view audioio/AudioPortAudioTarget.cpp @ 256:f3f9e3d647c1

Give a dedicated key to toggling the centre line, and move it out of the overlay level setting -- reducing number of overlay levels to 3. Introduce two distinct vertical scale types (so that we can hide the spectrogram colour scale part easily)
author Chris Cannam
date Mon, 30 Jan 2012 16:02:14 +0000
parents a9f1850b873d
children 0136555495ae 068235cf5bf7
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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_2_0

#include "AudioPortAudioTarget.h"
#include "AudioCallbackPlaySource.h"

#include <iostream>
#include <cassert>
#include <cmath>

#ifndef _WIN32
#include <pthread.h>
#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 = 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;
        std::cerr << "Note: device ID was " << op.device << 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()
{
    SVDEBUG << "AudioPortAudioTarget::~AudioPortAudioTarget()" << endl;

    if (m_source) {
        m_source->setTarget(0, m_bufferSize);
    }

    shutdown();

    if (m_stream) {

        SVDEBUG << "closing stream" << 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;

    SVDEBUG << "AudioPortAudioTarget::~AudioPortAudioTarget() done" << endl;
}

void 
AudioPortAudioTarget::shutdown()
{
#ifdef DEBUG_PORT_AUDIO_TARGET
    SVDEBUG << "AudioPortAudioTarget::shutdown" << 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    
    SVDEBUG << "AudioPortAudioTarget::process(" << nframes << ")" << endl;
#endif

    if (!m_source || m_done) {
#ifdef DEBUG_AUDIO_PORT_AUDIO_TARGET
        SVDEBUG << "AudioPortAudioTarget::process: Doing nothing, no source or application done" << endl;
#endif
        return 0;
    }

    if (!m_prioritySet) {
#ifndef _WIN32
        sched_param param;
        param.sched_priority = 20;
        if (pthread_setschedparam(pthread_self(), SCHED_RR, &param)) {
            SVDEBUG << "AudioPortAudioTarget: NOTE: couldn't set RT scheduling class" << endl;
        } else {
            SVDEBUG << "AudioPortAudioTarget: NOTE: successfully set RT scheduling class" << 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 */