Work in progress on bug #6701. The ebuild is masked for now.

4 files changed, 846 insertions, 0 deletions

diff --git a/media-sound/audacity/audacity-1.1.0.ebuild b/media-sound/audacity/audacity-1.1.0.ebuild
new file mode 100644
index 000000000000..6d9703aabab1
--- /dev/null
+++ b/media-sound/audacity/audacity-1.1.0.ebuild
@@ -0,0 +1,46 @@
+# Copyright 1999-2002 Gentoo Technologies, Inc.
+# Distributed under the terms of the GNU General Public License, v2 or later
+# $Header: /var/cvsroot/gentoo-x86/media-sound/audacity/audacity-1.1.0.ebuild,v 1.1 2002/08/27 22:46:01 agenkin Exp $
+
+DESCRIPTION="A free, crossplatform audio editor."
+HOMEPAGE="http://audacity.sourceforge.net/"
+LICENSE="GPL-2"
+
+# doesn't compile with wxGTK-2.3.2
+DEPEND="~x11-libs/wxGTK-2.2.9
+	oggvorbis? ( media-libs/libvorbis )
+	app-arch/zip
+	media-sound/mad
+	media-libs/libsndfile"
+	
+RDEPEND="${DEPEND}"
+
+SLOT="0"
+KEYWORDS="x86"
+SRC_URI="mirror://sourceforge/${PN}/${PN}-src-${PV}.tgz"
+S="${WORKDIR}/${PN}-src-${PV}"
+
+src_unpack() {
+	unpack "${PN}-src-${PV}.tgz"
+	## Patches from http://www.hcsw.org/audacity/
+	patch -p0 < "${FILESDIR}/${PN}-src-${PV}-timestretch.patch" || die
+	patch -p0 < "${FILESDIR}/${PN}-src-${PV}-phonograph.patch" || die
+}
+
+src_compile() {
+	local myconf
+	
+	myconf="--with-id3tag --with-libmad"
+	# vorbis breaks 4 me (rigo@home.nl)
+	# use oggvorbis && myconf="${myconf} --without-vorbis"
+	myconf="${myconf} --without-vorbis"
+
+	use arts && myconf="${myconf} --with-arts-soundserver"
+	
+	./configure --prefix=/usr $myconf || die
+	make || die
+}
+
+src_install () {
+	make PREFIX="${D}/usr" install || die
+}
diff --git a/media-sound/audacity/files/audacity-src-1.1.0-phonograph.patch b/media-sound/audacity/files/audacity-src-1.1.0-phonograph.patch
new file mode 100644
index 000000000000..4f2f6fa64efa
--- /dev/null
+++ b/media-sound/audacity/files/audacity-src-1.1.0-phonograph.patch
@@ -0,0 +1,199 @@
+# Audacity 1.1.0 Phonograph Patch
+#
+# By Doug Hoyte
+#
+# This patch adds an effect "Phonograph", which creates pops and crackles
+# that sound somewhat like a 33 1/3 LP in the selected audio
+#
+# Apply like so:
+#
+# cd audacity-1.1.0/
+# patch -p1 < /tmp/audacity-src-1.1.0-phonograph.patch
+#
+# Replace /tmp/ with wherever you put the patch, obviously.
+# configure, make, and enjoy.
