cleanup

5 files changed, 0 insertions, 1050 deletions

diff --git a/media-sound/audacity/Manifest b/media-sound/audacity/Manifest
index 206d7daedce9..d7f04630f865 100644
--- a/media-sound/audacity/Manifest
+++ b/media-sound/audacity/Manifest
@@ -1,9 +1,4 @@
 MD5 bf3f6434c87df998d331decbd97f2617 ChangeLog 3541
 MD5 5f8d028a16df282d4bef4395296c1ca9 audacity-1.1.3.ebuild 1297
-MD5 c151b63a0eca16e885c23226c03d28f0 files/audacity-1.0.0-gcc31.patch 2153
-MD5 f779279f19d249c6d735b28e72c30c6b files/audacity-src-1.1.0-phonograph.patch 5137
-MD5 f34b93cc26c85c6789d21fb94b213e09 files/audacity-src-1.1.0-timestretch.patch 19316
 MD5 d1064c1800175d5fb3113f74785568f6 files/basecfg-1.1.1 35
 MD5 cb2f75d37538e9688fe397cf178f57ad files/digest-audacity-1.1.3 68
-MD5 21b2333db59cc116a7e3c96563fe9ccd files/id3lib-3.8.0-gcc3.patch 2707
-MD5 8f5051d60b802ecf74497bc2ec4bb32e files/mono_mp3_export.patch 6261
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
deleted file mode 100644
index 4f2f6fa64efa..000000000000
--- a/media-sound/audacity/files/audacity-src-1.1.0-phonograph.patch
+++ /dev/null
@@ -1,199 +0,0 @@
-# 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
deleted file mode 100644
index d32bda37477a..000000000000
--- a/media-sound/audacity/files/audacity-src-1.1.0-timestretch.patch
+++ /dev/null
@@ -1,599 +0,0 @@
-# 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/id3lib-3.8.0-gcc3.patch b/media-sound/audacity/files/id3lib-3.8.0-gcc3.patch
deleted file mode 100644
index 12c8bc5ab37a..000000000000
--- a/media-sound/audacity/files/id3lib-3.8.0-gcc3.patch
+++ /dev/null
@@ -1,97 +0,0 @@
---- include/id3/id3lib_strings.h.orig	2002-08-21 16:41:11.000000000 -0600
-+++ include/id3/id3lib_strings.h	2002-08-21 16:40:32.000000000 -0600
-@@ -30,6 +30,94 @@
- 
- #include <string>
- 
-+#if (defined(__GNUC__) && (__GNUC__ >= 3) || (defined(_MSC_VER) && _MSC_VER > 1000))
-+namespace std
-+{
-+  template<>
-+    struct char_traits<unsigned char>
-+    {
-+      typedef unsigned char char_type;
-+      // Unsigned as wint_t in unsigned.
-+      typedef unsigned long  	int_type;
-+      typedef streampos 	pos_type;
-+      typedef streamoff 	off_type;
-+      typedef mbstate_t 	state_type;
-+      
-+      static void 
-+      assign(char_type& __c1, const char_type& __c2)
-+      { __c1 = __c2; }
-+
-+      static bool 
-+      eq(const char_type& __c1, const char_type& __c2)
-+      { return __c1 == __c2; }
-+
-+      static bool 
-+      lt(const char_type& __c1, const char_type& __c2)
-+      { return __c1 < __c2; }
-+
-+      static int 
-+      compare(const char_type* __s1, const char_type* __s2, size_t __n)
-+      { 
-+	for (size_t __i = 0; __i < __n; ++__i)
-+	  if (!eq(__s1[__i], __s2[__i]))
-+	    return lt(__s1[__i], __s2[__i]) ? -1 : 1;
-+	return 0; 
-+      }
-+
-+      static size_t
-+      length(const char_type* __s)
-+      { 
-+	const char_type* __p = __s; 
-+	while (__p) 
-+	  ++__p; 
-+	return (__p - __s); 
-+      }
-+
-+      static const char_type* 
-+      find(const char_type* __s, size_t __n, const char_type& __a)
-+      { 
-+	for (const char_type* __p = __s; size_t(__p - __s) < __n; ++__p)
-+	  if (*__p == __a) return __p;
-+	return 0;
-+      }
-+
-+      static char_type* 
-+      move(char_type* __s1, const char_type* __s2, size_t __n)
-+      { return (char_type*) memmove(__s1, __s2, __n * sizeof(char_type)); }
-+
-+      static char_type* 
-+      copy(char_type* __s1, const char_type* __s2, size_t __n)
-+      { return (char_type*) memcpy(__s1, __s2, __n * sizeof(char_type)); }
-+
-+      static char_type* 
-+      assign(char_type* __s, size_t __n, char_type __a)
-+      { 
-+	for (char_type* __p = __s; __p < __s + __n; ++__p) 
