remove old, use eqmake4 thanks Davide Pesavento <davidepesa@gmail.com> in bug 189641

(Portage version: 2.1.3.7)

11 files changed, 28 insertions, 2443 deletions

diff --git a/media-sound/lastfmplayer/ChangeLog b/media-sound/lastfmplayer/ChangeLog
index c2aa52063bd0..cd6ea6c0b899 100644
--- a/media-sound/lastfmplayer/ChangeLog
+++ b/media-sound/lastfmplayer/ChangeLog
@@ -1,6 +1,16 @@
 # ChangeLog for media-sound/lastfmplayer
 # Copyright 1999-2007 Gentoo Foundation; Distributed under the GPL v2
-# $Header: /var/cvsroot/gentoo-x86/media-sound/lastfmplayer/ChangeLog,v 1.22 2007/07/27 20:06:20 peper Exp $
+# $Header: /var/cvsroot/gentoo-x86/media-sound/lastfmplayer/ChangeLog,v 1.23 2007/08/29 18:30:00 genstef Exp $
+
+*lastfmplayer-1.3.1.0.ebuild (29 Aug 2007)
+
+  29 Aug 2007; Stefan Schweizer <genstef@gentoo.org>
+  -files/lastfmplayer-1.1.3-mp3transcode.patch,
+  -files/lastfmplayer-amd64.patch, -files/lastfmplayer-1.1.3-pic.patch,
+  -files/13_alsa-r1.diff, -files/qt4.patch, -lastfmplayer-1.1.3-r2.ebuild,
+  -lastfmplayer-1.3.0.58.ebuild, lastfmplayer-1.3.1.0.ebuild:
+  remove old, use eqmake4 thanks Davide Pesavento <davidepesa@gmail.com> in
+  bug 189641
 
 *lastfmplayer-1.3.1.0 (27 Jul 2007)
 
