Mercurial > hg > svapp
view audioio/AudioPortAudioTarget.cpp @ 154:386b02c926bf
* Merge from one-fftdataserver-per-fftmodel branch. This bit of
reworking (which is not described very accurately by the title of
the branch) turns the MatrixFile object into something that either
reads or writes, but not both, and separates the FFT file cache
reader and writer implementations separately. This allows the
FFT data server to have a single thread owning writers and one reader
per "customer" thread, and for all locking to be vastly simplified
and concentrated in the data server alone (because none of the
classes it makes use of is used in more than one thread at a time).
The result is faster and more trustworthy code.
author | Chris Cannam |
---|---|
date | Tue, 27 Jan 2009 13:25:10 +0000 |
parents | 4c9c04645685 |
children | 7dae51741cc9 |
line wrap: on
line source
/* -*- 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 <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), 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 = 0.2; 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() { 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::cout << "AudioPortAudioTarget::process(" << nframes << ")" << std::endl; #endif if (!m_source || m_done) 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 || 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 */