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
185
src/audio.cpp
185
src/audio.cpp
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@ -48,81 +48,6 @@
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#define SE_SOURCES 6
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#define SE_CACHE_MEM (10*1024*1024) // 10 MB
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static void genFloatSamples(int sampleCount, int sdlFormat, const void *in, float *out)
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{
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(void) genFloatSamples;
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int i = sampleCount;
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/* Convert from the possible fixed point
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* formats to [-1;1] floats */
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switch (sdlFormat)
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{
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case AUDIO_U8 :
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{
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const uint8_t *_in = static_cast<const uint8_t*>(in);
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while (i--)
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out[i] = (((float) _in[i] / 255.f) - 0.5) * 2;
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break;
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}
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case AUDIO_S8 :
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{
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const int8_t *_in = static_cast<const int8_t*>(in);
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while (i--)
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out[i] = (float) _in[i] / 128.f; // ???
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break;
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}
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case AUDIO_U16LSB :
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{
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const uint16_t *_in = static_cast<const uint16_t*>(in);
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while (i--)
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out[i] = (((float) _in[i] / 65536.f) - 0.5) * 2;
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break;
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}
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case AUDIO_U16MSB :
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{
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const int16_t *_in = static_cast<const int16_t*>(in);
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while (i--)
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{
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int16_t swp = SDL_Swap16(_in[i]);
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out[i] = (((float) swp / 65536.f) - 0.5) * 2;
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}
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break;
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}
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case AUDIO_S16LSB :
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{
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const int16_t *_in = static_cast<const int16_t*>(in);
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while (i--)
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out[i] = (float) _in[i] / 32768.f; // ???
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break;
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}
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case AUDIO_S16MSB :
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{
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const int16_t *_in = static_cast<const int16_t*>(in);
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while (i--)
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{
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int16_t swp = SDL_Swap16(_in[i]);
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out[i] = (float) swp / 32768.f;
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}
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break;
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}
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qDebug() << "Unhandled sample format";
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default: Q_ASSERT(0); // XXX
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}
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}
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static uint8_t formatSampleSize(int sdlFormat)
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{
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switch (sdlFormat)
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@ -183,10 +108,8 @@ static const int streamBufSize = 32768;
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struct SoundBuffer
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{
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/* Filename, pitch.
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* Uniquely identifies this or equal buffer */
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typedef QPair<QByteArray, int> Key;
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Key key;
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/* Uniquely identifies this or equal buffer */
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QByteArray key;
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AL::Buffer::ID alBuffer;
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@ -229,7 +152,7 @@ struct SoundBuffer
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struct SoundEmitter
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{
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IntruList<SoundBuffer> buffers;
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QHash<SoundBuffer::Key, SoundBuffer*> bufferHash;
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QHash<QByteArray, SoundBuffer*> bufferHash;
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/* Byte count sum of all cached / playing buffers */
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uint32_t bufferBytes;
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@ -239,10 +162,6 @@ struct SoundEmitter
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/* Index of next source to be used */
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int srcIndex;
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#ifdef RUBBERBAND
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std::vector<float> floatBuf;
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#endif
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SoundEmitter()
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: bufferBytes(0),
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srcIndex(0)
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@ -252,9 +171,6 @@ struct SoundEmitter
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alSrcs[i] = AL::Source::gen();
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atchBufs[i] = 0;
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}
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#ifdef RUBBERBAND
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floatBuf.resize(streamBufSize/4);
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#endif
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}
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~SoundEmitter()
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@ -268,7 +184,7 @@ struct SoundEmitter
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SoundBuffer::deref(atchBufs[i]);
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}
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QHash<SoundBuffer::Key, SoundBuffer*>::iterator iter;
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QHash<QByteArray, SoundBuffer*>::iterator iter;
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for (iter = bufferHash.begin(); iter != bufferHash.end(); ++iter)
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delete iter.value();
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}
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@ -278,8 +194,9 @@ struct SoundEmitter
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int pitch)
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{
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float _volume = clamp<int>(volume, 0, 100) / 100.f;
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float _pitch = clamp<int>(pitch, 50, 150) / 100.f;
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SoundBuffer *buffer = allocateBuffer(filename, pitch);
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SoundBuffer *buffer = allocateBuffer(filename);
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int soundIndex = srcIndex++;
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if (srcIndex > SE_SOURCES-1)
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@ -297,11 +214,9 @@ struct SoundEmitter
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atchBufs[soundIndex] = SoundBuffer::ref(buffer);
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AL::Source::attachBuffer(src, buffer->alBuffer);
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AL::Source::setVolume(src, _volume);
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#ifndef RUBBERBAND
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AL::Source::setPitch(src, clamp<int>(pitch, 50, 150) / 100.f);
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#endif
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AL::Source::setVolume(src, _volume);
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AL::Source::setPitch(src, _pitch);
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AL::Source::play(src);
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}
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@ -313,26 +228,9 @@ struct SoundEmitter
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}
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private:
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#ifdef RUBBERBAND
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void ensureFloatBufSize(uint32_t size)
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SoundBuffer *allocateBuffer(const QByteArray &filename)
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{
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if (size <= floatBuf.size())
