/* -*- 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_2_0 #include "AudioPortAudioTarget.h" #include "AudioCallbackPlaySource.h" #include #include #include #ifndef _WIN32 #include #endif //#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), m_prioritySet(false), m_done(false) { PaError err; #ifdef DEBUG_AUDIO_PORT_AUDIO_TARGET std::cerr << "AudioPortAudioTarget: Initialising for PortAudio v19" << std::endl; #endif err = Pa_Initialize(); if (err != paNoError) { std::cerr << "ERROR: AudioPortAudioTarget: Failed to initialize PortAudio: " << Pa_GetErrorText(err) << std::endl; return; } m_bufferSize = 2048; m_sampleRate = 44100; if (m_source && (m_source->getSourceSampleRate() != 0)) { m_sampleRate = m_source->getSourceSampleRate(); } PaStreamParameters op; op.device = Pa_GetDefaultOutputDevice(); op.channelCount = 2; op.sampleFormat = paFloat32; op.suggestedLatency = 1.0; op.hostApiSpecificStreamInfo = 0; err = Pa_OpenStream(&m_stream, 0, &op, m_sampleRate, paFramesPerBufferUnspecified, paNoFlag, processStatic, this); if (err != paNoError) { std::cerr << "WARNING: AudioPortAudioTarget: Failed to open PortAudio stream with default frames per buffer, trying again with fixed frames per buffer..." << std::endl; err = Pa_OpenStream(&m_stream, 0, &op, m_sampleRate, 1024, paNoFlag, processStatic, this); m_bufferSize = 1024; } if (err != paNoError) { std::cerr << "ERROR: AudioPortAudioTarget: Failed to open PortAudio stream: " << Pa_GetErrorText(err) << std::endl; m_stream = 0; Pa_Terminate(); return; } const PaStreamInfo *info = Pa_GetStreamInfo(m_stream); m_latency = int(info->outputLatency * m_sampleRate + 0.001); if (m_bufferSize < m_latency) m_bufferSize = m_latency; std::cerr << "PortAudio latency = " << m_latency << " frames" << std::endl; err = Pa_StartStream(m_stream); if (err != paNoError) { std::cerr << "ERROR: AudioPortAudioTarget: Failed to start PortAudio stream: " << Pa_GetErrorText(err) << 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->setTarget(this, m_bufferSize); m_source->setTargetSampleRate(m_sampleRate); m_source->setTargetPlayLatency(m_latency); } #ifdef DEBUG_PORT_AUDIO_TARGET std::cerr << "AudioPortAudioTarget: initialised OK" << std::endl; #endif } AudioPortAudioTarget::~AudioPortAudioTarget() { std::cerr << "AudioPortAudioTarget::~AudioPortAudioTarget()" << std::endl; if (m_source) { m_source->setTarget(0, m_bufferSize); } shutdown(); if (m_stream) { std::cerr << "closing stream" << std::endl; PaError err; err = Pa_CloseStream(m_stream); if (err != paNoError) { std::cerr << "ERROR: AudioPortAudioTarget: Failed to close PortAudio stream: " << Pa_GetErrorText(err) << std::endl; } std::cerr << "terminating" << std::endl; err = Pa_Terminate(); if (err != paNoError) { std::cerr << "ERROR: AudioPortAudioTarget: Failed to terminate PortAudio: " << Pa_GetErrorText(err) << std::endl; } } m_stream = 0; std::cerr << "AudioPortAudioTarget::~AudioPortAudioTarget() done" << std::endl; } void AudioPortAudioTarget::shutdown() { #ifdef DEBUG_PORT_AUDIO_TARGET std::cerr << "AudioPortAudioTarget::shutdown" << std::endl; #endif m_done = true; } bool AudioPortAudioTarget::isOK() const { return (m_stream != 0); } double AudioPortAudioTarget::getCurrentTime() const { if (!m_stream) return 0.0; else return Pa_GetStreamTime(m_stream); } int AudioPortAudioTarget::processStatic(const void *input, void *output, unsigned long nframes, const PaStreamCallbackTimeInfo *timeInfo, PaStreamCallbackFlags flags, void *data) { return ((AudioPortAudioTarget *)data)->process(input, output, nframes, timeInfo, flags); } void AudioPortAudioTarget::sourceModelReplaced() { m_source->setTargetSampleRate(m_sampleRate); } int AudioPortAudioTarget::process(const void *, void *outputBuffer, unsigned long nframes, const PaStreamCallbackTimeInfo *, PaStreamCallbackFlags) { #ifdef DEBUG_AUDIO_PORT_AUDIO_TARGET std::cerr << "AudioPortAudioTarget::process(" << nframes << ")" << std::endl; #endif if (!m_source || m_done) { #ifdef DEBUG_AUDIO_PORT_AUDIO_TARGET std::cerr << "AudioPortAudioTarget::process: Doing nothing, no source or application done" << std::endl; #endif return 0; } if (!m_prioritySet) { #ifndef _WIN32 sched_param param; param.sched_priority = 20; if (pthread_setschedparam(pthread_self(), SCHED_RR, ¶m)) { std::cerr << "AudioPortAudioTarget: NOTE: couldn't set RT scheduling class" << std::endl; } else { std::cerr << "AudioPortAudioTarget: NOTE: successfully set RT scheduling class" << std::endl; } #endif m_prioritySet = true; } 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 || int(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]; } } size_t received = 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) { if (i < received) { output[i * 2 + ch] = tmpbuf[ch][i] * m_outputGain; float sample = fabsf(output[i * 2 + ch]); if (sample > peak) peak = sample; } else { output[i * 2 + ch] = 0; } } } else if (ch == 1 && sourceChannels == 1) { for (size_t i = 0; i < nframes; ++i) { if (i < received) { output[i * 2 + ch] = tmpbuf[0][i] * m_outputGain; float sample = fabsf(output[i * 2 + ch]); if (sample > peak) peak = sample; } else { output[i * 2 + ch] = 0; } } } 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); if (Pa_GetStreamCpuLoad(m_stream) > 0.7) { if (m_source) m_source->audioProcessingOverload(); } return 0; } #endif /* HAVE_PORTAUDIO */