++;
|
||||
if (srcIndex > SE_SOURCES-1)
|
||||
|
|
@ -297,11 +214,9 @@ struct SoundEmitter
|
|||
atchBufs[soundIndex] = SoundBuffer::ref(buffer);
|
||||
|
||||
AL::Source::attachBuffer(src, buffer->alBuffer);
|
||||
AL::Source::setVolume(src, _volume);
|
||||
|
||||
#ifndef RUBBERBAND
|
||||
AL::Source::setPitch(src, clamp<int>(pitch, 50, 150) / 100.f);
|
||||
#endif
|
||||
AL::Source::setVolume(src, _volume);
|
||||
AL::Source::setPitch(src, _pitch);
|
||||
|
||||
AL::Source::play(src);
|
||||
}
|
||||
|
|
@ -313,26 +228,9 @@ struct SoundEmitter
|
|||
}
|
||||
|
||||
private:
|
||||
#ifdef RUBBERBAND
|
||||
void ensureFloatBufSize(uint32_t size)
|
||||
SoundBuffer *allocateBuffer(const QByteArray &filename)
|
||||
{
|
||||
if (size <= floatBuf.size())
|
||||
return;
|
||||
|
||||
floatBuf.resize(size);
|
||||
}
|
||||
#endif
|
||||
|
||||
SoundBuffer *allocateBuffer(const QByteArray &filename,
|
||||
int pitch)
|
||||
{
|
||||
#ifndef RUBBERBAND
|
||||
pitch = 100;
|
||||
#endif
|
||||
|
||||
SoundBuffer::Key soundKey(filename, pitch);
|
||||
|
||||
SoundBuffer *buffer = bufferHash.value(soundKey, 0);
|
||||
SoundBuffer *buffer = bufferHash.value(filename, 0);
|
||||
|
||||
if (buffer)
|
||||
{
|
||||
|
|
@ -364,34 +262,15 @@ private:
|
|||
uint8_t sampleSize = formatSampleSize(sampleHandle->actual.format);
|
||||
uint32_t sampleCount = decBytes / sampleSize;
|
||||
|
||||
uint32_t actSampleSize;
|
||||
#ifdef RUBBERBAND
|
||||
actSampleSize = sizeof(float);
|
||||
#else
|
||||
actSampleSize = sampleSize;
|
||||
#endif
|
||||
|
||||
buffer = new SoundBuffer;
|
||||
buffer->key = soundKey;
|
||||
buffer->bytes = actSampleSize * sampleCount;
|
||||
buffer->key = filename;
|
||||
buffer->bytes = sampleSize * sampleCount;
|
||||
|
||||
ALenum alFormat = chooseALFormat(actSampleSize, sampleHandle->actual.channels);
|
||||
ALenum alFormat = chooseALFormat(sampleSize, sampleHandle->actual.channels);
|
||||
|
||||
#ifdef RUBBERBAND
|
||||
/* Fill float buffer */
|
||||
ensureFloatBufSize(sampleCount);
|
||||
genFloatSamples(sampleCount, sampleHandle->actual.format,
|
||||
sampleHandle->buffer, floatBuf.data());
|
||||
|
||||
// XXX apply stretcher
|
||||
|
||||
/* Upload data to AL */
|
||||
AL::Buffer::uploadData(buffer->alBuffer, alFormat, floatBuf.data(),
|
||||
buffer->bytes, sampleHandle->actual.rate);
|
||||
#else
|
||||
AL::Buffer::uploadData(buffer->alBuffer, alFormat, sampleHandle->buffer,
|
||||
buffer->bytes, sampleHandle->actual.rate);
|
||||
#endif
|
||||
|
||||
Sound_FreeSample(sampleHandle);
|
||||
|
||||
uint32_t wouldBeBytes = bufferBytes + buffer->bytes;
|
||||
|
|
@ -409,7 +288,7 @@ private:
|
|||
SoundBuffer::deref(last);
|
||||
}
|
||||
|
||||
bufferHash.insert(soundKey, buffer);
|
||||
bufferHash.insert(filename, buffer);
|