+#
+#
+#
+diff -urNB audacity-src-1.1.0/src/Makefile.in audacity-src-1.1.0-phonograph/src/Makefile.in
+--- audacity-src-1.1.0/src/Makefile.in	Wed Jun  5 00:45:54 2002
++++ audacity-src-1.1.0-phonograph/src/Makefile.in	Fri Aug  9 16:32:29 2002
+@@ -76,6 +76,7 @@
+ 	$(OBJDIR)/effects/Invert.o \
+ 	$(OBJDIR)/effects/NoiseRemoval.o \
+ 	$(OBJDIR)/effects/Phaser.o \
++	$(OBJDIR)/effects/Phonograph.o \
+ 	$(OBJDIR)/effects/Reverse.o \
+ 	$(OBJDIR)/effects/Wahwah.o \
+ 	$(OBJDIR)/export/Export.o \
+diff -urNB audacity-src-1.1.0/src/effects/LoadEffects.cpp audacity-src-1.1.0-phonograph/src/effects/LoadEffects.cpp
+--- audacity-src-1.1.0/src/effects/LoadEffects.cpp	Wed Jun  5 00:45:54 2002
++++ audacity-src-1.1.0-phonograph/src/effects/LoadEffects.cpp	Fri Aug  9 16:33:33 2002
+@@ -21,6 +21,7 @@
+ #include "Invert.h"
+ #include "NoiseRemoval.h"
+ #include "Phaser.h"
++#include "Phonograph.h"
+ #include "Reverse.h"
+ #include "Wahwah.h"
+ 
+@@ -52,6 +53,7 @@
+    Effect::RegisterEffect(new EffectInvert(), false);
+    Effect::RegisterEffect(new EffectNoiseRemoval(), false);
+    Effect::RegisterEffect(new EffectPhaser(), false);
++   Effect::RegisterEffect(new EffectPhonograph(), false);
+    Effect::RegisterEffect(new EffectReverse(), false);
+    Effect::RegisterEffect(new EffectWahwah(), false);
+ 
+diff -urNB audacity-src-1.1.0/src/effects/Phonograph.cpp audacity-src-1.1.0-phonograph/src/effects/Phonograph.cpp
+--- audacity-src-1.1.0/src/effects/Phonograph.cpp	Wed Dec 31 16:00:00 1969
++++ audacity-src-1.1.0-phonograph/src/effects/Phonograph.cpp	Fri Aug  9 16:31:04 2002
+@@ -0,0 +1,89 @@
++/**********************************************************************
++
++  Audacity: A Digital Audio Editor
++
++  Phonograph.cpp
++
++  Doug Hoyte
++  
++  This class gives the selection the good ol' fashion vibe of a
++  33 1/3 LP.
++
++**********************************************************************/
++
++#include <stdlib.h>
++
++
++#include "Phonograph.h"
++#include "../WaveTrack.h"
++
++//
++// EffectPhonograph
++//
++
++EffectPhonograph::EffectPhonograph()
++{
++}
++
++bool EffectPhonograph::Process()
++{
++   TrackListIterator iter(mWaveTracks);
++   VTrack *t = iter.First();
++   int count = 0;
++   while(t) {
++      sampleCount start, len;
++      GetSamples((WaveTrack *)t, &start, &len);
++      bool success = ProcessOne(count, (WaveTrack *)t, start, len);
++      
++      if (!success)
++         return false;
++   
++      t = iter.Next();
++      count++;
++   }
++   
++   return true;
++}
++
++bool EffectPhonograph::ProcessOne(int count, WaveTrack *t,
++                               sampleCount start, sampleCount len)
++{
++   sampleCount base = start;
++   float *buf = new float[len];
++
++   int j,next,tpwidth=0;
++   float tpheight,tpoff;
++
++   
++   srand(time(NULL));
++
++   t->Get(buf, base, len);
++
++   next = 5500+(int) (7500.0*rand()/(RAND_MAX+1.0));
++   for (int i = 0; i < len; i++) {
++
++     next--;
++     if(next <= 0) {
++       next = 5500+(int) (7500.0*rand()/(RAND_MAX+1.0));
++       tpheight = (0.2*rand()/(RAND_MAX+1.0));
++       tpwidth = 7+(int) (50.0*rand()/(RAND_MAX+1.0));
++
++       if (next%2 == 0) tpheight *= -1;
++
++       for (j = 0; j < tpwidth; j++) {
++         tpoff = (0.1*rand()/(RAND_MAX+1.0));
++         tpoff *= tpheight;
++         if (i+j < len) buf[i+j] += tpheight + tpoff;
++       }
++
++     }
++
++ 
++   }
++
++   t->Set(buf, base, len);
++
++   delete[] buf;
++
++   return true;
++}
+diff -urNB audacity-src-1.1.0/src/effects/Phonograph.h audacity-src-1.1.0-phonograph/src/effects/Phonograph.h
+--- audacity-src-1.1.0/src/effects/Phonograph.h	Wed Dec 31 16:00:00 1969
++++ audacity-src-1.1.0-phonograph/src/effects/Phonograph.h	Fri Aug  9 16:31:04 2002
+@@ -0,0 +1,55 @@
++/**********************************************************************
++
++  Audacity: A Digital Audio Editor
++
++  Phonograph.h
++
++  Doug Hoyte
++  
++  This class gives the selection the good ol' fashion vibe of a
++  33 1/3 LP.
