diff audioio/AudioPortAudioTarget.cpp @ 59:bf1a53489ccc

...

--- a/audioio/AudioPortAudioTarget.cpp	Sun Nov 11 20:31:12 2007 +0000
+++ b/audioio/AudioPortAudioTarget.cpp	Sun Nov 11 20:34:41 2007 +0000
@@ -1,254 +1,254 @@
-/* -*- 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 */
-
+/* -*- 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 */
+