diff --git a/media-sound/lastfmplayer/files/13_alsa-r1.diff b/media-sound/lastfmplayer/files/13_alsa-r1.diff
deleted file mode 100644
index e5289d3966fa..000000000000
--- a/media-sound/lastfmplayer/files/13_alsa-r1.diff
+++ /dev/null
@@ -1,2186 +0,0 @@
-diff -urN last.fm-1.1.3.orig/LastFM.pro last.fm-1.1.3/LastFM.pro
---- last.fm-1.1.3.orig/LastFM.pro	2007-03-18 20:57:23.000000000 -0700
-+++ last.fm-1.1.3/LastFM.pro	2007-03-18 20:59:36.000000000 -0700
-@@ -5,8 +5,19 @@
-            src/webservice/ src/settingsservice/ \
-            src/httpinput/ \
-            src/mp3transcode/ \
--           src/rtaudioplayback/ \
-            src/metadataextension/ src/sidebarextension/ src/searchextension/ src/userinfoextension/ \
-            src/mediadevices/itunes/
- 
- TRANSLATIONS = i18n/lastfm_jp_JP.ts
-+
-+win32 {
-+    SUBDIRS += src/rtaudioplayback/
-+}
-+
-+unix:!linux-g++ {
-+    SUBDIRS += src/rtaudioplayback/
-+}
-+
-+unix:linux-g++ {
-+    SUBDIRS += src/alsaplayback/
-+}
-diff -urN last.fm-1.1.3.orig/src/alsaplayback/alsaaudio.cpp last.fm-1.1.3/src/alsaplayback/alsaaudio.cpp
---- last.fm-1.1.3.orig/src/alsaplayback/alsaaudio.cpp	1969-12-31 16:00:00.000000000 -0800
-+++ last.fm-1.1.3/src/alsaplayback/alsaaudio.cpp	2007-03-18 20:57:28.000000000 -0700
-@@ -0,0 +1,851 @@
-+/***************************************************************************
-+ *   Copyright (C) 2007 by John Stamp                                      *
-+ *   jstamp@users.sourceforge.net                                          *
-+ *                                                                         *
-+ *   Large portions of this code are shamelessly copied from audio.c:      *
-+ *   The XMMS ALSA output plugin                                           *
-+ *   Copyright (C) 2001-2003 Matthieu Sozeau <mattam@altern.org>           *
-+ *   Copyright (C) 1998-2003  Peter Alm, Mikael Alm, Olle Hallnas,         *
-+ *                            Thomas Nilsson and 4Front Technologies       *
-+ *   Copyright (C) 1999-2006  Haavard Kvaalen                              *
-+ *   Copyright (C) 2005       Takashi Iwai                                 *
-+ *                                                                         *
-+ *   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.                                   *
-+ *                                                                         *
-+ *   This program is distributed in the hope that it will be useful,       *
-+ *   but WITHOUT ANY WARRANTY; without even the implied warranty of        *
-+ *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the         *
-+ *   GNU General Public License for more details.                          *
-+ *                                                                         *
-+ *   You should have received a copy of the GNU General Public License     *
-+ *   along with this program; if not, write to the                         *
-+ *   Free Software Foundation, Inc.,                                       *
-+ *   59 Temple Place - Suite 330, Boston, MA  02111-1307, USA.             *
-+ ***************************************************************************/
-+#include "alsaaudio.h"
-+#include "Loqqer.h"
-+#include <qendian.h>
-+
-+#include <QDebug>
-+
-+#define MIN( a, b )  ( ( ( a ) < ( b ) ) ? ( a ) : ( b ) )
-+#define MAX_BUFFER_SIZE 5
-+
-+QMutex AlsaAudio::mutex;
-+pthread_t AlsaAudio::audio_thread;
-+
-+QByteArray AlsaAudio::audioData;
-+snd_output_t* AlsaAudio::logs = NULL;
-+bool AlsaAudio::going = false;
-+snd_pcm_t *AlsaAudio::alsa_pcm = NULL;
-+
-+int AlsaAudio::hw_period_size_in = 0;
-+snd_format* AlsaAudio::inputf = NULL;
-+snd_format* AlsaAudio::outputf = NULL;
-+float AlsaAudio::volume = 1.0;
-+
-+convert_func_t         AlsaAudio::alsa_convert_func = NULL;
-+convert_channel_func_t AlsaAudio::alsa_stereo_convert_func =NULL;
-+convert_freq_func_t    AlsaAudio::alsa_frequency_convert_func =NULL;
-+xmms_convert_buffers   *AlsaAudio::convertb = NULL;
-+bool AlsaAudio::use_mmap = false;
-+
-+
-+AlsaAudio::AlsaAudio()
-+{
-+    maxBufferSize = 0;
-+}
-+
-+
-+AlsaAudio::~AlsaAudio()
-+{
-+}
-+
-+
-+/******************************************************************************
-+    Device Detection
-+******************************************************************************/
-+
-+int AlsaAudio::getCards( void )
-+{
-+    int card = -1;
-+    int err = 0;
-+    _devices.clear();
-+
-+    if ( ( err = snd_card_next( &card ) ) != 0 )
-+    {
-+        LOGL( 1, "AlsaAudio::getCards() failed: " << snd_strerror( -err ) );
-+        return -1;
-+    }
-+
-+    while ( card > -1 )
-+    {
-+        getDevicesForCard( card );
-+        if ( ( err = snd_card_next( &card ) ) != 0 )
-+        {
-+            LOGL( 1, "AlsaAudio::getCards() failed: " << snd_strerror( -err ) );
-+            return -1;
-+        }
-+    }
-+    return _devices.size();
-+}
-+
-+
-+void AlsaAudio::getDevicesForCard( int card )
-+{
-+    int pcm_device = -1, err;
-+    snd_pcm_info_t *pcm_info;
-+    snd_ctl_t *ctl;
-+    char devName[64], *card_name;
-+
-+    sprintf( devName, "hw:%i", card );
-+
-+    if ( ( err = snd_ctl_open( &ctl, devName, 0 ) ) < 0 )
-+    {
-+        LOGL( 1, "AlsaAudio::getDevicesForCard() failed: " << snd_strerror( -err ) );
-+        return;
-+    }
-+
-+    if ( ( err = snd_card_get_name( card, &card_name ) ) != 0 )
-+    {
-+        LOGL( 1, "AlsaAudio::getDevicesForCard() failed: " << snd_strerror( -err ) );
-+        card_name = "Unknown soundcard";
-+    }
-+
-+    // Each card has its own default device
-+    // But test, just to be sure it's there
-+    AlsaDeviceInfo dev;
-+    dev.name = QString( "%1: Default Device (default:%2)" ).arg( card_name ).arg( card );
-+    dev.device = QString( "default:%1" ).arg( card );
-+    snd_pcm_t *test_pcm;
-+    err = snd_pcm_open( &test_pcm, dev.device.toStdString().c_str(), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK );
-+    if ( err >= 0 )
-+        snd_pcm_close( test_pcm );
-+    if ( err == 0 || err == -EBUSY )
-+        _devices.push_back( dev );
-+
-+    snd_pcm_info_alloca( &pcm_info );
-+
-+    for ( ;; )
-+    {
-+        char device[64], descr[128];
-+        if ( ( err = snd_ctl_pcm_next_device( ctl, &pcm_device ) ) < 0 )
-+        {
-+            LOGL( 1, "AlsaAudio::getDevicesForCard() failed: " << snd_strerror( -err ) );
-+            pcm_device = -1;
-+        }
-+        if ( pcm_device < 0 )
-+            break;
-+
-+        snd_pcm_info_set_device( pcm_info, pcm_device );
-+        snd_pcm_info_set_subdevice( pcm_info, 0 );
-+        snd_pcm_info_set_stream( pcm_info, SND_PCM_STREAM_PLAYBACK );
-+
-+        if ( ( err = snd_ctl_pcm_info( ctl, pcm_info ) ) < 0 )
-+        {
-+            if ( err != -ENOENT )
-+                LOGL( 1, "AlsaAudio::getDevicesForCard: snd_ctl_pcm_info() failed ("
-+                         << card << ":" << pcm_device << "): " << snd_strerror( -err ) );
-+            continue;
-+        }
-+
-+        sprintf( device, "hw:%d,%d", card, pcm_device );
-+        sprintf( descr, "%s: %s (%s)", card_name,
-+                                       snd_pcm_info_get_name( pcm_info ),
-+                                       device );
-+        dev.name = descr;
-+        dev.device = device;
-+        _devices.push_back( dev );
-+    }
-+
-+    snd_ctl_close( ctl );
-+}
-+
-+
-+AlsaDeviceInfo AlsaAudio::getDeviceInfo( int device )
-+{
-+    return _devices[device];
-+}
-+
-+
-+/******************************************************************************
-+    Device Setup
-+******************************************************************************/
-+
-+bool AlsaAudio::alsaSetup( QString device, snd_pcm_uframes_t periodSize, uint periodCount, snd_format *f )
-+{
-+    int err, hw_period_size;
-+    snd_pcm_hw_params_t *hwparams;
-+    snd_pcm_sw_params_t *swparams;
-+    snd_pcm_uframes_t alsa_buffer_size, alsa_period_size;
-+    snd_pcm_access_mask_t *mask;
-+
-+#ifndef QT_NO_DEBUG
-+    qDebug() << "AlsaAudio::alsaSetup()";
-+    snd_output_stdio_attach( &logs, stderr, 0 );
-+#endif
-+
-+    alsa_convert_func = NULL;
-+    alsa_stereo_convert_func = NULL;
-+    alsa_frequency_convert_func = NULL;
-+
-+    free( outputf );
-+    outputf = snd_format_from_xmms( f->xmms_format, f->rate, f->channels );
-+
-+#ifndef QT_NO_DEBUG
-+    qDebug() << "Opening device:" << device;
-+#endif
-+    // FIXME: Can snd_pcm_open() return EAGAIN?
-+    if ( ( err = snd_pcm_open( &alsa_pcm, device.toStdString().c_str(),
-+                 SND_PCM_STREAM_PLAYBACK,
-+                 SND_PCM_NONBLOCK ) ) < 0 )
-+    {
-+        LOGL( 1, "AlsaAudio::alsaSetup(): Failed to open pcm device (" << device << "): " << snd_strerror( -err ) );
-+        alsa_pcm = NULL;
-+        free( outputf );
-+        outputf = NULL;
-+        return false;
-+    }
-+
-+#ifndef QT_NO_DEBUG
-+    snd_pcm_info_t *info;
-+    int alsa_card, alsa_device, alsa_subdevice;
-+
-+    snd_pcm_info_alloca( &info );
-+    snd_pcm_info( alsa_pcm, info );
-+    alsa_card = snd_pcm_info_get_card( info );
-+    alsa_device = snd_pcm_info_get_device( info );
-+    alsa_subdevice = snd_pcm_info_get_subdevice( info );
-+    qDebug() << "Card:" << alsa_card << "Device:" << alsa_device << "Subdevice:" << alsa_subdevice;
-+#endif
-+
-+    snd_pcm_hw_params_alloca( &hwparams );
-+
-+    if ( ( err = snd_pcm_hw_params_any( alsa_pcm, hwparams ) ) < 0 )
-+    {
-+        LOGL( 1, "AlsaAudio::alsaSetup(): No configuration available for playback: " << snd_strerror( -err ) );
-+        return false;
-+    }
-+
-+    // First try to set up mmapped access
-+    mask = (snd_pcm_access_mask_t*)alloca(snd_pcm_access_mask_sizeof());
-+    snd_pcm_access_mask_none(mask);
-+    snd_pcm_access_mask_set(mask, SND_PCM_ACCESS_MMAP_INTERLEAVED);
-+    snd_pcm_access_mask_set(mask, SND_PCM_ACCESS_MMAP_NONINTERLEAVED);
-+    snd_pcm_access_mask_set(mask, SND_PCM_ACCESS_MMAP_COMPLEX);
-+
-+#ifndef QT_NO_DEBUG
-+    qDebug() << "Trying to set mmapped write mode";
-+#endif
-+    if ( ( err = snd_pcm_hw_params_set_access_mask( alsa_pcm, hwparams, mask ) ) < 0 )
-+    {
-+        use_mmap = false;
-+#ifndef QT_NO_DEBUG
-+        qDebug() << "Setting mmapped write mode failed: " << snd_strerror( -err ) << "\n Trying normal write mode";
-+#endif
-+        if ( ( err = snd_pcm_hw_params_set_access( alsa_pcm, hwparams,
-+               SND_PCM_ACCESS_RW_INTERLEAVED ) ) < 0 )
-+        {
-+            LOGL( 1, "AlsaAudio::alsaSetup(): Cannot set normal write mode: " << snd_strerror( -err ) );
-+            return false;
-+        }
-+    }
-+    else
-+        use_mmap = true;
-+
-+    if ( ( err = snd_pcm_hw_params_set_format( alsa_pcm, hwparams, outputf->format ) ) < 0 )
-+    {
-+        //Try if one of these format work (one of them should work