++
++**********************************************************************/
++
++#ifndef __AUDACITY_EFFECT_PHONOGRAPH__
++#define __AUDACITY_EFFECT_PHONOGRAPH__
++
++#include <wx/checkbox.h>
++#include <wx/button.h>
++#include <wx/dialog.h>
++#include <wx/stattext.h>
++#include <wx/slider.h>
++#include <wx/textctrl.h>
++#include <wx/sizer.h>
++#include <wx/intl.h>
++
++#include "Effect.h"
++
++#define __UNINITIALIZED__ (-1)
++
++class WaveTrack;
++
++class EffectPhonograph:public Effect {
++
++ public:
++   EffectPhonograph();
++
++   virtual wxString GetEffectName() {
++      return wxString(_("Phonograph"));
++   }
++   
++   virtual wxString GetEffectAction() {
++      return wxString(_("\"Phonographing\""));
++   }
++
++   virtual bool Process();
++
++ private:
++   bool ProcessOne(int count, WaveTrack * t,
++                   sampleCount start, sampleCount len);
++
++ private:
++
++};
++
++#endif
diff --git a/media-sound/audacity/files/audacity-src-1.1.0-timestretch.patch b/media-sound/audacity/files/audacity-src-1.1.0-timestretch.patch
new file mode 100644
index 000000000000..d32bda37477a
--- /dev/null
+++ b/media-sound/audacity/files/audacity-src-1.1.0-timestretch.patch
@@ -0,0 +1,599 @@
+# Audacity 1.1.0 TimeStretch Patch
+#
+# By Doug Hoyte
+#
+# Some code (smsPitchScale.cpp) is
+# COPYRIGHT 1999 Stephan M. Sprenger <sms@dspdimension.com>
+#
+#
+# This patch adds an effect that can "Stretch" the time of a selected
+# sample. It makes it longer or shorter depending on a ratio given.
+#
+# Also included is the experimental "Pitch Scaling" that will keep a
+# sample at the same pitch, even after being stretched.
+#
+#
+# Limitations:
+# - The GUI sucks. Ideally, I'd like it to be one window that has a slide
+#   bar and a checkbox. Also, it would show you the length of the selected
+#   sample (in seconds) and you could choose the desired length in seconds
+#   or via the ratio. Unfortunatley, my wxWindows skills suck...
+# - After being stretched, the current selection should be the new audio,
+#   not where the old audio was. This isn't a problem with other effects
+#   because they don't change length. I wasn't sure how to change the
+#   selection boundaries, so I just left it.
+# - The Pitch Scale code is SLOOOW.
+# - Sometimes the Pitch Scale code leaves a tiny bit of garbage at the
+#   beginning of a clip.
+# - The Pitch Scale code degrades the quality of the sample somewhat.
+#
+#
+# Apply like so:
+#
+# cd audacity-1.1.0/
+# patch -p1 < /tmp/audacity-src-1.1.0-timestretch.patch
+#
+# Replace /tmp/ with wherever you put the patch, obviously.
+# configure, make, and enjoy.
+#
+#
+#
+diff -urNb audacity-src-1.1.0/src/Makefile.in audacity-src-1.1.0-timestretch/src/Makefile.in
+--- audacity-src-1.1.0/src/Makefile.in	Wed Jun  5 00:45:54 2002
++++ audacity-src-1.1.0-timestretch/src/Makefile.in	Sat Aug 10 01:16:52 2002
+@@ -77,6 +77,8 @@
+ 	$(OBJDIR)/effects/NoiseRemoval.o \
+ 	$(OBJDIR)/effects/Phaser.o \
+ 	$(OBJDIR)/effects/Reverse.o \
++	$(OBJDIR)/effects/smsPitchScale.o \
++	$(OBJDIR)/effects/TimeStretch.o \
+ 	$(OBJDIR)/effects/Wahwah.o \
+ 	$(OBJDIR)/export/Export.o \
+ 	$(OBJDIR)/export/ExportMP3.o \
+diff -urNb audacity-src-1.1.0/src/effects/LoadEffects.cpp audacity-src-1.1.0-timestretch/src/effects/LoadEffects.cpp
+--- audacity-src-1.1.0/src/effects/LoadEffects.cpp	Wed Jun  5 00:45:54 2002
++++ audacity-src-1.1.0-timestretch/src/effects/LoadEffects.cpp	Sat Aug 10 01:17:56 2002
+@@ -22,6 +22,7 @@
+ #include "NoiseRemoval.h"
+ #include "Phaser.h"
+ #include "Reverse.h"
++#include "TimeStretch.h"
+ #include "Wahwah.h"
+ 
+ #ifdef USE_WAVELET
+@@ -53,6 +54,7 @@
+    Effect::RegisterEffect(new EffectNoiseRemoval(), false);
+    Effect::RegisterEffect(new EffectPhaser(), false);
+    Effect::RegisterEffect(new EffectReverse(), false);
++   Effect::RegisterEffect(new EffectTimeStretch(), false);
+    Effect::RegisterEffect(new EffectWahwah(), false);
+ 
+ #ifdef USE_WAVELET
+diff -urNb audacity-src-1.1.0/src/effects/TimeStretch.cpp audacity-src-1.1.0-timestretch/src/effects/TimeStretch.cpp
+--- audacity-src-1.1.0/src/effects/TimeStretch.cpp	Wed Dec 31 16:00:00 1969
++++ audacity-src-1.1.0-timestretch/src/effects/TimeStretch.cpp	Sat Aug 10 01:17:12 2002
+@@ -0,0 +1,137 @@
++/**********************************************************************
++
++  Audacity: A Digital Audio Editor
++
++  TimeStretch.cpp
++
++  Doug Hoyte
++  
++  This class is able to stretch a selection, making it take up more or
++  less time.
