Mercurial > hg > sonic-visualiser
view audioio/AudioPortAudioTarget.cpp @ 15:cc566264c935
* timestretcher improvements -- simplify API (it can calculate its own
processing block sizes etc)
author | Chris Cannam |
---|---|
date | Wed, 13 Sep 2006 11:56:44 +0000 |
parents | cd5d7ff8ef38 |
children | 52409ab73526 |
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/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ /* Sonic Visualiser An audio file viewer and annotation editor. Centre for Digital Music, Queen Mary, University of London. This file copyright 2006 Chris Cannam. This program is free software; you can redistribute it and/or modify it under the terms of the GNU General Public License as published by the Free Software Foundation; either version 2 of the License, or (at your option) any later version. See the file COPYING included with this distribution for more information. */ #ifdef HAVE_PORTAUDIO #include "AudioPortAudioTarget.h" #include "AudioCallbackPlaySource.h" #include <iostream> #include <cassert> #include <cmath> //#define DEBUG_AUDIO_PORT_AUDIO_TARGET 1 AudioPortAudioTarget::AudioPortAudioTarget(AudioCallbackPlaySource *source) : AudioCallbackPlayTarget(source), m_stream(0), m_bufferSize(0), m_sampleRate(0), m_latency(0) { PaError err; err = Pa_Initialize(); if (err != paNoError) { std::cerr << "ERROR: AudioPortAudioTarget: Failed to initialize PortAudio" << std::endl; return; } m_bufferSize = 1024; m_sampleRate = 44100; if (m_source && (m_source->getSourceSampleRate() != 0)) { m_sampleRate = m_source->getSourceSampleRate(); } m_latency = Pa_GetMinNumBuffers(m_bufferSize, m_sampleRate) * m_bufferSize; std::cerr << "\n\n\nLATENCY= " << m_latency << std::endl; err = Pa_OpenDefaultStream(&m_stream, 0, 2, paFloat32, m_sampleRate, m_bufferSize, 0, processStatic, this); if (err != paNoError) { std::cerr << "ERROR: AudioPortAudioTarget: Failed to open PortAudio stream" << std::endl; m_stream = 0; Pa_Terminate(); return; } err = Pa_StartStream(m_stream); if (err != paNoError) { std::cerr << "ERROR: AudioPortAudioTarget: Failed to start PortAudio stream" << std::endl; Pa_CloseStream(m_stream); m_stream = 0; Pa_Terminate(); return; } if (m_source) { std::cerr << "AudioPortAudioTarget: block size " << m_bufferSize << std::endl; m_source->setTargetBlockSize(m_bufferSize); m_source->setTargetSampleRate(m_sampleRate); m_source->setTargetPlayLatency(m_latency); } } AudioPortAudioTarget::~AudioPortAudioTarget() { if (m_stream) { PaError err; err = Pa_CloseStream(m_stream); if (err != paNoError) { std::cerr << "ERROR: AudioPortAudioTarget: Failed to close PortAudio stream" << std::endl; } Pa_Terminate(); } } bool AudioPortAudioTarget::isOK() const { return (m_stream != 0); } int AudioPortAudioTarget::processStatic(void *input, void *output, unsigned long nframes, PaTimestamp outTime, void *data) { return ((AudioPortAudioTarget *)data)->process(input, output, nframes, outTime); } void AudioPortAudioTarget::sourceModelReplaced() { m_source->setTargetSampleRate(m_sampleRate); } int AudioPortAudioTarget::process(void *inputBuffer, void *outputBuffer, unsigned long nframes, PaTimestamp) { #ifdef DEBUG_AUDIO_PORT_AUDIO_TARGET std::cout << "AudioPortAudioTarget::process(" << nframes << ")" << std::endl; #endif if (!m_source) return 0; float *output = (float *)outputBuffer; assert(nframes <= m_bufferSize); static float **tmpbuf = 0; static size_t tmpbufch = 0; static size_t tmpbufsz = 0; size_t sourceChannels = m_source->getSourceChannelCount(); // Because we offer pan, we always want at least 2 channels if (sourceChannels < 2) sourceChannels = 2; if (!tmpbuf || tmpbufch != sourceChannels || tmpbufsz < m_bufferSize) { if (tmpbuf) { for (size_t i = 0; i < tmpbufch; ++i) { delete[] tmpbuf[i]; } delete[] tmpbuf; } tmpbufch = sourceChannels; tmpbufsz = m_bufferSize; tmpbuf = new float *[tmpbufch]; for (size_t i = 0; i < tmpbufch; ++i) { tmpbuf[i] = new float[tmpbufsz]; } } m_source->getSourceSamples(nframes, tmpbuf); float peakLeft = 0.0, peakRight = 0.0; for (size_t ch = 0; ch < 2; ++ch) { float peak = 0.0; if (ch < sourceChannels) { // PortAudio samples are interleaved for (size_t i = 0; i < nframes; ++i) { output[i * 2 + ch] = tmpbuf[ch][i] * m_outputGain; float sample = fabsf(output[i * 2 + ch]); if (sample > peak) peak = sample; } } else if (ch == 1 && sourceChannels == 1) { for (size_t i = 0; i < nframes; ++i) { output[i * 2 + ch] = tmpbuf[0][i] * m_outputGain; float sample = fabsf(output[i * 2 + ch]); if (sample > peak) peak = sample; } } else { for (size_t i = 0; i < nframes; ++i) { output[i * 2 + ch] = 0; } } if (ch == 0) peakLeft = peak; if (ch > 0 || sourceChannels == 1) peakRight = peak; } m_source->setOutputLevels(peakLeft, peakRight); return 0; } #ifdef INCLUDE_MOCFILES #include "AudioPortAudioTarget.moc.cpp" #endif #endif /* HAVE_PORTAUDIO */