-+        //on almost all soundcards)
-+
-+        snd_pcm_format_t formats[] = { SND_PCM_FORMAT_S16_LE,
-+                                       SND_PCM_FORMAT_S16_BE,
-+                                       SND_PCM_FORMAT_U8
-+                                     };
-+        uint i;
-+
-+        for ( i = 0; i < sizeof( formats ) / sizeof( formats[0] ); i++ )
-+        {
-+            if ( snd_pcm_hw_params_set_format( alsa_pcm, hwparams, formats[i] ) == 0 )
-+            {
-+                outputf->format = formats[i];
-+                break;
-+            }
-+        }
-+        if ( outputf->format != f->format )
-+        {
-+            outputf->xmms_format = (AFormat)format_from_alsa( outputf->format );
-+#ifndef QT_NO_DEBUG
-+            qDebug() << "Converting format from" << f->xmms_format << "to" << outputf->xmms_format;
-+#endif
-+            if ( outputf->xmms_format < 0 )
-+                return -1;
-+            alsa_convert_func = xmms_convert_get_func( outputf->xmms_format, f->xmms_format );
-+            if ( alsa_convert_func == NULL )
-+            {
-+                LOGL( 1, "Format translation needed, but not available.  Input: " << f->xmms_format << "; Output: " << outputf->xmms_format );
-+                return false;
-+            }
-+        }
-+        else
-+        {
-+            LOGL( 1, "AlsaAudio::alsaSetup(): Sample format not available for playback: " << snd_strerror( -err ) );
-+            return false;
-+        }
-+    }
-+
-+    snd_pcm_hw_params_set_channels_near( alsa_pcm, hwparams, &outputf->channels );
-+    if ( outputf->channels != f->channels )
-+    {
-+#ifndef QT_NO_DEBUG
-+        qDebug() << "Converting channels from" << f->channels << "to" << outputf->channels;
-+#endif
-+        alsa_stereo_convert_func =
-+                xmms_convert_get_channel_func( outputf->xmms_format,
-+                                               outputf->channels,
-+                                               f->channels );
-+        if ( alsa_stereo_convert_func == NULL )
-+        {
-+            LOGL( 1, "No stereo conversion available.  Format: " << outputf->xmms_format << "; Input Channels: " << f->channels << "; Output Channels: " << outputf->channels );
-+            return false;
-+        }
-+    }
-+
-+    snd_pcm_hw_params_set_rate_near( alsa_pcm, hwparams, &outputf->rate, 0 );
-+    if ( outputf->rate == 0 )
-+    {
-+        LOGL( 1, "AlsaAudio::alsaSetup: No usable samplerate available." );
-+        return false;
-+    }
-+    if ( outputf->rate != f->rate )
-+    {
-+        LOGL( 3, "Converting samplerate from " << f->rate << "  to " << outputf->rate );
-+        if ( outputf->channels < 1 || outputf->channels > 2 )
-+        {
-+            LOGL( 1, "Unsupported number of channels: " << outputf->channels << ". Resample function not available" );
-+            alsa_frequency_convert_func = NULL;
-+            return false;
-+        }
-+        alsa_frequency_convert_func =
-+                xmms_convert_get_frequency_func( outputf->xmms_format,
-+                                                 outputf->channels );
-+        if ( alsa_frequency_convert_func == NULL )
-+        {
-+            LOGL( 1, "Resample function not available.  Format " << outputf->xmms_format );
-+            return false;
-+        }
-+    }
-+
-+    outputf->sample_bits = snd_pcm_format_physical_width( outputf->format );
-+    outputf->bps = ( outputf->rate * outputf->sample_bits * outputf->channels ) >> 3;
-+
-+    if ( ( err = snd_pcm_hw_params_set_period_size_near( alsa_pcm, hwparams,
-+                                                         &periodSize, NULL ) ) < 0 )
-+    {
-+        LOGL( 1, "AlsaAudio::alsaSetup(): Set period size failed: " << snd_strerror( -err ) );
-+        return false;
-+    }
-+
-+    if ( ( err = snd_pcm_hw_params_set_periods_near( alsa_pcm, hwparams,
-+         &periodCount, 0 ) ) < 0 )
-+    {
-+        LOGL( 1, "AlsaAudio::alsaSetup(): Set period count failed: " << snd_strerror( -err ) );
-+        return false;
-+    }
-+
-+    if ( snd_pcm_hw_params( alsa_pcm, hwparams ) < 0 )
-+    {
-+#ifndef QT_NO_DEBUG
-+        snd_pcm_hw_params_dump( hwparams, logs );
-+#endif
-+        LOGL( 1, "AlsaAudio::alsaSetup(): Unable to install hw params" );
-+        return false;
-+    }
-+
-+    if ( ( err = snd_pcm_hw_params_get_buffer_size( hwparams, &alsa_buffer_size ) ) < 0 )
-+    {
-+        LOGL( 1, "AlsaAudio::alsaSetup(): snd_pcm_hw_params_get_buffer_size() failed: " << snd_strerror( -err ) );
-+        return false;
-+    }
-+
-+    if ( ( err = snd_pcm_hw_params_get_period_size( hwparams, &alsa_period_size, 0 ) ) < 0 )
-+    {
-+        LOGL( 1, "AlsaAudio::alsaSetup(): snd_pcm_hw_params_get_period_size() failed: " << snd_strerror( -err ) );
-+        return false;
-+    }
-+    snd_pcm_sw_params_alloca( &swparams );
-+    snd_pcm_sw_params_current( alsa_pcm, swparams );
-+
-+    if ( ( err = snd_pcm_sw_params_set_start_threshold( alsa_pcm,
-+         swparams, alsa_buffer_size - alsa_period_size ) < 0 ) )
-+        LOGL( 1, "AlsaAudio::alsaSetup(): setting start threshold failed: " << snd_strerror( -err ) );
-+    if ( snd_pcm_sw_params( alsa_pcm, swparams ) < 0 )
-+    {
-+        LOGL( 1, "AlsaAudio::alsaSetup(): Unable to install sw params" );
-+        return false;
-+    }
-+
-+#ifndef QT_NO_DEBUG
-+    snd_pcm_sw_params_dump( swparams, logs );
-+    snd_pcm_dump( alsa_pcm, logs );
-+#endif
-+    hw_period_size = snd_pcm_frames_to_bytes( alsa_pcm, alsa_period_size );
-+    if ( inputf->bps != outputf->bps )
-+    {
-+        int align = ( inputf->sample_bits * inputf->channels ) / 8;
-+        hw_period_size_in = ( (quint64)hw_period_size * inputf->bps +
-+                            outputf->bps/2 ) / outputf->bps;
-+        hw_period_size_in -= hw_period_size_in % align;
-+    }
-+    else
-+    {
-+        hw_period_size_in = hw_period_size;
-+    }
-+
-+#ifndef QT_NO_DEBUG
-+    qDebug() << "Device setup: period size:" << hw_period_size;
-+    qDebug() << "bits per sample:" <<  snd_pcm_format_physical_width( outputf->format ) <<
-+            "frame size:" <<  snd_pcm_frames_to_bytes( alsa_pcm, 1 ) <<
-+            "Bps:" << outputf->bps;
-+#endif
-+    return true;
-+}
-+
-+bool AlsaAudio::alsaOpen( QString device, AFormat format, unsigned int rate, unsigned int channels, unsigned int buffer_time, unsigned int period_time )
-+{
-+#ifndef QT_NO_DEBUG
-+    qDebug() << "Opening device";
-+#endif
-+    inputf = snd_format_from_xmms( format, rate, channels );
-+
-+    // We'll be using this in alsaWrite
-+    maxBufferSize = inputf->bps * MAX_BUFFER_SIZE;
-+    // And clear the buffer, just in case
-+    clearBuffer();
-+
-+    if ( alsaSetup( device, buffer_time, period_time, inputf ) == false )
-+    {
-+        alsaClose();
-+        return false;
-+    }
-+
-+    going = true;
-+    convertb = xmms_convert_buffers_new();
-+
-+    AlsaAudio* aaThread = new AlsaAudio();
-+#ifndef QT_NO_DEBUG
-+    qDebug() << "Starting thread";
-+#endif
-+    pthread_create( &audio_thread, NULL, &alsa_loop, (void*)aaThread );
-+
-+    return true;
-+}
-+
-+void AlsaAudio::clearBuffer( void )
-+{
-+    mutex.lock();
-+    audioData.clear();
-+    mutex.unlock();
-+}
-+
-+/******************************************************************************
-+    Play Interface
-+******************************************************************************/
-+
-+void AlsaAudio::alsaWrite( const QByteArray* inputData )
-+{
-+    if ( ( audioData.size() + inputData->size() ) < maxBufferSize )
-+    {
-+        mutex.lock();
-+#ifndef QT_NO_DEBUG
-+    qDebug( "max buffer size: %d; buffer data: %d; input data: %d", maxBufferSize, audioData.size(), inputData->size() );
-+#endif
-+        audioData.append( *inputData );
-+        mutex.unlock();
-+    }
-+    else
-+    {
-+        mutex.lock();
-+        int inSize = maxBufferSize - audioData.size();
-+#ifndef QT_NO_DEBUG
-+    qDebug( "max buffer size: %d; buffer data: %d; input data: %d; truncated to: %d", 
-+            maxBufferSize, audioData.size(), inputData->size(), inSize );
-+#endif
-+        audioData.append( inputData->left( inSize ) );
-+        mutex.unlock();
-+        LOGL( 1, "Max data buffer size reached.  Bytes dropped: " << inSize );
-+    }
-+}
-+
-+
-+int AlsaAudio::bufferSize( void )
-+{
-+    return audioData.size();
-+}
-+
-+void AlsaAudio::setVolume ( float vol )
-+{
-+    volume = vol;
-+}
-+
-+
-+void AlsaAudio::alsaClose( void )
-+{
-+    if ( !going )
-+        return;
-+
-+#ifndef QT_NO_DEBUG
-+    qDebug() << "Closing device";
-+#endif
-+
-+    going = false;
-+
-+    pthread_join( audio_thread, NULL );
-+
-+    xmms_convert_buffers_destroy( convertb );
-+    convertb = NULL;
-+    free( inputf );
-+    inputf = NULL;
-+    free( outputf );
-+    outputf = NULL;
-+
-+#ifndef QT_NO_DEBUG
-+    snd_output_close( logs );
-+    qDebug() << "Device closed";
-+#endif
-+}
-+
-+
-+/******************************************************************************
-+    Play Thread
-+******************************************************************************/
-+
-+void* AlsaAudio::alsa_loop( void* pthis )
-+{
-+  AlsaAudio* aaThread = (AlsaAudio*)pthis;
-+  aaThread->run();
-+  return NULL;
-+}
-+
-+
-+void AlsaAudio::run( void )
-+{
-+    int npfds = snd_pcm_poll_descriptors_count( alsa_pcm );
-+    struct pollfd *pfds;
-+    unsigned short *revents;
-+
-+    if ( npfds <= 0 )
-+        goto _error;
-+    pfds = (struct pollfd*)malloc( sizeof( *pfds ) * npfds );
-+    revents = (unsigned short*)malloc( sizeof( *revents ) * npfds );
-+    while ( going && alsa_pcm )
-+    {
-+        if ( audioData.size() > hw_period_size_in )
-+        {
-+            snd_pcm_poll_descriptors( alsa_pcm, pfds, npfds );
-+            if ( poll( pfds, npfds, 10 ) > 0 )
-+            {
-+                // need to check revents.  poll() with
-+                // dmix returns a postive value even
-+                // if no data is available
-+                int i;
-+                snd_pcm_poll_descriptors_revents( alsa_pcm, pfds,
-+                        npfds, revents );
-+                for ( i = 0; i < npfds; i++ )
-+                    if ( revents[i] & POLLOUT )
-+                {
-+                    pumpThreadData();
-+                    break;
-+                }
-+            }
-+        }
-+        else
-+        {
-+            struct timespec req;
-+            req.tv_sec = 0;
-+            req.tv_nsec = 10000000;
-+            nanosleep( &req, NULL );
-+        }
-+    }
-+    free( pfds );
-+    free( revents );
-+
-+ _error:
-+    alsa_close_pcm();
-+    mutex.lock();
-+    audioData.clear();
-+    mutex.unlock();
-+#ifndef QT_NO_DEBUG
-+    qDebug() << "Exiting thread";
-+#endif