++
++**********************************************************************/
++
++
++#include <wx/generic/textdlgg.h>
++#include <wx/intl.h>
++
++#include "TimeStretch.h"
++#include "smsPitchScale.h"
++#include "../WaveTrack.h"
++
++//
++// EffectTimeStretch
++//
++
++EffectTimeStretch::EffectTimeStretch()
++{
++   ratio = 1.0;
++   pitchscale = false;
++}
++
++
++bool EffectTimeStretch::PromptUser()
++{
++   wxString temp;
++   wxString title = _("TimeStretch by Doug Hoyte");
++   wxString caption = _("Ratio (new length/orig length): ");
++   wxString default_value = wxString::Format("%f", ratio);
++
++   ratio = 1.0;
++   pitchscale = false;
++
++   temp = wxGetTextFromUser(caption, title,
++                            default_value, mParent, -1, -1, TRUE);
++   if (temp == "")
++      return false;
++   while (sscanf((const char *) temp, "%f", &ratio) < 0) {
++      caption = _("Please enter a positive number for the ratio: ");
++      temp = wxGetTextFromUser(caption, title,
++                               default_value, mParent, -1, -1, TRUE);
++      if (temp == "")
++         return false;
++   }
++
++   caption = _("Do you want pitch scaling (EXPERIMENTAL)? (y/n): ");
++   default_value = wxString::Format("n");
++   temp = wxGetTextFromUser(caption, title,
++                            default_value, mParent, -1, -1, TRUE);
++   if (temp == "")
++      return false;
++   while (temp != "n" && temp != "y") {
++      caption = _("Please enter y or n for the pitch scaling: ");
++      temp = wxGetTextFromUser(caption, title,
++                               default_value, mParent, -1, -1, TRUE);
++      if (temp == "")
++         return false;
++   }
++
++   if (temp == "y") pitchscale = true;
++     else pitchscale = false;
++
++
++   return true;
++}
++
++
++
++bool EffectTimeStretch::Process()
++{
++   TrackListIterator iter(mWaveTracks);
++   VTrack *t = iter.First();
++   int count = 0;
++   while(t) {
++      sampleCount start, len;
++      GetSamples((WaveTrack *)t, &start, &len);
++      bool success = ProcessOne(count, (WaveTrack *)t, start, len);
++      
++      if (!success)
++         return false;
++   
++      t = iter.Next();
++      count++;
++   }
++   
++   return true;
++}
++
++
++
++bool EffectTimeStretch::ProcessOne(int count, WaveTrack *t,
++                               sampleCount base, sampleCount len)
++{
++   int newlen=(int) (len*ratio) + 1;
++   double basesecs=0;
++   float *inbuf = new float[len];
++   float *outbuf = new float[newlen];
++   int i,nextloc,currloc=0;
++
++
++   if (ratio == 1) goto bail;
++
++   basesecs = base / (t->GetRate());
++
++   t->Get(inbuf, base, len);
++
++   for (i=0; i<len-1; i++) {
++     if (i%10000 == 0) TrackProgress(count, 2*i/(double)len);
++     nextloc = (int) (((float)i+1) * ratio);
++     for (; currloc < nextloc; currloc++) outbuf[currloc] = inbuf[i];
++     outbuf[nextloc] = inbuf[i+1];
++   }
++
++   if (ratio > 1) t->InsertSilence(basesecs, (newlen-len)/(t->GetRate()) );
++   if (ratio < 1) t->Clear(basesecs, basesecs+((len-newlen) / (t->GetRate())) );
++
++
++   if (pitchscale == true) smsPitchScale(ratio, newlen, 2048, 4, t->GetRate(), outbuf, outbuf);
++
++   t->Set(outbuf, base, newlen);
++
++   bail:
++
++   delete[] inbuf;
++   delete[] outbuf;
++
++   return true;
++}
+diff -urNb audacity-src-1.1.0/src/effects/TimeStretch.h audacity-src-1.1.0-timestretch/src/effects/TimeStretch.h
+--- audacity-src-1.1.0/src/effects/TimeStretch.h	Wed Dec 31 16:00:00 1969
++++ audacity-src-1.1.0-timestretch/src/effects/TimeStretch.h	Sat Aug 10 01:17:12 2002
+@@ -0,0 +1,62 @@
++/**********************************************************************
++
++  Audacity: A Digital Audio Editor
++
++  TimeStretch.h
++
++  Doug Hoyte
++  
++  This class is able to stretch a selection, making it take up more or
++  less time.
