Mercurial > hg > svapp
view audioio/AudioPortAudioTarget.cpp @ 60:7b71da2d0631
* Some work on correct alignment when moving panes during playback
* Overhaul alignment for playback frame values (view manager now always
refers to reference-timeline values, only the play source deals in
playback model timeline values)
* When making a selection, ensure the selection regions shown in other
panes (and used for playback constraints if appropriate) are aligned
correctly. This may be the coolest feature ever implemented in any
program ever.
author | Chris Cannam |
---|---|
date | Thu, 22 Nov 2007 14:17:19 +0000 |
parents | bf1a53489ccc |
children | 716e9d2f91c7 89a689720ee9 |
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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; #ifdef DEBUG_AUDIO_PORT_AUDIO_TARGET #ifdef HAVE_PORTAUDIO_V18 std::cerr << "AudioPortAudioTarget: Initialising for PortAudio v18" << std::endl; #else std::cerr << "AudioPortAudioTarget: Initialising for PortAudio v19" << std::endl; #endif #endif err = Pa_Initialize(); if (err != paNoError) { std::cerr << "ERROR: AudioPortAudioTarget: Failed to initialize PortAudio: " << Pa_GetErrorText(err) << std::endl; return; } m_bufferSize = 1024; m_sampleRate = 44100; if (m_source && (m_source->getSourceSampleRate() != 0)) { m_sampleRate = m_source->getSourceSampleRate(); } #ifdef HAVE_PORTAUDIO_V18 m_latency = Pa_GetMinNumBuffers(m_bufferSize, m_sampleRate) * m_bufferSize; #endif #ifdef HAVE_PORTAUDIO_V18 err = Pa_OpenDefaultStream(&m_stream, 0, 2, paFloat32, m_sampleRate, m_bufferSize, 0, processStatic, this); #else err = Pa_OpenDefaultStream(&m_stream, 0, 2, paFloat32, m_sampleRate, m_bufferSize, processStatic, this); #endif if (err != paNoError) { std::cerr << "ERROR: AudioPortAudioTarget: Failed to open PortAudio stream: " << Pa_GetErrorText(err) << std::endl; m_stream = 0; Pa_Terminate(); return; } #ifndef HAVE_PORTAUDIO_V18 const PaStreamInfo *info = Pa_GetStreamInfo(m_stream); m_latency = int(info->outputLatency * m_sampleRate + 0.001); #endif 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->setTargetBlockSize(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() { if (m_stream) { PaError err; err = Pa_CloseStream(m_stream); if (err != paNoError) { std::cerr << "ERROR: AudioPortAudioTarget: Failed to close PortAudio stream: " << Pa_GetErrorText(err) << std::endl; } err = Pa_Terminate(); if (err != paNoError) { std::cerr << "ERROR: AudioPortAudioTarget: Failed to terminate PortAudio: " << Pa_GetErrorText(err) << std::endl; } } } bool AudioPortAudioTarget::isOK() const { return (m_stream != 0); } #ifdef HAVE_PORTAUDIO_V18 int AudioPortAudioTarget::processStatic(void *input, void *output, unsigned long nframes, PaTimestamp outTime, void *data) { return ((AudioPortAudioTarget *)data)->process(input, output, nframes, outTime); } #else 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); } #endif void AudioPortAudioTarget::sourceModelReplaced() { m_source->setTargetSampleRate(m_sampleRate); } #ifdef HAVE_PORTAUDIO_V18 int AudioPortAudioTarget::process(void *inputBuffer, void *outputBuffer, unsigned long nframes, PaTimestamp) #else int AudioPortAudioTarget::process(const void *, void *outputBuffer, unsigned long nframes, const PaStreamCallbackTimeInfo *, PaStreamCallbackFlags) #endif { #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 || 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); return 0; } #endif /* HAVE_PORTAUDIO */