-+    pthread_exit( NULL );
-+}
-+
-+
-+/* transfer audio data from thread buffer to h/w */
-+void AlsaAudio::pumpThreadData( void )
-+{
-+    int length, cnt, avail, datSize;
-+
-+    datSize = audioData.size();
-+    length = MIN( hw_period_size_in, datSize );
-+    avail = snd_pcm_frames_to_bytes( alsa_pcm, getAvailableFrames() );
-+    length = MIN( length, avail );
-+    while ( length > 0 )
-+    {
-+        cnt = MIN( length, datSize );
-+        convertData( audioData.left( cnt ).data(), cnt );
-+        mutex.lock();
-+        audioData.remove( 0, cnt );
-+        mutex.unlock();
-+        length -= cnt;
-+    }
-+}
-+
-+
-+/* update and get the available space on h/w buffer (in frames) */
-+snd_pcm_sframes_t AlsaAudio::getAvailableFrames( void )
-+{
-+    snd_pcm_sframes_t ret;
-+
-+    if ( alsa_pcm == NULL )
-+        return 0;
-+
-+    while ( ( ret = snd_pcm_avail_update( alsa_pcm ) ) < 0 )
-+    {
-+        ret = alsa_handle_error( ret );
-+        if ( ret < 0 )
-+        {
-+            LOGL( 1, "alsa_get_avail(): snd_pcm_avail_update() failed: " << snd_strerror( -ret ) );
-+            return 0;
-+        }
-+        return 0;
-+    }
-+    return ret;
-+}
-+
-+
-+/* transfer data to audio h/w; length is given in bytes
-+ *
-+ * data can be modified via rate conversion or
-+ * software volume before passed to audio h/w
-+ */
-+void AlsaAudio::convertData( void* data, int length )
-+{
-+    if ( alsa_convert_func != NULL )
-+        length = alsa_convert_func( convertb, &data, length );
-+    if ( alsa_stereo_convert_func != NULL )
-+        length = alsa_stereo_convert_func( convertb, &data, length );
-+    if ( alsa_frequency_convert_func != NULL )
-+    {
-+        length = alsa_frequency_convert_func( convertb, &data, length,
-+                                              inputf->rate,
-+                                              outputf->rate );
-+    }
-+
-+    adjustVolume( data, length, outputf->xmms_format );
-+
-+    writeToCard( (char*)data, length );
-+}
-+
-+
-+#define VOLUME_ADJUST( type, endian )                                       \
-+do {                                                                        \
-+    type *ptr = (type*)data;                                                \
-+    for ( i = 0; i < length; i += 2 )                                       \
-+    {                                                                       \
-+        *ptr = qTo##endian( (type)( qFrom##endian( *ptr ) * volume  ) );    \
-+        ptr++;                                                              \
-+    }                                                                       \
-+} while ( 0 )
-+
-+#define VOLUME_ADJUST8( type )           \
-+do {                                     \
-+    type *ptr = (type*)data;             \
-+    for ( i = 0; i < length; i++ )       \
-+    {                                    \
-+        *ptr = (type)( *ptr * volume );  \
-+        ptr++;                           \
-+    }                                    \
-+} while ( 0 )
-+
-+void AlsaAudio::adjustVolume( void* data, int length, AFormat fmt )
-+{
-+    int i;
-+    if ( volume == 1.0 )
-+        return;
-+
-+    switch ( fmt )
-+    {
-+        case FMT_S16_LE:
-+            VOLUME_ADJUST( qint16, LittleEndian );
-+            break;
-+        case FMT_U16_LE:
-+            VOLUME_ADJUST( quint16, LittleEndian );
-+            break;
-+        case FMT_S16_BE:
-+            VOLUME_ADJUST( qint16, BigEndian );
-+            break;
-+        case FMT_U16_BE:
-+            VOLUME_ADJUST( quint16, BigEndian );
-+            break;
-+        case FMT_S8:
-+            VOLUME_ADJUST8( qint8 );
-+            break;
-+        case FMT_U8:
-+            VOLUME_ADJUST8( quint8 );
-+            break;
-+        default:
-+            LOGL( 1, "AlsaAudio::adjustVolume(): unhandled format: " << fmt );
-+            break;
-+    }
-+}
-+
-+
-+/* transfer data to audio h/w via normal write */
-+void AlsaAudio::writeToCard( char *data, int length )
-+{
-+    snd_pcm_sframes_t written_frames;
-+
-+    while ( length > 0 )
-+    {
-+        int frames = snd_pcm_bytes_to_frames( alsa_pcm, length );
-+
-+        if ( use_mmap )
-+        {
-+            written_frames = snd_pcm_mmap_writei( alsa_pcm, data, frames );
-+        }
-+        else
-+            written_frames = snd_pcm_writei( alsa_pcm, data, frames );
-+
-+        if ( written_frames > 0 )
-+        {
-+            int written = snd_pcm_frames_to_bytes( alsa_pcm, written_frames );
-+            length -= written;
-+            data += written;
-+        }
-+        else
-+        {
-+            int err = alsa_handle_error( (int)written_frames );
-+            if ( err < 0 )
-+            {
-+                LOGL( 1, "AlsaAudio::writeToCard(): write error: " << snd_strerror( -err ) );
-+                break;
-+            }
-+        }
-+    }
-+}
-+
-+
-+/* handle generic errors */
-+int AlsaAudio::alsa_handle_error( int err )
-+{
-+    switch ( err )
-+    {
-+        case -EPIPE:
-+            return xrun_recover();
-+        case -ESTRPIPE:
-+            return suspend_recover();
-+    }
-+
-+    return err;
-+}
-+
-+
-+/* close PCM and release associated resources */
-+void AlsaAudio::alsa_close_pcm( void )
-+{
-+    if ( alsa_pcm )
-+    {
-+        int err;
-+        snd_pcm_drop( alsa_pcm );
-+        if ( ( err = snd_pcm_close( alsa_pcm ) ) < 0 )
-+            LOGL( 1, "alsa_pcm_close() failed: " << snd_strerror( -err ) );
-+        alsa_pcm = NULL;
-+    }
-+}
-+
-+
-+int AlsaAudio::format_from_alsa( snd_pcm_format_t fmt )
-+{
-+    uint i;
-+    for ( i = 0; i < sizeof( format_table ) / sizeof( format_table[0] ); i++ )
-+        if ( format_table[i].alsa == fmt )
-+            return format_table[i].xmms;
-+    LOGL( 1, "Unsupported format: " << snd_pcm_format_name( fmt ) );
-+    return -1;
-+}
-+
-+struct snd_format * AlsaAudio::snd_format_from_xmms( AFormat fmt, int rate, int channels )
-+{
-+    struct snd_format *f = (struct snd_format*)malloc( sizeof( struct snd_format ) );
-+    uint i;
-+
-+    f->xmms_format = fmt;
-+    f->format = SND_PCM_FORMAT_UNKNOWN;
-+
-+    for ( i = 0; i < sizeof( format_table ) / sizeof( format_table[0] ); i++ )
-+    {
-+        if ( format_table[i].xmms == fmt )
-+        {
-+            f->format = format_table[i].alsa;
-+            break;
-+        }
-+    }
-+
-+    /* Get rid of _NE */
-+    for ( i = 0; i < sizeof( format_table ) / sizeof( format_table[0] ); i++ )
-+    {
-+        if ( format_table[i].alsa == f->format )
-+        {
-+            f->xmms_format = format_table[i].xmms;
-+            break;
-+        }
-+    }
-+
-+    f->rate = rate;
-+    f->channels = channels;
-+    f->sample_bits = snd_pcm_format_physical_width( f->format );
-+    f->bps = ( rate * f->sample_bits * channels ) >> 3;
-+
-+    return f;
-+}
-+
-+
-+int AlsaAudio::xrun_recover( void )
-+{
-+
-+#ifndef QT_NO_DEBUG
-+    snd_pcm_status_t *alsa_status;
-+    snd_pcm_status_alloca( &alsa_status );
-+    if ( snd_pcm_status( alsa_pcm, alsa_status ) < 0 )
-+    {
-+        qDebug() << "AlsaAudio::xrun_recover(): snd_pcm_status() failed";
-+    }
-+    else
-+    {
-+        snd_pcm_status_dump( alsa_status, logs );
-+        qDebug() << "Status:\n" << logs;
-+    }
-+#endif
-+
-+    return snd_pcm_prepare( alsa_pcm );
-+}
-+
-+int AlsaAudio::suspend_recover( void )
-+{
-+    int err;
-+
-+    while ( ( err = snd_pcm_resume( alsa_pcm ) ) == -EAGAIN )
-+        /* wait until suspend flag is released */
-+        sleep( 1 );
-+    if ( err < 0 )
-+    {
-+        LOGL( 3, "alsa_handle_error(): snd_pcm_resume() failed." );
-+        return snd_pcm_prepare( alsa_pcm );
-+    }
-+    return err;
-+}
-diff -urN last.fm-1.1.3.orig/src/alsaplayback/alsaaudio.h last.fm-1.1.3/src/alsaplayback/alsaaudio.h
---- last.fm-1.1.3.orig/src/alsaplayback/alsaaudio.h	1969-12-31 16:00:00.000000000 -0800
-+++ last.fm-1.1.3/src/alsaplayback/alsaaudio.h	2007-03-18 20:57:28.000000000 -0700
-@@ -0,0 +1,121 @@
-+/***************************************************************************
-+ *   Copyright (C) 2007 by John Stamp                                      *
-+ *   jstamp@users.sourceforge.net                                          *
-+ *                                                                         *
-+ *   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.                                   *
-+ *                                                                         *
-+ *   This program is distributed in the hope that it will be useful,       *
-+ *   but WITHOUT ANY WARRANTY; without even the implied warranty of        *
-+ *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the         *
-+ *   GNU General Public License for more details.                          *
-+ *                                                                         *
-+ *   You should have received a copy of the GNU General Public License     *
-+ *   along with this program; if not, write to the                         *
-+ *   Free Software Foundation, Inc.,                                       *
-+ *   59 Temple Place - Suite 330, Boston, MA  02111-1307, USA.             *
-+ ***************************************************************************/
-+#ifndef ALSAAUDIO_H
-+#define ALSAAUDIO_H
-+
-+#include <alsa/asoundlib.h>
-+#include <QByteArray>
-+#include <QList>
-+#include <QString>
-+#include <QMutex>
-+#include "xconvert.h"
-+
-+struct AlsaDeviceInfo {
-+    QString name;
-+    QString device;
-+};
-+
-+struct snd_format {
-+    unsigned int rate;
-+    unsigned int channels;
-+    snd_pcm_format_t format;
-+    AFormat xmms_format;
-+    int sample_bits;
-+    int bps;
-+};
-+
-+static const struct {
-+    AFormat xmms;
-+    snd_pcm_format_t alsa;
-+} format_table[] =
-+                   { { FMT_S16_LE, SND_PCM_FORMAT_S16_LE },
-+                     { FMT_S16_BE, SND_PCM_FORMAT_S16_BE },
-+                     { FMT_S16_NE, SND_PCM_FORMAT_S16    },
-+                     { FMT_U16_LE, SND_PCM_FORMAT_U16_LE },
-+                     { FMT_U16_BE, SND_PCM_FORMAT_U16_BE },
-+                     { FMT_U16_NE, SND_PCM_FORMAT_U16    },
-+                     { FMT_U8,     SND_PCM_FORMAT_U8     },
-+                     { FMT_S8,     SND_PCM_FORMAT_S8     },
-+                   };
-+
-+class AlsaAudio {
-+public:
-+    AlsaAudio();
-+
-+    ~AlsaAudio();
-+
-+    int getCards();
-+    AlsaDeviceInfo getDeviceInfo( int device );
-+
-+    bool alsaOpen( QString device, AFormat format, unsigned int rate,
-+                   unsigned int channels, unsigned int buffer_time,
-+                   unsigned int period_time );
-+    void alsaWrite( const QByteArray* inputData );
-+
-+    void alsaClose( void );
-+
-+    void setVolume ( float vol );