++
++**********************************************************************/
++
++#ifndef __AUDACITY_EFFECT_TIMESTRETCH__
++#define __AUDACITY_EFFECT_TIMESTRETCH__
++
++class wxString;
++
++#include <wx/checkbox.h>
++#include <wx/button.h>
++#include <wx/dialog.h>
++#include <wx/stattext.h>
++#include <wx/slider.h>
++#include <wx/textctrl.h>
++#include <wx/sizer.h>
++#include <wx/intl.h>
++
++#include "Effect.h"
++#include "smsPitchScale.h"
++
++
++
++#define __UNINITIALIZED__ (-1)
++
++class WaveTrack;
++
++class EffectTimeStretch:public Effect {
++
++ public:
++   EffectTimeStretch();
++
++   virtual wxString GetEffectName() {
++      return wxString(_("TimeStretch"));
++   }
++   
++   virtual wxString GetEffectAction() {
++      return wxString(_("Stretching Time"));
++   }
++
++   virtual bool PromptUser();
++
++   virtual bool Process();
++
++ private:
++   bool ProcessOne(int count, WaveTrack * t, sampleCount start, sampleCount len);
++
++
++   float ratio;
++   bool pitchscale;
++};
++
++#endif
+diff -urNb audacity-src-1.1.0/src/effects/smsPitchScale.cpp audacity-src-1.1.0-timestretch/src/effects/smsPitchScale.cpp
+--- audacity-src-1.1.0/src/effects/smsPitchScale.cpp	Wed Dec 31 16:00:00 1969
++++ audacity-src-1.1.0-timestretch/src/effects/smsPitchScale.cpp	Sat Aug 10 01:17:18 2002
+@@ -0,0 +1,280 @@
++/****************************************************************************
++*
++* NAME: smsPitchScale.cpp
++* VERSION: 1.01
++* HOME URL: http://www.dspdimension.com
++* KNOWN BUGS: none
++*
++* SYNOPSIS: Routine for doing pitch scaling while maintaining
++* duration using the Short Time Fourier Transform.
++*
++* DESCRIPTION: The routine takes a pitchScale factor value which is between 0.5
++* (one octave down) and 2. (one octave up). A value of exactly 1 does not change
++* the pitch. numSampsToProcess tells the routine how many samples in indata[0...
++* numSampsToProcess-1] should be pitch scaled and moved to outdata[0 ...
++* numSampsToProcess-1]. The two buffers can be identical (ie. it can process the
++* data in-place). fftFrameSize defines the FFT frame size used for the
++* processing. Typical values are 1024, 2048 and 4096. It may be any value <=
++* MAX_FFT_FRAME_LENGTH but it MUST be a power of 2. osamp is the STFT
++* oversampling factor which also determines the overlap between adjacent STFT
++* frames. It should at least be 4 for moderate scaling ratios. A value of 32 is
++* recommended for best quality. sampleRate takes the sample rate for the signal 
++* in unit Hz, ie. 44100 for 44.1 kHz audio. The data passed to the routine in 
++* indata[] should be in the range [-1.0, 1.0), which is also the output range 
++* for the data. 
++*
++* COPYRIGHT 1999 Stephan M. Sprenger <sms@dspdimension.com>
++*
++* 						The Wide Open License (WOL)
++*
++* Permission to use, copy, modify, distribute and sell this software and its
++* documentation for any purpose is hereby granted without fee, provided that
++* the above copyright notice and this license appear in all source copies. 
