Mercurial > hg > sonic-visualiser
changeset 31:37af203dbd15
* Buffer size fixes in the time stretcher, to avoid running out of input data
for large or small ratios
author | Chris Cannam |
---|---|
date | Thu, 21 Sep 2006 09:43:41 +0000 |
parents | 56e1d4242bb4 |
children | e3b32dc5180b |
files | audioio/AudioCallbackPlaySource.cpp audioio/PhaseVocoderTimeStretcher.cpp audioio/PhaseVocoderTimeStretcher.h |
diffstat | 3 files changed, 86 insertions(+), 80 deletions(-) [+] |
line wrap: on
line diff
--- a/audioio/AudioCallbackPlaySource.cpp Wed Sep 20 16:02:42 2006 +0000 +++ b/audioio/AudioCallbackPlaySource.cpp Thu Sep 21 09:43:41 2006 +0000 @@ -619,7 +619,7 @@ channels, factor, sharpen, - lrintf(getTargetBlockSize() / factor)); + getTargetBlockSize()); m_timeStretcher = newStretcher; @@ -688,6 +688,21 @@ size_t available; + int warned = 0; + + + + //!!! + // We want output blocks of e.g. 1024 (probably fixed, certainly + // bounded). We can provide input blocks of any size (unbounded) + // at the timestretcher's request. The input block for a given + // output is approx output / ratio, but we can't predict it + // exactly, for an adaptive timestretcher. The stretcher will + // need some additional buffer space. + + + + while ((available = ts->getAvailableOutputSamples()) < count) { size_t reqd = lrintf((count - available) / ratio); @@ -735,8 +750,8 @@ if (got == 0) break; if (ts->getAvailableOutputSamples() == available) { - std::cerr << "WARNING: AudioCallbackPlaySource::getSamples: Added " << got << " samples to time stretcher, created no new available output samples" << std::endl; - break; + std::cerr << "WARNING: AudioCallbackPlaySource::getSamples: Added " << got << " samples to time stretcher, created no new available output samples (warned = " << warned << ")" << std::endl; + if (++warned == 5) break; } }
--- a/audioio/PhaseVocoderTimeStretcher.cpp Wed Sep 20 16:02:42 2006 +0000 +++ b/audioio/PhaseVocoderTimeStretcher.cpp Thu Sep 21 09:43:41 2006 +0000 @@ -26,10 +26,10 @@ size_t channels, float ratio, bool sharpen, - size_t maxProcessInputBlockSize) : + size_t maxOutputBlockSize) : m_sampleRate(sampleRate), m_channels(channels), - m_maxProcessInputBlockSize(maxProcessInputBlockSize), + m_maxOutputBlockSize(maxOutputBlockSize), m_ratio(ratio), m_sharpen(sharpen), m_totalCount(0), @@ -92,27 +92,31 @@ m_plan[c] = fftwf_plan_dft_r2c_1d(m_wlen, m_time[c], m_freq[c], FFTW_ESTIMATE); m_iplan[c] = fftwf_plan_dft_c2r_1d(m_wlen, m_freq[c], m_time[c], FFTW_ESTIMATE); - m_inbuf[c] = new RingBuffer<float>(m_wlen); m_outbuf[c] = new RingBuffer<float> - (lrintf((m_maxProcessInputBlockSize + m_wlen) * m_ratio)); - + ((m_maxOutputBlockSize + m_wlen) * 2); + m_inbuf[c] = new RingBuffer<float> + (lrintf(m_outbuf[c]->getSize() / m_ratio) + m_wlen); + + std::cerr << "making inbuf size " << m_inbuf[c]->getSize() << " (outbuf size is " << m_outbuf[c]->getSize() << ", ratio " << m_ratio << ")" << std::endl; + + m_mashbuf[c] = (float *)fftwf_malloc(sizeof(float) * m_wlen); - for (int i = 0; i < m_wlen; ++i) { + for (size_t i = 0; i < m_wlen; ++i) { m_mashbuf[c][i] = 0.0; } - for (int i = 0; i <= m_wlen/2; ++i) { + for (size_t i = 0; i <= m_wlen/2; ++i) { m_prevPhase[c][i] = 0.0; m_prevAdjustedPhase[c][i] = 0.0; } } - for (int i = 0; i < m_wlen; ++i) { + for (size_t i = 0; i < m_wlen; ++i) { m_modulationbuf[i] = 0.0; } - for (int i = 0; i <= m_wlen/2; ++i) { + for (size_t i = 0; i <= m_wlen/2; ++i) { m_prevTransientMag[i] = 0.0; } } @@ -143,7 +147,7 @@ if (m_sharpen) { m_wlen = 2048; } - m_n2 = m_n1 * m_ratio; + m_n2 = lrintf(m_n1 * m_ratio); } else { if (m_ratio > 2) { m_n2 = 512; @@ -157,10 +161,10 @@ if (m_sharpen) { if (m_wlen < 2048) m_wlen = 2048; } - m_n1 = m_n2 / m_ratio; + m_n1 = lrintf(m_n2 / m_ratio); } - m_transientThreshold = m_wlen / 4.5; + m_transientThreshold = lrintf(m_wlen / 4.5); m_totalCount = 0; m_transientCount = 0; @@ -170,7 +174,7 @@ std::cerr << "PhaseVocoderTimeStretcher: channels = " << m_channels << ", ratio = " << m_ratio << ", n1 = " << m_n1 << ", n2 = " << m_n2 << ", wlen = " - << m_wlen << ", max = " << m_maxProcessInputBlockSize << std::endl; + << m_wlen << ", max = " << m_maxOutputBlockSize << std::endl; // << ", outbuflen = " << m_outbuf[0]->getSize() << std::endl; } @@ -218,9 +222,7 @@ { QMutexLocker locker(m_mutex); - float formerRatio = m_ratio; size_t formerWlen = m_wlen; - m_ratio = ratio; calculateParameters(); @@ -229,36 +231,43 @@ // This is the only container whose size depends on m_ratio - RingBuffer<float> **newout = new RingBuffer<float> *[m_channels]; + RingBuffer<float> **newin = new RingBuffer<float> *[m_channels]; - size_t formerSize = m_outbuf[0]->getSize(); - size_t newSize = lrintf((m_maxProcessInputBlockSize + m_wlen) * m_ratio); - size_t ready = m_outbuf[0]->getReadSpace(); + size_t formerSize = m_inbuf[0]->getSize(); + size_t newSize = lrintf(m_outbuf[0]->getSize() / m_ratio) + m_wlen; - for (size_t c = 0; c < m_channels; ++c) { - newout[c] = new RingBuffer<float>(newSize); - } + std::cerr << "resizing inbuf from " << formerSize << " to " + << newSize << " (outbuf size is " << m_outbuf[0]->getSize() << ", ratio " << m_ratio << ")" << std::endl; - if (ready > 0) { + if (formerSize != newSize) { - size_t copy = std::min(ready, newSize); - float *tmp = new float[ready]; + size_t ready = m_inbuf[0]->getReadSpace(); for (size_t c = 0; c < m_channels; ++c) { - m_outbuf[c]->read(tmp, ready); - newout[c]->write(tmp + ready - copy, copy); + newin[c] = new RingBuffer<float>(newSize); } - delete[] tmp; + if (ready > 0) { + + size_t copy = std::min(ready, newSize); + float *tmp = new float[ready]; + + for (size_t c = 0; c < m_channels; ++c) { + m_inbuf[c]->read(tmp, ready); + newin[c]->write(tmp + ready - copy, copy); + } + + delete[] tmp; + } + + for (size_t c = 0; c < m_channels; ++c) { + delete m_inbuf[c]; + } + + delete[] m_inbuf; + m_inbuf = newin; } - for (size_t c = 0; c < m_channels; ++c) { - delete m_outbuf[c]; - } - - delete[] m_outbuf; - m_outbuf = newout; - } else { std::cerr << "wlen changed" << std::endl; @@ -273,13 +282,6 @@ return getWindowSize() - getInputIncrement(); } -void -PhaseVocoderTimeStretcher::process(float **input, float **output, size_t samples) -{ - putInput(input, samples); - getOutput(output, lrintf(samples * m_ratio)); -} - size_t PhaseVocoderTimeStretcher::getRequiredInputSamples() const { @@ -317,18 +319,23 @@ if (writable == 0) { //!!! then what? I don't think this should happen, but - std::cerr << "WARNING: PhaseVocoderTimeStretcher::putInput: writable == 0" << std::endl; - break; - } + std::cerr << "WARNING: PhaseVocoderTimeStretcher::putInput: writable == 0 (inbuf has " << m_inbuf[0]->getReadSpace() << " samples available for reading, space for " << m_inbuf[0]->getWriteSpace() << " more)" << std::endl; + if (m_inbuf[0]->getReadSpace() < m_wlen || + m_outbuf[0]->getWriteSpace() < m_n2) { + std::cerr << "Outbuf has space for " << m_outbuf[0]->getWriteSpace() << " (n2 = " << m_n2 << "), won't be able to process" << std::endl; + break; + } + } else { #ifdef DEBUG_PHASE_VOCODER_TIME_STRETCHER - std::cerr << "writing " << writable << " from index " << consumed << " to inbuf, consumed will be " << consumed + writable << std::endl; + std::cerr << "writing " << writable << " from index " << consumed << " to inbuf, consumed will be " << consumed + writable << std::endl; #endif - for (size_t c = 0; c < m_channels; ++c) { - m_inbuf[c]->write(input[c] + consumed, writable); + for (size_t c = 0; c < m_channels; ++c) { + m_inbuf[c]->write(input[c] + consumed, writable); + } + consumed += writable; } - consumed += writable; while (m_inbuf[0]->getReadSpace() >= m_wlen && m_outbuf[0]->getWriteSpace() >= m_n2) { @@ -501,7 +508,7 @@ { int count = 0; - for (int i = 0; i <= m_wlen/2; ++i) { + for (size_t i = 0; i <= m_wlen/2; ++i) { float real = 0.f, imag = 0.f; @@ -546,11 +553,9 @@ float *modulation, size_t lastStep) { - int i; - bool unchanged = (lastStep == m_n1); - for (i = 0; i <= m_wlen/2; ++i) { + for (size_t i = 0; i <= m_wlen/2; ++i) { float phase = princargf(atan2f(m_freq[c][i][1], m_freq[c][i][0])); float adjustedPhase = phase; @@ -583,19 +588,19 @@ fftwf_execute(m_iplan[c]); // m_freq -> m_time, inverse fft - for (i = 0; i < m_wlen/2; ++i) { + for (size_t i = 0; i < m_wlen/2; ++i) { float temp = m_time[c][i]; m_time[c][i] = m_time[c][i + m_wlen/2]; m_time[c][i + m_wlen/2] = temp; } - for (i = 0; i < m_wlen; ++i) { + for (size_t i = 0; i < m_wlen; ++i) { m_time[c][i] = m_time[c][i] / m_wlen; } m_synthesisWindow->cut(m_time[c]); - for (i = 0; i < m_wlen; ++i) { + for (size_t i = 0; i < m_wlen; ++i) { out[i] += m_time[c][i]; } @@ -603,7 +608,7 @@ float area = m_analysisWindow->getArea(); - for (i = 0; i < m_wlen; ++i) { + for (size_t i = 0; i < m_wlen; ++i) { float val = m_synthesisWindow->getValue(i); modulation[i] += val * area; }
--- a/audioio/PhaseVocoderTimeStretcher.h Wed Sep 20 16:02:42 2006 +0000 +++ b/audioio/PhaseVocoderTimeStretcher.h Thu Sep 21 09:43:41 2006 +0000 @@ -43,36 +43,22 @@ size_t channels, float ratio, bool sharpen, - size_t maxProcessInputBlockSize); + size_t maxOutputBlockSize); virtual ~PhaseVocoderTimeStretcher(); /** - * Process a block. The input array contains the given number of - * samples (on each channel); the output must have space for - * lrintf(samples * m_ratio). - * - * This function isn't really recommended, and I may yet remove it. - * It should work correctly for some ratios, e.g. small powers of - * two, if transient sharpening is off. For other ratios it may - * drop samples -- use putInput in a loop followed by getOutput - * (when getAvailableOutputSamples reports enough) instead. - * - * Do not mix process calls with putInput/getOutput calls. - */ - void process(float **input, float **output, size_t samples); - - /** * Return the number of samples that would need to be added via * putInput in order to provoke the time stretcher into doing some * time stretching and making more output samples available. - * This will be an estimate, if transient sharpening is on. + * This will be an estimate, if transient sharpening is on; the + * caller may need to do the put/get/test cycle more than once. */ size_t getRequiredInputSamples() const; /** * Put (and possibly process) a given number of input samples. - * Number must not exceed the maxProcessInputBlockSize passed to - * constructor. + * Number should usually equal the value returned from + * getRequiredInputSamples(). */ void putInput(float **input, size_t samples); @@ -159,7 +145,7 @@ size_t m_sampleRate; size_t m_channels; - size_t m_maxProcessInputBlockSize; + size_t m_maxOutputBlockSize; float m_ratio; bool m_sharpen; size_t m_n1;