||||
buffers.prepend(buffer->link);
|
||||
|
||||
bufferBytes = wouldBeBytes;
|
||||
|
|
@ -457,12 +336,9 @@ struct SDLSoundSource : ALDataSource
|
|||
|
||||
SDLSoundSource(const std::string &filename,
|
||||
uint32_t maxBufSize,
|
||||
bool looped,
|
||||
float pitch)
|
||||
bool looped)
|
||||
: looped(looped)
|
||||
{
|
||||
(void) pitch;
|
||||
|
||||
const char *extension;
|
||||
shState->fileSystem().openRead(ops,
|
||||
filename.c_str(),
|
||||
|
|
@ -505,8 +381,6 @@ struct SDLSoundSource : ALDataSource
|
|||
if (sample->flags & SOUND_SAMPLEFLAG_ERROR)
|
||||
return ALDataSource::Error;
|
||||
|
||||
// XXX apply stretcher here
|
||||
|
||||
AL::Buffer::uploadData(alBuffer, alFormat, sample->buffer, decoded, alFreq);
|
||||
|
||||
if (sample->flags & SOUND_SAMPLEFLAG_EOF)
|
||||
|
|
@ -644,8 +518,7 @@ struct ALStream
|
|||
}
|
||||
}
|
||||
|
||||
void open(const std::string &filename,
|
||||
float pitch)
|
||||
void open(const std::string &filename)
|
||||
{
|
||||
checkStopped();
|
||||
|
||||
|
|
@ -657,7 +530,7 @@ struct ALStream
|
|||
case Stopped:
|
||||
closeSource();
|
||||
case Closed:
|
||||
openSource(filename, pitch);
|
||||
openSource(filename);
|
||||
}
|
||||
|
||||
state = Stopped;
|
||||
|
|
@ -730,6 +603,11 @@ struct ALStream
|
|||
AL::Source::setVolume(alSrc, value);
|
||||
}
|
||||
|
||||
void setPitch(float value)
|
||||
{
|
||||
AL::Source::setPitch(alSrc, value);
|
||||
}
|
||||
|
||||
State queryState()
|
||||
{
|
||||
checkStopped();
|
||||
|
|
@ -754,15 +632,10 @@ private:
|
|||
delete source;
|
||||
}
|
||||
|
||||
void openSource(const std::string &filename,
|
||||
float pitch)
|
||||
void openSource(const std::string &filename)
|
||||
{
|
||||
source = new SDLSoundSource(filename, streamBufSize, looped, pitch);
|
||||
source = new SDLSoundSource(filename, streamBufSize, looped);
|
||||
needsRewind = false;
|
||||
|
||||
#ifndef RUBBERBAND
|
||||
AL::Source::setPitch(alSrc, pitch);
|
||||
#endif
|
||||
}
|
||||
|
||||
void stopStream()
|
||||
|
|
@ -1084,19 +957,25 @@ struct AudioStream
|
|||
return;
|
||||
}
|
||||
|
||||
/* Requested audio file is different from current one */
|
||||
bool diffFile = (filename != current.filename);
|
||||
|
||||
switch (sState)
|
||||
{
|
||||
case ALStream::Paused :
|
||||
case ALStream::Playing :
|
||||
stream.stop();
|
||||
case ALStream::Stopped :
|
||||
stream.close();
|
||||
if (diffFile)
|
||||
stream.close();
|
||||
case ALStream::Closed :
|
||||
if (diffFile)
|
||||
stream.open(filename);
|
||||
break;
|
||||
}
|
||||
|
||||
stream.open(filename, _pitch);
|
||||
setBaseVolume(_volume);
|
||||
stream.setPitch(_pitch);
|
||||
|
||||
current.filename = filename;
|
||||
current.volume = _volume;
|
||||
|
|
|
|||
Loading…
Add table
Add a link
Reference in a new issue