++* THIS SOFTWARE IS PROVIDED "AS IS" WITHOUT EXPRESS OR IMPLIED WARRANTY OF
++* ANY KIND. See http://www.dspguru.com/wol.htm for more information.
++*
++*****************************************************************************/ 
++
++#include <string.h>
++#include <math.h>
++
++#include "smsPitchScale.h"
++
++
++void smsPitchScale(float pitchScale, long numSampsToProcess, long fftFrameSize, long osamp, float sampleRate, float *indata, float *outdata)
++/*
++	Routine smsPitchScale(). See top of file for explanation
++	Purpose: doing pitch scaling while maintaining duration using the Short
++	Time Fourier Transform.
++	Author: (c)1999 Stephan M. Sprenger <sms@dspdimension.com>
++*/
++{
++
++	static float gInFIFO[MAX_FRAME_LENGTH];
++	static float gOutFIFO[MAX_FRAME_LENGTH];
++	static float gFFTworksp[2*MAX_FRAME_LENGTH];
++	static float gLastPhase[MAX_FRAME_LENGTH/2];
++	static float gSumPhase[MAX_FRAME_LENGTH/2];
++	static float gOutputAccum[2*MAX_FRAME_LENGTH];
++	static float gAnaFreq[MAX_FRAME_LENGTH];
++	static float gAnaMagn[MAX_FRAME_LENGTH];
++	static float gSynFreq[MAX_FRAME_LENGTH];
++	static float gSynMagn[MAX_FRAME_LENGTH];
++	static long gRover = false, gInit = false;
++	double magn, phase, tmp, window, real, imag;
++	double freqPerBin, expct, fadeZoneLen;
++	long i,k, qpd, index, inFifoLatency, stepSize, fftFrameSize2;
++
++	/* set up some handy variables */
++	fadeZoneLen = fftFrameSize/2;
++	fftFrameSize2 = fftFrameSize/2;
++	stepSize = fftFrameSize/osamp;
++	freqPerBin = sampleRate/(double)fftFrameSize;
++	expct = 2.*M_PI*(double)stepSize/(double)fftFrameSize;
++	inFifoLatency = fftFrameSize-stepSize;
++	if (gRover == false) gRover = inFifoLatency;
++
++	/* initialize our static arrays */
++	if (gInit == false) {
++		memset(gInFIFO, 0, MAX_FRAME_LENGTH*sizeof(float));
++		memset(gOutFIFO, 0, MAX_FRAME_LENGTH*sizeof(float));
++		memset(gFFTworksp, 0, 2*MAX_FRAME_LENGTH*sizeof(float));
++		memset(gLastPhase, 0, MAX_FRAME_LENGTH*sizeof(float)/2);
++		memset(gSumPhase, 0, MAX_FRAME_LENGTH*sizeof(float)/2);
++		memset(gOutputAccum, 0, 2*MAX_FRAME_LENGTH*sizeof(float));
++		memset(gAnaFreq, 0, MAX_FRAME_LENGTH*sizeof(float));
++		memset(gAnaMagn, 0, MAX_FRAME_LENGTH*sizeof(float));
++		gInit = true;
++	}
++
++	/* main processing loop */
++	for (i = 0; i < numSampsToProcess; i++){
++
++		/* As long as we have not yet collected enough data just read in */
++		gInFIFO[gRover] = indata[i];
++		outdata[i] = gOutFIFO[gRover-inFifoLatency];
++		gRover++;
++
++		/* now we have enough data for processing */
++		if (gRover >= fftFrameSize) {
++			gRover = inFifoLatency;
++
++			/* do windowing and re,im interleave */
++			for (k = 0; k < fftFrameSize;k++) {
++				window = -.5*cos(2.*M_PI*(double)k/(double)fftFrameSize)+.5;
++				gFFTworksp[2*k] = gInFIFO[k] * window;
++				gFFTworksp[2*k+1] = 0.;
++			}
++
++
++			/* ***************** ANALYSIS ******************* */
++			/* do transform */
++			smsFft(gFFTworksp, fftFrameSize, -1);
++
++			/* this is the analysis step */
++			for (k = 0; k <= fftFrameSize2; k++) {
++
++				/* de-interlace FFT buffer */
++				real = gFFTworksp[2*k];
++				imag = gFFTworksp[2*k+1];
++
++				/* compute magnitude and phase */
++				magn = 2.*sqrt(real*real + imag*imag);
++				phase = atan2(imag,real);
++
++				/* compute phase difference */