-+    int bufferSize( void );
-+    void clearBuffer( void );
-+
-+private:
-+    QList<AlsaDeviceInfo> _devices;
-+
-+    int maxBufferSize;
-+
-+    // The following static variables are configured in either
-+    // alsaOpen or alsaSetup and used later in the audio thread
-+    static int hw_period_size_in;
-+    static snd_output_t *logs;
-+    static bool going;
-+    static snd_pcm_t *alsa_pcm;
-+    static snd_format* inputf;
-+    static snd_format* outputf;
-+    static float volume;
-+    static QByteArray audioData;
-+    static convert_func_t alsa_convert_func;
-+    static convert_channel_func_t alsa_stereo_convert_func;
-+    static convert_freq_func_t alsa_frequency_convert_func;
-+    static xmms_convert_buffers *convertb;
-+    static pthread_t audio_thread;
-+    static QMutex mutex;
-+    static bool use_mmap;
-+
-+    void getDevicesForCard( int card );
-+    bool alsaSetup( QString device, snd_pcm_uframes_t periodSize, uint periodCount, snd_format *f );
-+
-+    static void* alsa_loop( void* );
-+    void run( void );
-+    void pumpThreadData( void );
-+    void convertData( void* data, int length );
-+    void adjustVolume( void* data, int length, AFormat fmt );
-+    void writeToCard( char *data, int length );
-+
-+    snd_pcm_sframes_t getAvailableFrames( void );
-+    int alsa_handle_error( int err );
-+    int xrun_recover( void );
-+    int suspend_recover( void );
-+    int format_from_alsa( snd_pcm_format_t fmt );
-+    snd_format* snd_format_from_xmms( AFormat fmt, int rate, int channels );
-+
-+    void alsa_close_pcm( void );
-+};
-+
-+#endif
-diff -urN last.fm-1.1.3.orig/src/alsaplayback/alsaplayback.cpp last.fm-1.1.3/src/alsaplayback/alsaplayback.cpp
---- last.fm-1.1.3.orig/src/alsaplayback/alsaplayback.cpp	1969-12-31 16:00:00.000000000 -0800
-+++ last.fm-1.1.3/src/alsaplayback/alsaplayback.cpp	2007-03-18 20:57:28.000000000 -0700
-@@ -0,0 +1,205 @@
-+/***************************************************************************
-+ *   Copyright (C) 2005 - 2006 by                                          *
-+ *      Christian Muehlhaeuser, Last.fm Ltd <chris@last.fm>                *
-+ *      Erik Jaelevik, Last.fm Ltd <erik@last.fm>                          *
-+ *                                                                         *
-+ *   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.                                   *
-+ *                                                                         *
-+ *   This program is distributed in the hope that it will be useful,       *
-+ *   but WITHOUT ANY WARRANTY; without even the implied warranty of        *
-+ *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the         *
-+ *   GNU General Public License for more details.                          *
-+ *                                                                         *
-+ *   You should have received a copy of the GNU General Public License     *
-+ *   along with this program; if not, write to the                         *
-+ *   Free Software Foundation, Inc.,                                       *
-+ *   51 Franklin Steet, Fifth Floor, Boston, MA  02111-1307, USA.          *
-+ ***************************************************************************/
-+
-+#include <QtGui>
-+
-+#include "alsaplayback.h"
-+#include "containerutils.h"
-+#include "Loqqer.h"
-+
-+float AlsaPlayback::m_volume = 0.5;
-+
-+
-+AlsaPlayback::AlsaPlayback() :
-+    m_audio( 0 ),
-+    m_timer( new QTimer( this ) )
-+{
-+    gLogger.Init( savePath( "playback.log" ), false );
-+    gLogger.SetLevel( 4 );
-+
-+    LOGL( 3, "Initialising AlsaAudio Playback" );
-+    initAudio();
-+
-+    connect( m_timer, SIGNAL( timeout() ), this, SLOT( checkBuffer() ) );
-+}
-+
-+AlsaPlayback::~AlsaPlayback()
-+{
-+    delete m_audio;
-+}
-+
-+void
-+AlsaPlayback::setVolume( int volume )
-+{
-+    if ( m_audio )
-+        m_audio->setVolume( (float)volume / 100.0 );
-+}
-+
-+
-+QStringList
-+AlsaPlayback::soundSystems()
-+{
-+    QStringList l;
-+    l << "Alsa";
-+    return l;
-+}
-+
-+
-+QStringList
-+AlsaPlayback::devices()
-+{
-+#ifndef QT_NO_DEBUG
-+    qDebug() << "Querying audio devices";
-+#endif
-+    QStringList l;
-+
-+    if ( !m_audio )
-+        return l;
-+
-+    int cards = m_audio->getCards();
-+#ifndef QT_NO_DEBUG
-+    qDebug() << "Device nums:" << cards;
-+#endif
-+
-+    for ( int i = 0; i < cards; i++ )
-+    {
-+        AlsaDeviceInfo info;
-+        info = m_audio->getDeviceInfo( i );
-+#ifndef QT_NO_DEBUG
-+        qDebug() << "Device name:" << info.name ;
-+#endif
-+
-+        l << info.name;
-+    }
-+
-+    return l;
-+}
-+
-+
-+void
-+AlsaPlayback::startPlayback()
-+{
-+    int channels = 2;
-+    int sampleRate = 44100;
-+    int periodSize = 1024;  // According to mplayer, these two are good defaults.
-+    int periodCount = 16;   // They create a buffer size of 16384 frames.
-+    QString cardDevice;
-+
-+    if ( !m_audio )
-+    {
-+        LOGL( 1, "No AlsaAudio instance available." );
-+        goto _error;
-+    }
-+
-+    cardDevice = internalSoundCardID();
-+
-+    if ( !m_audio->alsaOpen( cardDevice, FMT_S16_LE, sampleRate, channels, periodSize, periodCount ) )
-+        goto _error;
-+
-+    if ( !m_timer->isActive() )
-+        m_timer->start( 15 );
-+
-+    return;
-+
-+_error:
-+    // We need to send a stop signal to m_iInput here, otherwise
-+    // it will keep running and filling up the buffers even though
-+    // there is no available device.
-+    emit stop();
-+    QMessageBox::critical( qApp->activeWindow(), tr("Audio Error"), tr("No soundcard available."));
-+    return;
-+}
-+
-+
-+void
-+AlsaPlayback::stopPlayback()
-+{
-+    if ( !m_audio )
-+    {
-+        return;
-+    }
-+
-+    m_timer->stop();
-+    m_audio->alsaClose();
-+}
-+
-+
-+void
-+AlsaPlayback::initAudio()
-+{
-+    m_audio = new AlsaAudio();
-+
-+    if ( m_audio )
-+    {
-+        LOGL( 1, "AlsaAudio successfully initialised." );
-+    }
-+    else
-+    {
-+        LOGL( 1, "Initialising AlsaAudio failed." );
-+    }
-+}
-+
-+
-+void
-+AlsaPlayback::dataAvailable( const QByteArray &buffer )
-+{
-+    m_audio->alsaWrite( &buffer );
-+}
-+
-+
-+void
-+AlsaPlayback::clearBuffers()
-+{
-+    m_audio->clearBuffer();
-+}
-+
-+
-+void
-+AlsaPlayback::checkBuffer()
-+{
-+    // NOTE: this is wavedata! 16kb of wavedata aprox. equals 1.6kb of mp3
-+    if ( m_audio->bufferSize() < 16384 * 12 )
-+        emit needData();
-+}
-+
-+
-+QString
-+AlsaPlayback::internalSoundCardID()
-+{
-+    if ( !m_audio )
-+        return "";
-+
-+    int cards = m_audio->getCards();
-+
-+    int cardnum = settingsService()->soundCard();
-+
-+    if ( cardnum < cards )
-+    {
-+        AlsaDeviceInfo info;
-+        info = m_audio->getDeviceInfo( cardnum );
-+        return info.device;
-+    }
-+    else
-+        return "default";
-+}
-+
-+
-+Q_EXPORT_PLUGIN2( playback, AlsaPlayback )
-diff -urN last.fm-1.1.3.orig/src/alsaplayback/alsaplayback.h last.fm-1.1.3/src/alsaplayback/alsaplayback.h
---- last.fm-1.1.3.orig/src/alsaplayback/alsaplayback.h	1969-12-31 16:00:00.000000000 -0800
-+++ last.fm-1.1.3/src/alsaplayback/alsaplayback.h	2007-03-18 20:57:28.000000000 -0700
-@@ -0,0 +1,71 @@
-+/***************************************************************************
-+ *   Copyright (C) 2005 - 2006 by                                          *
-+ *      Christian Muehlhaeuser, Last.fm Ltd <chris@last.fm>                *
-+ *      Erik Jaelevik, Last.fm Ltd <erik@last.fm>                          *
-+ *                                                                         *
-+ *   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.                                   *
-+ *                                                                         *
-+ *   This program is distributed in the hope that it will be useful,       *
-+ *   but WITHOUT ANY WARRANTY; without even the implied warranty of        *
-+ *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the         *
-+ *   GNU General Public License for more details.                          *
-+ *                                                                         *
-+ *   You should have received a copy of the GNU General Public License     *
-+ *   along with this program; if not, write to the                         *
-+ *   Free Software Foundation, Inc.,                                       *
-+ *   51 Franklin Steet, Fifth Floor, Boston, MA  02111-1307, USA.          *
-+ ***************************************************************************/
-+
-+#ifndef ALSAPLAYBACK_H
-+#define ALSAPLAYBACK_H
-+
-+#include <playbackinterface.h>
-+#include "alsaaudio.h"
-+
-+#include <QObject>
-+#include <QTimer>
-+
-+class AlsaPlayback : public PlaybackInterface
-+{
-+    Q_OBJECT
-+    Q_INTERFACES( PlaybackInterface )
-+
-+    public:
-+        AlsaPlayback();
-+        ~AlsaPlayback();
-+
-+        void initAudio();
-+        float volume() { return m_volume; }
-+
-+        QStringList soundSystems();
-+        QStringList devices();
-+
-+        static float m_volume;
-+
-+    signals:
-+        void needData();
-+        void stop();    // connected to m_iInput::stop()
-+
-+    public slots:
-+        void dataAvailable( const QByteArray &buffer );
-+        void clearBuffers();
-+
-+        void startPlayback();
-+        void stopPlayback();
-+
-+        void setVolume( int volume );
-+
-+    private:
-+        AlsaAudio *m_audio;
-+        QTimer *m_timer;
-+
-+        QString internalSoundCardID();
-+
-+    private slots:
-+        void checkBuffer();
-+};
-+
-+#endif
-diff -urN last.fm-1.1.3.orig/src/alsaplayback/alsaplayback.pro last.fm-1.1.3/src/alsaplayback/alsaplayback.pro
---- last.fm-1.1.3.orig/src/alsaplayback/alsaplayback.pro	1969-12-31 16:00:00.000000000 -0800
-+++ last.fm-1.1.3/src/alsaplayback/alsaplayback.pro	2007-03-18 20:57:28.000000000 -0700
-@@ -0,0 +1,19 @@
-+TEMPLATE      = lib
-+CONFIG       += plugin
-+INCLUDEPATH  += ../ ../libLastFMTools
-+LOGGERDIR     = ../build
-+LIBS         += -L../../bin -lLastFMTools -lasound
-+OBJECTS_DIR = ../build
-+MOC_DIR = ../build
-+UI_DIR = ../build
-+TARGET  = playback_alsaaudio
-+DESTDIR = ../../bin/services
-+
-+include(../../definitions.pro.inc)
-+
-+HEADERS       = alsaplayback.h alsaaudio.h xconvert.h
-+SOURCES       = alsaplayback.cpp alsaaudio.cpp xconvert.c
-+
-+LIBS += $$LOGGERDIR/Loqqer.o
-+
-+QT += gui xml network
-diff -urN last.fm-1.1.3.orig/src/alsaplayback/precompiled.h last.fm-1.1.3/src/alsaplayback/precompiled.h
---- last.fm-1.1.3.orig/src/alsaplayback/precompiled.h	1969-12-31 16:00:00.000000000 -0800
-+++ last.fm-1.1.3/src/alsaplayback/precompiled.h	2007-03-18 20:57:28.000000000 -0700
-@@ -0,0 +1,30 @@
-+/***************************************************************************