-+	  assign(*__p, __a);
-+        return __s; 
-+      }
-+
-+      static char_type 
-+      to_char_type(const int_type& __c)
-+      { return char_type(); }
-+
-+      static int_type 
-+      to_int_type(const char_type& __c) { return int_type(); }
-+
-+      static bool 
-+      eq_int_type(const int_type& __c1, const int_type& __c2)
-+      { return __c1 == __c2; }
-+
-+      static int_type 
-+      eof() { return static_cast<int_type>(-1); }
-+
-+      static int_type 
-+      not_eof(const int_type& __c)
-+      { return eq_int_type(__c, eof()) ? int_type(0) : __c; }
-+    };
-+} // namespace std
-+#endif
-+
- namespace dami
- {
-   typedef std::basic_string<char>           String;
diff --git a/media-sound/audacity/files/mono_mp3_export.patch b/media-sound/audacity/files/mono_mp3_export.patch
deleted file mode 100644
index 18bed1aeeb09..000000000000
--- a/media-sound/audacity/files/mono_mp3_export.patch
+++ /dev/null
@@ -1,150 +0,0 @@
-diff -ru audacity-src-1.1.1-3/src/export/ExportMP3.cpp /var/tmp/portage/audacity-1.1.1/work/audacity-src-1.1.1-3/src/export/ExportMP3.cpp
---- audacity-src-1.1.1-3/src/export/ExportMP3.cpp	2002-11-11 00:58:21.000000000 -0600
-+++ /var/tmp/portage/audacity-1.1.1/work/audacity-src-1.1.1-3/src/export/ExportMP3.cpp	2003-01-15 00:30:29.000000000 -0600
-@@ -190,6 +190,7 @@
-          /* function pointers to the symbols we get from the library */
-          lame_init_t* lame_init;
-          lame_init_params_t* lame_init_params;
-+         lame_encode_buffer_t* lame_encode_buffer;
-          lame_encode_buffer_interleaved_t* lame_encode_buffer_interleaved;
-          lame_encode_flush_t* lame_encode_flush;
-          lame_close_t* lame_close;
-@@ -259,6 +260,10 @@
-                                                                          "get_lame_version");
-             lame_init_params = 
-                (lame_init_params_t *) wxDllLoader::GetSymbol(libHandle, "lame_init_params");
-+
-+            lame_encode_buffer =
-+                (lame_encode_buffer_t *) wxDllLoader::GetSymbol(libHandle,
-+                                                          "lame_encode_buffer");
-             lame_encode_buffer_interleaved =
-                 (lame_encode_buffer_interleaved_t *) wxDllLoader::GetSymbol(libHandle,
-                                                           "lame_encode_buffer_interleaved");
-@@ -294,6 +299,7 @@
-             if (!lame_init ||
-                 !get_lame_version ||
-                 !lame_init_params ||
-+                !lame_encode_buffer ||
-                 !lame_encode_buffer_interleaved ||
-                 !lame_encode_flush ||
-                 !lame_close ||
-@@ -346,6 +352,22 @@
-             mOutBufferSize);
-       }
- 
-+
-+
-+      int EncodeBufferMono(short int inbuffer[], unsigned char outbuffer[]) {
-+         if(!mEncoding) return -1;
-+
-+         return lame_encode_buffer(mGF, inbuffer,inbuffer, mSamplesPerChunk,
-+            outbuffer, mOutBufferSize);
-+      }
-+
-+      int EncodeRemainderMono(short int inbuffer[], int nSamples,
-+                        unsigned char outbuffer[]) {
-+         return lame_encode_buffer(mGF, inbuffer, inbuffer, nSamples, outbuffer,
-+            mOutBufferSize);
-+      }
-+
-+
-       int FinishStream(unsigned char outbuffer[]) {
-          mEncoding = false;
-          int result = lame_encode_flush(mGF, outbuffer, mOutBufferSize);
-@@ -425,6 +447,7 @@
-          /* function pointers to the symbols we get from the library */
-          lame_init_t* lame_init;
-          lame_init_params_t* lame_init_params;
-+         lame_encode_buffer_t* lame_encode_buffer;