++				tmp = phase - gLastPhase[k];
++				gLastPhase[k] = phase;
++
++				/* subtract expected phase difference */
++				tmp -= (double)k*expct;
++
++				/* map delta phase into +/- Pi interval */
++				qpd = (long int) (tmp/M_PI);
++				if (qpd >= 0) qpd += qpd&1;
++				else qpd -= qpd&1;
++				tmp -= M_PI*(double)qpd;
++
++				/* get deviation from bin frequency from the +/- Pi interval */
++				tmp = osamp*tmp/(2.*M_PI);
++
++				/* compute the k-th partials' true frequency */
++				tmp = (double)k*freqPerBin + tmp*freqPerBin;
++
++				/* store magnitude and true frequency in analysis arrays */
++				gAnaMagn[k] = magn;
++				gAnaFreq[k] = tmp;
++
++			}
++
++
++
++			/* ***************** PROCESSING ******************* */
++			/* this does the actual pitch scaling */
++			memset(gSynMagn, 0, fftFrameSize*sizeof(float));
++			memset(gSynFreq, 0, fftFrameSize*sizeof(float));
++			for (k = 0; k <= fftFrameSize2; k++) {
++				index = (long int) (k/pitchScale);
++				if (index <= fftFrameSize2) {
++					/* new bin overrides existing if magnitude is higher */ 
++					if (gAnaMagn[index] > gSynMagn[k]) {
++						gSynMagn[k] = gAnaMagn[index];
++						gSynFreq[k] = gAnaFreq[index] * pitchScale;
++					}
++					/* fill empty bins with nearest neighbour */
++					if ((gSynFreq[k] == 0.) && (k > 0)) {
++						gSynFreq[k] = gSynFreq[k-1];
++						gSynMagn[k] = gSynMagn[k-1];
++					}
++				}
++			}
++
++
++			/* ***************** SYNTHESIS ******************* */
++			/* this is the synthesis step */
++			for (k = 0; k <= fftFrameSize2; k++) {
++
++				/* get magnitude and true frequency from synthesis arrays */
++				magn = gSynMagn[k];
++				tmp = gSynFreq[k];
++
++				/* subtract bin mid frequency */
++				tmp -= (double)k*freqPerBin;
++
++				/* get bin deviation from freq deviation */
++				tmp /= freqPerBin;
++
++				/* take osamp into account */
++				tmp = 2.*M_PI*tmp/osamp;
++
++				/* add the overlap phase advance back in */
++				tmp += (double)k*expct;
++
++				/* accumulate delta phase to get bin phase */
++				gSumPhase[k] += tmp;
++				phase = gSumPhase[k];
++
++				/* get real and imag part and re-interleave */
++				gFFTworksp[2*k] = magn*cos(phase);
++				gFFTworksp[2*k+1] = magn*sin(phase);
++			} 
++
++			/* zero negative frequencies */
++			for (k = fftFrameSize+2; k < 2*fftFrameSize; k++) gFFTworksp[k] = 0.;
++
++			/* do inverse transform */
++			smsFft(gFFTworksp, fftFrameSize, 1);
++
++			/* do windowing and add to output accumulator */ 
++			for(k=0; k < fftFrameSize; k++) {
++				window = -.5*cos(2.*M_PI*(double)k/(double)fftFrameSize)+.5;
++				gOutputAccum[k] += 2.*window*gFFTworksp[2*k]/(fftFrameSize2*osamp);
++			}
++			for (k = 0; k < stepSize; k++) gOutFIFO[k] = gOutputAccum[k];
++
++			/* shift accumulator */
++			memmove(gOutputAccum, gOutputAccum+stepSize, fftFrameSize*sizeof(float));
++
++			/* move input FIFO */
++			for (k = 0; k < inFifoLatency; k++) gInFIFO[k] = gInFIFO[k+stepSize];
++		}
++	}
++}
++
++
++
++void smsFft(float *fftBuffer, long fftFrameSize, long sign)
++/* 
++	FFT routine, (C)1996 S.M.Sprenger. Sign = -1 is FFT, 1 is iFFT (inverse)
++	Fills fftBuffer[0...2*fftFrameSize-1] with the Fourier transform of the
++	time domain data in fftBuffer[0...2*fftFrameSize-1]. The FFT array takes
++	and returns the cosine and sine parts in an interleaved manner, ie.