-+ *   Copyright (C) 2005 - 2006 by                                          *

-+ *      Christian Muehlhaeuser, Last.fm Ltd <chris@last.fm>                *

-+ *      Erik Jaelevik, Last.fm Ltd <erik@last.fm>                          *

-+ *                                                                         *

-+ *   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.                                   *

-+ *                                                                         *

-+ *   This program is distributed in the hope that it will be useful,       *

-+ *   but WITHOUT ANY WARRANTY; without even the implied warranty of        *

-+ *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the         *

-+ *   GNU General Public License for more details.                          *

-+ *                                                                         *

-+ *   You should have received a copy of the GNU General Public License     *

-+ *   along with this program; if not, write to the                         *

-+ *   Free Software Foundation, Inc.,                                       *

-+ *   51 Franklin Steet, Fifth Floor, Boston, MA  02111-1307, USA.          *

-+ ***************************************************************************/

-+

-+/* Add C includes here */

-+

-+#if defined __cplusplus

-+    /* Add C++ includes here */

-+    #include <QtCore>

-+    #include <QtGui>

-+    #include <QtNetwork>

-+    #include <QtXml>

-+#endif // __cplusplus

-diff -urN last.fm-1.1.3.orig/src/alsaplayback/xconvert.c last.fm-1.1.3/src/alsaplayback/xconvert.c
---- last.fm-1.1.3.orig/src/alsaplayback/xconvert.c	1969-12-31 16:00:00.000000000 -0800
-+++ last.fm-1.1.3/src/alsaplayback/xconvert.c	2007-03-18 20:57:28.000000000 -0700
-@@ -0,0 +1,771 @@
-+/*
-+ *  Copyright (C) 2001-2003  Haavard Kvaalen <havardk@xmms.org>
-+ *
-+ *  Licensed under GNU LGPL version 2.
-+ */
-+
-+#include <stdlib.h>
-+#include <stdint.h>
-+#include "xconvert.h"
-+
-+// These are adapted from defines in gtypes.h and glibconfig.h
-+#ifndef FALSE
-+#define FALSE    ( 0 )
-+#endif
-+
-+#ifndef TRUE
-+#define TRUE    ( !FALSE )
-+#endif
-+
-+# define GUINT16_SWAP_LE_BE( val )                          \
-+    ( ( uint16_t )                                          \
-+        (                                                   \
-+            ( uint16_t ) ( ( uint16_t ) ( val ) >> 8 ) |    \
-+            ( uint16_t ) ( ( uint16_t ) ( val ) << 8 )      \
-+        )                                                   \
-+    )
-+
-+# define GINT16_SWAP_LE_BE( val )    ( ( int16_t ) GUINT16_SWAP_LE_BE ( val ) )
-+
-+#ifdef WORDS_BIGENDIAN
-+
-+# define IS_BIG_ENDIAN TRUE
-+
-+# define GINT16_TO_BE( val )       ( ( int16_t ) ( val ) )
-+# define GINT16_FROM_BE( val )     ( ( int16_t ) ( val ) )
-+# define GUINT16_TO_BE( val )      ( ( uint16_t ) ( val ) )
-+# define GUINT16_FROM_BE( val )    ( ( uint16_t ) ( val ) )
-+
-+# define GUINT16_TO_LE( val )      ( GUINT16_SWAP_LE_BE ( val ) )
-+# define GUINT16_FROM_LE( val )    ( GUINT16_SWAP_LE_BE ( val ) )
-+# define GINT16_TO_LE( val )       ( ( int16_t ) GUINT16_SWAP_LE_BE ( val ) )
-+# define GINT16_FROM_LE( val )     ( ( int16_t ) GUINT16_SWAP_LE_BE ( val ) )
-+
-+#else
-+
-+# define IS_BIG_ENDIAN FALSE
-+
-+# define GINT16_TO_LE( val )       ( ( int16_t ) ( val ) )
-+# define GINT16_FROM_LE( val )     ( ( int16_t ) ( val ) )
-+# define GUINT16_TO_LE( val )      ( ( uint16_t ) ( val ) )
-+# define GUINT16_FROM_LE( val )    ( ( uint16_t ) ( val ) )
-+
-+# define GUINT16_TO_BE( val )      ( GUINT16_SWAP_LE_BE ( val ) )
-+# define GUINT16_FROM_BE( val )    ( GUINT16_SWAP_LE_BE ( val ) )
-+# define GINT16_TO_BE( val )       ( ( int16_t ) GUINT16_SWAP_LE_BE ( val ) )
-+# define GINT16_FROM_BE( val )     ( ( int16_t ) GUINT16_SWAP_LE_BE ( val ) )
-+
-+#endif
-+
-+
-+struct buffer {
-+    void *buffer;
-+    uint size;
-+};
-+
-+struct xmms_convert_buffers {
-+    struct buffer format_buffer, stereo_buffer, freq_buffer;
-+};
-+
-+struct xmms_convert_buffers* xmms_convert_buffers_new( void )
-+{
-+    return calloc( 1, sizeof( struct xmms_convert_buffers ) );
-+}
-+
-+static void* convert_get_buffer( struct buffer *buffer, size_t size )
-+{
-+    if ( size > 0 && size <= buffer->size )
-+        return buffer->buffer;
-+
-+    buffer->size = size;
-+    buffer->buffer = realloc( buffer->buffer, size );
-+    return buffer->buffer;
-+}
-+
-+void xmms_convert_buffers_free( struct xmms_convert_buffers* buf )
-+{
-+    convert_get_buffer( &buf->format_buffer, 0 );
-+    convert_get_buffer( &buf->stereo_buffer, 0 );
-+    convert_get_buffer( &buf->freq_buffer, 0 );
-+}
-+
-+void xmms_convert_buffers_destroy( struct xmms_convert_buffers* buf )
-+{
-+    if ( !buf )
-+        return;
-+    xmms_convert_buffers_free( buf );
-+    free( buf );
-+}
-+
-+static int convert_swap_endian( struct xmms_convert_buffers* buf, void **data, int length )
-+{
-+    uint16_t *ptr = *data;
-+    int i;
-+    for ( i = 0; i < length; i += 2, ptr++ )
-+        *ptr = GUINT16_SWAP_LE_BE( *ptr );
-+
-+    return i;
-+}
-+
-+static int convert_swap_sign_and_endian_to_native( struct xmms_convert_buffers* buf, void **data, int length )
-+{
-+    uint16_t *ptr = *data;
-+    int i;
-+    for ( i = 0; i < length; i += 2, ptr++ )
-+        *ptr = GUINT16_SWAP_LE_BE( *ptr ) ^ 1 << 15;
-+
-+    return i;
-+}
-+
-+static int convert_swap_sign_and_endian_to_alien( struct xmms_convert_buffers* buf, void **data, int length )
-+{
-+    uint16_t *ptr = *data;
-+    int i;
-+    for ( i = 0; i < length; i += 2, ptr++ )
-+        *ptr = GUINT16_SWAP_LE_BE( *ptr ^ 1 << 15 );
-+
-+    return i;
-+}
-+
-+static int convert_swap_sign16( struct xmms_convert_buffers* buf, void **data, int length )
-+{
-+    int16_t *ptr = *data;
-+    int i;
-+    for ( i = 0; i < length; i += 2, ptr++ )
-+        *ptr ^= 1 << 15;
-+
-+    return i;
-+}
-+
-+static int convert_swap_sign8( struct xmms_convert_buffers* buf, void **data, int length )
-+{
-+    int8_t *ptr = *data;
-+    int i;
-+    for ( i = 0; i < length; i++ )
-+        *ptr++ ^= 1 << 7;
-+
-+    return i;
-+}
-+
-+static int convert_to_8_native_endian( struct xmms_convert_buffers* buf, void **data, int length )
-+{
-+    int8_t *output = *data;
-+    int16_t *input = *data;
-+    int i;
-+    for ( i = 0; i < length / 2; i++ )
-+        *output++ = *input++ >> 8;
-+
-+    return i;
-+}
-+
-+static int convert_to_8_native_endian_swap_sign( struct xmms_convert_buffers* buf, void **data, int length )
-+{
-+    int8_t *output = *data;
-+    int16_t *input = *data;
-+    int i;
-+    for ( i = 0; i < length / 2; i++ )
-+        *output++ = ( *input++ >> 8 ) ^ ( 1 << 7 );
-+
-+    return i;
-+}
-+
-+
-+static int convert_to_8_alien_endian( struct xmms_convert_buffers* buf, void **data, int length )
-+{
-+    int8_t *output = *data;
-+    int16_t *input = *data;
-+    int i;
-+    for ( i = 0; i < length / 2; i++ )
-+        *output++ = *input++ & 0xff;
-+
-+    return i;
-+}
-+
-+static int convert_to_8_alien_endian_swap_sign( struct xmms_convert_buffers* buf, void **data, int length )
-+{
-+    int8_t *output = *data;
-+    int16_t *input = *data;
-+    int i;
-+    for ( i = 0; i < length / 2; i++ )
-+        *output++ = ( *input++ & 0xff ) ^ ( 1 << 7 );
-+
-+    return i;
-+}
-+
-+static int convert_to_16_native_endian( struct xmms_convert_buffers* buf, void **data, int length )
-+{
-+    uint8_t *input = *data;
-+    uint16_t *output;
-+    int i;
-+    *data = convert_get_buffer( &buf->format_buffer, length * 2 );
-+    output = *data;
-+    for ( i = 0; i < length; i++ )
-+        *output++ = *input++ << 8;
-+
-+    return i * 2;
-+}
-+
-+static int convert_to_16_native_endian_swap_sign( struct xmms_convert_buffers* buf, void **data, int length )
-+{
-+    uint8_t *input = *data;
-+    uint16_t *output;
-+    int i;
-+    *data = convert_get_buffer( &buf->format_buffer, length * 2 );
-+    output = *data;
-+    for ( i = 0; i < length; i++ )
-+        *output++ = ( *input++ << 8 ) ^ ( 1 << 15 );
-+
-+    return i * 2;
-+}
-+
-+
-+static int convert_to_16_alien_endian( struct xmms_convert_buffers* buf, void **data, int length )
-+{
-+    uint8_t *input = *data;
-+    uint16_t *output;
-+    int i;
-+    *data = convert_get_buffer( &buf->format_buffer, length * 2 );
-+    output = *data;
-+    for ( i = 0; i < length; i++ )
-+        *output++ = *input++;
-+
-+    return i * 2;
-+}
-+
-+static int convert_to_16_alien_endian_swap_sign( struct xmms_convert_buffers* buf, void **data, int length )
-+{
-+    uint8_t *input = *data;
-+    uint16_t *output;
-+    int i;
-+    *data = convert_get_buffer( &buf->format_buffer, length * 2 );
-+    output = *data;
-+    for ( i = 0; i < length; i++ )
-+        *output++ = *input++ ^ ( 1 << 7 );
-+
-+    return i * 2;
-+}
-+
-+static AFormat unnativize( AFormat fmt )
-+{
-+    if ( fmt == FMT_S16_NE )
-+    {