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return;
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floatBuf.resize(size);
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}
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#endif
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SoundBuffer *allocateBuffer(const QByteArray &filename,
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int pitch)
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{
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#ifndef RUBBERBAND
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pitch = 100;
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#endif
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SoundBuffer::Key soundKey(filename, pitch);
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SoundBuffer *buffer = bufferHash.value(soundKey, 0);
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SoundBuffer *buffer = bufferHash.value(filename, 0);
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if (buffer)
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{
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@ -364,34 +262,15 @@ private:
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uint8_t sampleSize = formatSampleSize(sampleHandle->actual.format);
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uint32_t sampleCount = decBytes / sampleSize;
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uint32_t actSampleSize;
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#ifdef RUBBERBAND
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actSampleSize = sizeof(float);
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#else
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actSampleSize = sampleSize;
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#endif
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buffer = new SoundBuffer;
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buffer->key = soundKey;
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buffer->bytes = actSampleSize * sampleCount;
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buffer->key = filename;
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buffer->bytes = sampleSize * sampleCount;
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ALenum alFormat = chooseALFormat(actSampleSize, sampleHandle->actual.channels);
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ALenum alFormat = chooseALFormat(sampleSize, sampleHandle->actual.channels);
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#ifdef RUBBERBAND
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/* Fill float buffer */
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ensureFloatBufSize(sampleCount);
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genFloatSamples(sampleCount, sampleHandle->actual.format,
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sampleHandle->buffer, floatBuf.data());
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// XXX apply stretcher
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/* Upload data to AL */
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AL::Buffer::uploadData(buffer->alBuffer, alFormat, floatBuf.data(),
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buffer->bytes, sampleHandle->actual.rate);
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#else
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AL::Buffer::uploadData(buffer->alBuffer, alFormat, sampleHandle->buffer,
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buffer->bytes, sampleHandle->actual.rate);
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#endif
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Sound_FreeSample(sampleHandle);
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uint32_t wouldBeBytes = bufferBytes + buffer->bytes;
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@ -409,7 +288,7 @@ private:
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SoundBuffer::deref(last);
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}
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bufferHash.insert(soundKey, buffer);
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bufferHash.insert(filename, buffer);
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buffers.prepend(buffer->link);
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bufferBytes = wouldBeBytes;
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@ -457,12 +336,9 @@ struct SDLSoundSource : ALDataSource
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SDLSoundSource(const std::string &filename,
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uint32_t maxBufSize,
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bool looped,
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float pitch)
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bool looped)
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: looped(looped)
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{
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(void) pitch;
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const char *extension;
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shState->fileSystem().openRead(ops,
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filename.c_str(),
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@ -505,8 +381,6 @@ struct SDLSoundSource : ALDataSource
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if (sample->flags & SOUND_SAMPLEFLAG_ERROR)
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return ALDataSource::Error;
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// XXX apply stretcher here
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AL::Buffer::uploadData(alBuffer, alFormat, sample->buffer, decoded, alFreq);
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if (sample->flags & SOUND_SAMPLEFLAG_EOF)
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@ -644,8 +518,7 @@ struct ALStream
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}
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}
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void open(const std::string &filename,
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float pitch)
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void open(const std::string &filename)
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{
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checkStopped();
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@ -657,7 +530,7 @@ struct ALStream
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case Stopped:
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closeSource();
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case Closed:
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openSource(filename, pitch);
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openSource(filename);
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}
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state = Stopped;
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@ -730,6 +603,11 @@ struct ALStream
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AL::Source::setVolume(alSrc, value);
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}
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void setPitch(float value)
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{
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AL::Source::setPitch(alSrc, value);
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}
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State queryState()
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{
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checkStopped();
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@ -754,15 +632,10 @@ private:
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delete source;
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}
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void openSource(const std::string &filename,
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float pitch)
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void openSource(const std::string &filename)
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{
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source = new SDLSoundSource(filename, streamBufSize, looped, pitch);
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source = new SDLSoundSource(filename, streamBufSize, looped);
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needsRewind = false;
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#ifndef RUBBERBAND
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AL::Source::setPitch(alSrc, pitch);
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#endif
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}
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void stopStream()
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@ -1084,19 +957,25 @@ struct AudioStream
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return;
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}
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/* Requested audio file is different from current one */
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bool diffFile = (filename != current.filename);
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switch (sState)
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{
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case ALStream::Paused :
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case ALStream::Playing :
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stream.stop();
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case ALStream::Stopped :
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if (diffFile)
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stream.close();
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case ALStream::Closed :
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if (diffFile)
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stream.open(filename);
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break;
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}
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stream.open(filename, _pitch);
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setBaseVolume(_volume);
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stream.setPitch(_pitch);
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current.filename = filename;
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current.volume = _volume;
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