++	fftBuffer[0] = cosPart[0], fftBuffer[1] = sinPart[0], asf. fftFrameSize
++	must be a power of 2. It expects a complex input signal (see footnote 2),
++	ie. when working with 'common' audio signals our input signal has to be
++	passed as {in[0],0.,in[1],0.,in[2],0.,...} asf. In that case, the transform
++	of the frequencies of interest is in fftBuffer[0...fftFrameSize].
++*/
++{
++	float wr, wi, arg, *p1, *p2, temp;
++	float tr, ti, ur, ui, *p1r, *p1i, *p2r, *p2i;
++	long i, bitm, j, le, le2, k;
++
++	for (i = 2; i < 2*fftFrameSize-2; i += 2) {
++		for (bitm = 2, j = 0; bitm < 2*fftFrameSize; bitm <<= 1) {
++			if (i & bitm) j++;
++			j <<= 1;
++		}
++		if (i < j) {
++			p1 = fftBuffer+i; p2 = fftBuffer+j;
++			temp = *p1; *(p1++) = *p2;
++			*(p2++) = temp; temp = *p1;
++			*p1 = *p2; *p2 = temp;
++		}
++	}
++	for (k = 0, le = 2; k < log(fftFrameSize)/log(2.); k++) {
++		le <<= 1;
++		le2 = le>>1;
++		ur = 1.0;
++		ui = 0.0;
++		arg = M_PI / (le2>>1);
++		wr = cos(arg);
++		wi = sign*sin(arg);
++		for (j = 0; j < le2; j += 2) {
++			p1r = fftBuffer+j; p1i = p1r+1;
++			p2r = p1r+le2; p2i = p2r+1;
++			for (i = j; i < 2*fftFrameSize; i += le) {
++				tr = *p2r * ur - *p2i * ui;
++				ti = *p2r * ui + *p2i * ur;
++				*p2r = *p1r - tr; *p2i = *p1i - ti;
++				*p1r += tr; *p1i += ti;
++				p1r += le; p1i += le;
++				p2r += le; p2i += le;
++			}
++			tr = ur*wr - ui*wi;
++			ui = ur*wi + ui*wr;
++			ur = tr;
++		}
++	}
++}
++
+diff -urNb audacity-src-1.1.0/src/effects/smsPitchScale.h audacity-src-1.1.0-timestretch/src/effects/smsPitchScale.h
+--- audacity-src-1.1.0/src/effects/smsPitchScale.h	Wed Dec 31 16:00:00 1969
++++ audacity-src-1.1.0-timestretch/src/effects/smsPitchScale.h	Sat Aug 10 01:17:18 2002
+@@ -0,0 +1,33 @@
++/**********************************************************************
++
++  Audacity: A Digital Audio Editor
++
++  smsPitchScale.cpp
++
++  Doug Hoyte
++
++  These functions are used by the TimeStretch class in order to provide
++  pitch scaling. Parts of these functions may have been modified by me.
++
++  smsPitchScale.cpp and smsPitchScale.h are...
++
++* COPYRIGHT 1999 Stephan M. Sprenger <sms@dspdimension.com>
++*
++*                                               The Wide Open License (WOL)
++*
++* Permission to use, copy, modify, distribute and sell this software and its
++* documentation for any purpose is hereby granted without fee, provided that
++* the above copyright notice and this license appear in all source copies.
++* THIS SOFTWARE IS PROVIDED "AS IS" WITHOUT EXPRESS OR IMPLIED WARRANTY OF
++* ANY KIND. See http://www.dspguru.com/wol.htm for more information.
++*
++*****************************************************************************/
++
++
++#define M_PI 3.14159265358979323846
++#define MAX_FRAME_LENGTH 8192
++
++void smsFft(float *fftBuffer, long fftFrameSize, long sign);
++
++void smsPitchScale(float pitchScale, long numSampsToProcess, long fftFrameSize,
++long osamp, float sampleRate, float *indata, float *outdata);
diff --git a/media-sound/audacity/files/digest-audacity-1.1.0 b/media-sound/audacity/files/digest-audacity-1.1.0
new file mode 100644
index 000000000000..f884593e5372
--- /dev/null
+++ b/media-sound/audacity/files/digest-audacity-1.1.0
@@ -0,0 +1,2 @@
+MD5 92896e20912c40e0fcc9e605bd8d660a audacity-src-1.1.0.tgz 1490458
+MD5 9d5947e2c25c846b03faadfb8c6e8601 id3lib-3.8.0.tar.gz 934333