-+        if ( IS_BIG_ENDIAN )
-+            return FMT_S16_BE;
-+        else
-+            return FMT_S16_LE;
-+    }
-+    if ( fmt == FMT_U16_NE )
-+    {
-+        if ( IS_BIG_ENDIAN )
-+            return FMT_U16_BE;
-+        else
-+            return FMT_U16_LE;
-+    }
-+    return fmt;
-+}
-+
-+convert_func_t xmms_convert_get_func( AFormat output, AFormat input )
-+{
-+    output = unnativize( output );
-+    input = unnativize( input );
-+
-+    if ( output == input )
-+        return NULL;
-+
-+    if ( ( output == FMT_U16_BE && input == FMT_U16_LE ) ||
-+         ( output == FMT_U16_LE && input == FMT_U16_BE ) ||
-+         ( output == FMT_S16_BE && input == FMT_S16_LE ) ||
-+         ( output == FMT_S16_LE && input == FMT_S16_BE ) )
-+        return convert_swap_endian;
-+
-+    if ( ( output == FMT_U16_BE && input == FMT_S16_BE ) ||
-+         ( output == FMT_U16_LE && input == FMT_S16_LE ) ||
-+         ( output == FMT_S16_BE && input == FMT_U16_BE ) ||
-+         ( output == FMT_S16_LE && input == FMT_U16_LE ) )
-+        return convert_swap_sign16;
-+
-+    if ( ( IS_BIG_ENDIAN &&
-+         ( ( output == FMT_U16_BE && input == FMT_S16_LE ) ||
-+           ( output == FMT_S16_BE && input == FMT_U16_LE ) ) ) ||
-+        ( !IS_BIG_ENDIAN &&
-+         ( ( output == FMT_U16_LE && input == FMT_S16_BE ) ||
-+           ( output == FMT_S16_LE && input == FMT_U16_BE ) ) ) )
-+        return convert_swap_sign_and_endian_to_native;
-+
-+    if ( ( !IS_BIG_ENDIAN &&
-+         ( ( output == FMT_U16_BE && input == FMT_S16_LE ) ||
-+           ( output == FMT_S16_BE && input == FMT_U16_LE ) ) ) ||
-+        ( IS_BIG_ENDIAN &&
-+         ( ( output == FMT_U16_LE && input == FMT_S16_BE ) ||
-+           ( output == FMT_S16_LE && input == FMT_U16_BE ) ) ) )
-+        return convert_swap_sign_and_endian_to_alien;
-+
-+    if ( ( IS_BIG_ENDIAN &&
-+         ( ( output == FMT_U8 && input == FMT_U16_BE ) ||
-+           ( output == FMT_S8 && input == FMT_S16_BE ) ) ) ||
-+        ( !IS_BIG_ENDIAN &&
-+         ( ( output == FMT_U8 && input == FMT_U16_LE ) ||
-+           ( output == FMT_S8 && input == FMT_S16_LE ) ) ) )
-+        return convert_to_8_native_endian;
-+
-+    if ( ( IS_BIG_ENDIAN &&
-+         ( ( output == FMT_U8 && input == FMT_S16_BE ) ||
-+           ( output == FMT_S8 && input == FMT_U16_BE ) ) ) ||
-+        ( !IS_BIG_ENDIAN &&
-+         ( ( output == FMT_U8 && input == FMT_S16_LE ) ||
-+           ( output == FMT_S8 && input == FMT_U16_LE ) ) ) )
-+        return convert_to_8_native_endian_swap_sign;
-+
-+    if ( ( !IS_BIG_ENDIAN &&
-+         ( ( output == FMT_U8 && input == FMT_U16_BE ) ||
-+           ( output == FMT_S8 && input == FMT_S16_BE ) ) ) ||
-+        ( IS_BIG_ENDIAN &&
-+         ( ( output == FMT_U8 && input == FMT_U16_LE ) ||
-+           ( output == FMT_S8 && input == FMT_S16_LE ) ) ) )
-+        return convert_to_8_alien_endian;
-+
-+    if ( ( !IS_BIG_ENDIAN &&
-+         ( ( output == FMT_U8 && input == FMT_S16_BE ) ||
-+           ( output == FMT_S8 && input == FMT_U16_BE ) ) ) ||
-+        ( IS_BIG_ENDIAN &&
-+         ( ( output == FMT_U8 && input == FMT_S16_LE ) ||
-+           ( output == FMT_S8 && input == FMT_U16_LE ) ) ) )
-+        return convert_to_8_alien_endian_swap_sign;
-+
-+    if ( ( output == FMT_U8 && input == FMT_S8 ) ||
-+         ( output == FMT_S8 && input == FMT_U8 ) )
-+        return convert_swap_sign8;
-+
-+    if ( ( IS_BIG_ENDIAN &&
-+         ( ( output == FMT_U16_BE && input == FMT_U8 ) ||
-+           ( output == FMT_S16_BE && input == FMT_S8 ) ) ) ||
-+        ( !IS_BIG_ENDIAN &&
-+         ( ( output == FMT_U16_LE && input == FMT_U8 ) ||
-+           ( output == FMT_S16_LE && input == FMT_S8 ) ) ) )
-+        return convert_to_16_native_endian;
-+
-+    if ( ( IS_BIG_ENDIAN &&
-+         ( ( output == FMT_U16_BE && input == FMT_S8 ) ||
-+           ( output == FMT_S16_BE && input == FMT_U8 ) ) ) ||
-+        ( !IS_BIG_ENDIAN &&
-+         ( ( output == FMT_U16_LE && input == FMT_S8 ) ||
-+           ( output == FMT_S16_LE && input == FMT_U8 ) ) ) )
-+        return convert_to_16_native_endian_swap_sign;
-+
-+    if ( ( !IS_BIG_ENDIAN &&
-+         ( ( output == FMT_U16_BE && input == FMT_U8 ) ||
-+           ( output == FMT_S16_BE && input == FMT_S8 ) ) ) ||
-+        ( IS_BIG_ENDIAN &&
-+         ( ( output == FMT_U16_LE && input == FMT_U8 ) ||
-+           ( output == FMT_S16_LE && input == FMT_S8 ) ) ) )
-+        return convert_to_16_alien_endian;
-+
-+    if ( ( !IS_BIG_ENDIAN &&
-+         ( ( output == FMT_U16_BE && input == FMT_S8 ) ||
-+           ( output == FMT_S16_BE && input == FMT_U8 ) ) ) ||
-+        ( IS_BIG_ENDIAN &&
-+         ( ( output == FMT_U16_LE && input == FMT_S8 ) ||
-+           ( output == FMT_S16_LE && input == FMT_U8 ) ) ) )
-+        return convert_to_16_alien_endian_swap_sign;
-+
-+    //g_warning( "Translation needed, but not available.\n"
-+    //      "Input: %d; Output %d.", input, output );
-+    return NULL;
-+}
-+
-+static int convert_mono_to_stereo( struct xmms_convert_buffers* buf, void **data, int length, int b16 )
-+{
-+    int i;
-+    void *outbuf = convert_get_buffer( &buf->stereo_buffer, length * 2 );
-+
-+    if ( b16 )
-+    {
-+        uint16_t *output = outbuf, *input = *data;
-+        for ( i = 0; i < length / 2; i++ )
-+        {
-+            *output++ = *input;
-+            *output++ = *input;
-+            input++;
-+        }
-+    }
-+    else
-+    {
-+        uint8_t *output = outbuf, *input = *data;
-+        for ( i = 0; i < length; i++ )
-+        {
-+            *output++ = *input;
-+            *output++ = *input;
-+            input++;
-+        }
-+    }
-+    *data = outbuf;
-+
-+    return length * 2;
-+}
-+
-+static int convert_mono_to_stereo_8( struct xmms_convert_buffers* buf, void **data, int length )
-+{
-+    return convert_mono_to_stereo( buf, data, length, FALSE );
-+}
-+
-+static int convert_mono_to_stereo_16( struct xmms_convert_buffers* buf, void **data, int length )
-+{
-+    return convert_mono_to_stereo( buf, data, length, TRUE );
-+}
-+
-+static int convert_stereo_to_mono_u8( struct xmms_convert_buffers* buf, void **data, int length )
-+{
-+    uint8_t *output = *data, *input = *data;
-+    int i;
-+    for ( i = 0; i < length / 2; i++ )
-+    {
-+        uint16_t tmp;
-+        tmp = *input++;
-+        tmp += *input++;
-+        *output++ = tmp / 2;
-+    }
-+    return length / 2;
-+}
-+static int convert_stereo_to_mono_s8( struct xmms_convert_buffers* buf, void **data, int length )
-+{
-+    int8_t *output = *data, *input = *data;
-+    int i;
-+    for ( i = 0; i < length / 2; i++ )
-+    {
-+        int16_t tmp;
-+        tmp = *input++;
-+        tmp += *input++;
-+        *output++ = tmp / 2;
-+    }
-+    return length / 2;
-+}
-+static int convert_stereo_to_mono_u16le( struct xmms_convert_buffers* buf, void **data, int length )
-+{
-+    uint16_t *output = *data, *input = *data;
-+    int i;
-+    for ( i = 0; i < length / 4; i++ )
-+    {
-+        uint32_t tmp;
-+        uint16_t stmp;
-+        tmp = GUINT16_FROM_LE( *input );
-+        input++;
-+        tmp += GUINT16_FROM_LE( *input );
-+        input++;
-+        stmp = tmp / 2;
-+        *output++ = GUINT16_TO_LE( stmp );
-+    }
-+    return length / 2;
-+}
-+
-+static int convert_stereo_to_mono_u16be( struct xmms_convert_buffers* buf, void **data, int length )
-+{
-+    uint16_t *output = *data, *input = *data;
-+    int i;
-+    for ( i = 0; i < length / 4; i++ )
-+    {
-+        uint32_t tmp;
-+        uint16_t stmp;
-+        tmp = GUINT16_FROM_BE( *input );
-+        input++;
-+        tmp += GUINT16_FROM_BE( *input );
-+        input++;
-+        stmp = tmp / 2;
-+        *output++ = GUINT16_TO_BE( stmp );
-+    }
-+    return length / 2;
-+}
-+
-+static int convert_stereo_to_mono_s16le( struct xmms_convert_buffers* buf, void **data, int length )
-+{
-+    int16_t *output = *data, *input = *data;
-+    int i;
-+    for ( i = 0; i < length / 4; i++ )
-+    {
-+        int32_t tmp;
-+        int16_t stmp;
-+        tmp = GINT16_FROM_LE( *input );
-+        input++;
-+        tmp += GINT16_FROM_LE( *input );
-+        input++;
-+        stmp = tmp / 2;
-+        *output++ = GINT16_TO_LE( stmp );
-+    }
-+    return length / 2;
-+}
-+
-+static int convert_stereo_to_mono_s16be( struct xmms_convert_buffers* buf, void **data, int length )
-+{
-+    int16_t *output = *data, *input = *data;
-+    int i;
-+    for ( i = 0; i < length / 4; i++ )
-+    {
-+        int32_t tmp;
-+        int16_t stmp;
-+        tmp = GINT16_FROM_BE( *input );
-+        input++;
-+        tmp += GINT16_FROM_BE( *input );
-+        input++;
-+        stmp = tmp / 2;
-+        *output++ = GINT16_TO_BE( stmp );
-+    }
-+    return length / 2;
-+}
-+
-+convert_channel_func_t xmms_convert_get_channel_func( AFormat fmt, int output, int input )
-+{
-+    fmt = unnativize( fmt );
-+