-          lame_encode_buffer_interleaved_t* lame_encode_buffer_interleaved;
-          lame_encode_flush_t* lame_encode_flush;
-          lame_close_t* lame_close;
-@@ -556,6 +579,10 @@
-             FindSymbol(connID, name, &mainAddr, &symClass);
-             lame_encode_buffer_interleaved = NewMachOFromCFM(mainAddr);
- 
-+            MakePString(name, "lame_encode_buffer");
-+            FindSymbol(connID, name, &mainAddr, &symClass);
-+            lame_encode_buffer = NewMachOFromCFM(mainAddr);
-+
-             MakePString(name, "lame_encode_flush");
-             FindSymbol(connID, name, &mainAddr, &symClass);
-             lame_encode_flush = NewMachOFromCFM(mainAddr);
-@@ -594,6 +621,7 @@
-             if (!lame_init ||
-                 !get_lame_version ||
-                 !lame_init_params ||
-+                !lame_encode_buffer ||
-                 !lame_encode_buffer_interleaved ||
-                 !lame_encode_flush ||
-                 !lame_close ||
-@@ -636,7 +664,7 @@
-       int EncodeBuffer(short int inbuffer[], unsigned char outbuffer[]) {
-          if(!mEncoding) return -1;
- 
--         return lame_encode_buffer_interleaved(mGF, inbuffer, mSamplesPerChunk,
-+         return lame_encode_buffer_interleaved(mGF, inbuffer, inbuffer, mSamplesPerChunk,
-             outbuffer, mOutBufferSize);
-       }
- 
-@@ -646,6 +674,19 @@
-             mOutBufferSize);
-       }
- 
-+      int EncodeBufferMono(short int inbuffer[], unsigned char outbuffer[]) {
-+         if(!mEncoding) return -1;
-+
-+         return lame_encode_buffer(mGF, inbuffer, inbuffer, mSamplesPerChunk,
-+            outbuffer, mOutBufferSize);
-+      }
-+
-+      int EncodeRemainderMono(short int inbuffer[], int nSamples,
-+                        unsigned char outbuffer[]) {
-+         return lame_encode_buffer(mGF, inbuffer, inbuffer , inbuffer, nSamples, outbuffer,
-+            mOutBufferSize);
-+      }
-+
-       int FinishStream(unsigned char outbuffer[]) {
-          mEncoding = false;
-          int result = lame_encode_flush(mGF, outbuffer, mOutBufferSize);
-@@ -1285,9 +1326,22 @@
-       short *mixed = (short *)mixer->GetBuffer();
- 
-       if(lastFrame)
--         bytes = GetMP3Exporter()->EncodeRemainder(mixed, numSamples, buffer);
--      else
--         bytes = GetMP3Exporter()->EncodeBuffer(mixed, buffer);
-+
-+	if (stereo) {
-+	  bytes = GetMP3Exporter()->EncodeRemainder(mixed, numSamples , buffer);
-+	}	   
-+	else{
-+	  bytes = GetMP3Exporter()->EncodeRemainderMono(mixed, numSamples , buffer);
-+	}
-+      else {
-+	
-+	if (stereo) {
-+	  bytes = GetMP3Exporter()->EncodeBuffer(mixed, buffer);
-+	}
-+	else { 
-+	  bytes = GetMP3Exporter()->EncodeBufferMono(mixed, buffer);
-+	}
-+      }
- 
-       outFile.Write(buffer, bytes);
- 
-diff -ru audacity-src-1.1.1-3/src/export/ExportMP3.h /var/tmp/portage/audacity-1.1.1/work/audacity-src-1.1.1-3/src/export/ExportMP3.h
---- audacity-src-1.1.1-3/src/export/ExportMP3.h	2002-11-11 00:58:21.000000000 -0600
-+++ /var/tmp/portage/audacity-1.1.1/work/audacity-src-1.1.1-3/src/export/ExportMP3.h	2003-01-15 00:30:29.000000000 -0600
-@@ -38,6 +38,11 @@
-       virtual int EncodeBuffer(short int inbuffer[], unsigned char outbuffer[]) = 0;
-       virtual int EncodeRemainder(short int inbuffer[], int nSamples,
-                                unsigned char outbuffer[]) = 0;
-+
-+      virtual int EncodeBufferMono(short int inbuffer[], unsigned char outbuffer[]) = 0;
-+      virtual int EncodeRemainderMono(short int inbuffer[], int nSamples,
-+                               unsigned char outbuffer[]) = 0;
-+
-       virtual int FinishStream(unsigned char outbuffer[]) = 0;
-       virtual void CancelEncoding() = 0;
-