-+    if ( output == input )
-+        return NULL;
-+
-+    if ( input == 1 && output == 2 )
-+        switch ( fmt )
-+        {
-+            case FMT_U8:
-+            case FMT_S8:
-+                return convert_mono_to_stereo_8;
-+            case FMT_U16_LE:
-+            case FMT_U16_BE:
-+            case FMT_S16_LE:
-+            case FMT_S16_BE:
-+                return convert_mono_to_stereo_16;
-+            default:
-+                //g_warning( "Unknown format: %d"
-+                //      "No conversion available.", fmt );
-+                return NULL;
-+        }
-+    if ( input == 2 && output == 1 )
-+        switch ( fmt )
-+        {
-+            case FMT_U8:
-+                return convert_stereo_to_mono_u8;
-+            case FMT_S8:
-+                return convert_stereo_to_mono_s8;
-+            case FMT_U16_LE:
-+                return convert_stereo_to_mono_u16le;
-+            case FMT_U16_BE:
-+                return convert_stereo_to_mono_u16be;
-+            case FMT_S16_LE:
-+                return convert_stereo_to_mono_s16le;
-+            case FMT_S16_BE:
-+                return convert_stereo_to_mono_s16be;
-+            default:
-+                //g_warning( "Unknown format: %d.  "
-+                //      "No conversion available.", fmt );
-+                return NULL;
-+        }
-+
-+    //g_warning( "Input has %d channels, soundcard uses %d channels\n"
-+    //      "No conversion is available", input, output );
-+    return NULL;
-+}
-+
-+
-+#define RESAMPLE_STEREO( sample_type, bswap )                   \
-+do {                                                            \
-+    const int shift = sizeof ( sample_type );                   \
-+        int i, in_samples, out_samples, x, delta;               \
-+    sample_type *inptr = *data, *outptr;                        \
-+    uint nlen = ( ( ( length >> shift ) * ofreq ) / ifreq );    \
-+    void *nbuf;                                                 \
-+    if ( nlen == 0 )                                            \
-+        break;                                                  \
-+    nlen <<= shift;                                             \
-+    if ( bswap )                                                \
-+        convert_swap_endian( NULL, data, length );              \
-+    nbuf = convert_get_buffer( &buf->freq_buffer, nlen );       \
-+    outptr = nbuf;                                              \
-+    in_samples = length >> shift;                               \
-+        out_samples = nlen >> shift;                            \
-+    delta = ( in_samples << 12 ) / out_samples;                 \
-+    for ( x = 0, i = 0; i < out_samples; i++ )                  \
-+    {                                                           \
-+        int x1, frac;                                           \
-+        x1 = ( x >> 12 ) << 12;                                 \
-+        frac = x - x1;                                          \
-+        *outptr++ =                                             \
-+            ( ( inptr[( x1 >> 12 ) << 1] *                      \
-+              ( ( 1<<12 ) - frac ) +                            \
-+              inptr[( ( x1 >> 12 ) + 1 ) << 1] *                \
-+              frac ) >> 12 );                                   \
-+        *outptr++ =                                             \
-+            ( ( inptr[( ( x1 >> 12 ) << 1 ) + 1] *              \
-+              ( ( 1<<12 ) - frac ) +                            \
-+              inptr[( ( ( x1 >> 12 ) + 1 ) << 1 ) + 1] *        \
-+              frac ) >> 12 );                                   \
-+        x += delta;                                             \
-+    }                                                           \
-+    if ( bswap )                                                \
-+        convert_swap_endian( NULL, &nbuf, nlen );               \
-+    *data = nbuf;                                               \
-+    return nlen;                                                \
-+} while ( 0 )
-+
-+
-+#define RESAMPLE_MONO( sample_type, bswap )                     \
-+do {                                                            \
-+    const int shift = sizeof ( sample_type ) - 1;               \
-+        int i, x, delta, in_samples, out_samples;               \
-+    sample_type *inptr = *data, *outptr;                        \
-+    uint nlen = ( ( ( length >> shift ) * ofreq ) / ifreq );    \
-+    void *nbuf;                                                 \
-+    if ( nlen == 0 )                                            \
-+        break;                                                  \
-+    nlen <<= shift;                                             \
-+    if ( bswap )                                                \
-+        convert_swap_endian( NULL, data, length );              \
-+    nbuf = convert_get_buffer( &buf->freq_buffer, nlen );       \
-+    outptr = nbuf;                                              \
-+    in_samples = length >> shift;                               \
-+        out_samples = nlen >> shift;                            \
-+    delta = ( ( length >> shift ) << 12 ) / out_samples;        \
-+    for ( x = 0, i = 0; i < out_samples; i++ )                  \
-+    {                                                           \
-+        int x1, frac;                                           \
-+        x1 = ( x >> 12 ) << 12;                                 \
-+        frac = x - x1;                                          \
-+        *outptr++ =                                             \
-+            ( ( inptr[x1 >> 12] * ( ( 1<<12 ) - frac ) +        \
-+              inptr[( x1 >> 12 ) + 1] * frac ) >> 12 );         \
-+        x += delta;                                             \
-+    }                                                           \
-+    if ( bswap )                                                \
-+        convert_swap_endian( NULL, &nbuf, nlen );               \
-+    *data = nbuf;                                               \
-+    return nlen;                                                \
-+} while ( 0 )
-+
-+static int convert_resample_stereo_s16ne( struct xmms_convert_buffers* buf, void **data, int length, int ifreq, int ofreq )
-+{
-+    RESAMPLE_STEREO( int16_t, FALSE );
-+    return 0;
-+}
-+
-+static int convert_resample_stereo_s16ae( struct xmms_convert_buffers* buf, void **data, int length, int ifreq, int ofreq )
-+{
-+    RESAMPLE_STEREO( int16_t, TRUE );
-+    return 0;
-+}
-+
-+static int convert_resample_stereo_u16ne( struct xmms_convert_buffers* buf, void **data, int length, int ifreq, int ofreq )
-+{
-+    RESAMPLE_STEREO( uint16_t, FALSE );
-+    return 0;
-+}
-+
-+static int convert_resample_stereo_u16ae( struct xmms_convert_buffers* buf, void **data, int length, int ifreq, int ofreq )
-+{
-+    RESAMPLE_STEREO( uint16_t, TRUE );
-+    return 0;
-+}
-+
-+static int convert_resample_mono_s16ne( struct xmms_convert_buffers* buf, void **data, int length, int ifreq, int ofreq )
-+{
-+    RESAMPLE_MONO( int16_t, FALSE );
-+    return 0;
-+}
-+
-+static int convert_resample_mono_s16ae( struct xmms_convert_buffers* buf, void **data, int length, int ifreq, int ofreq )
-+{
-+    RESAMPLE_MONO( int16_t, TRUE );
-+    return 0;
-+}
-+
-+static int convert_resample_mono_u16ne( struct xmms_convert_buffers* buf, void **data, int length, int ifreq, int ofreq )
-+{
-+    RESAMPLE_MONO( uint16_t, FALSE );
-+    return 0;
-+}
-+
-+static int convert_resample_mono_u16ae( struct xmms_convert_buffers* buf, void **data, int length, int ifreq, int ofreq )
-+{
-+    RESAMPLE_MONO( uint16_t, TRUE );
-+    return 0;
-+}
-+
-+static int convert_resample_stereo_u8( struct xmms_convert_buffers* buf, void **data, int length, int ifreq, int ofreq )
-+{
-+    RESAMPLE_STEREO( uint8_t, FALSE );
-+    return 0;
-+}
-+
-+static int convert_resample_mono_u8( struct xmms_convert_buffers* buf, void **data, int length, int ifreq, int ofreq )
-+{
-+    RESAMPLE_MONO( uint8_t, FALSE );
-+    return 0;
-+}
-+
-+static int convert_resample_stereo_s8( struct xmms_convert_buffers* buf, void **data, int length, int ifreq, int ofreq )
-+{
-+    RESAMPLE_STEREO( int8_t, FALSE );
-+    return 0;
-+}
-+
-+static int convert_resample_mono_s8( struct xmms_convert_buffers* buf, void **data, int length, int ifreq, int ofreq )
-+{
-+    RESAMPLE_MONO( int8_t, FALSE );
-+    return 0;
-+}
-+
-+
-+convert_freq_func_t xmms_convert_get_frequency_func( AFormat fmt, int channels )
-+{
-+    fmt = unnativize( fmt );
-+    //g_message( "fmt %d, channels: %d", fmt, channels );
-+
-+    if ( channels < 1 || channels > 2 )
-+    {
-+        //g_warning( "Unsupported number of channels: %d.  "
-+        //      "Resample function not available", channels );
-+        return NULL;
-+    }
-+    if ( (  IS_BIG_ENDIAN && fmt == FMT_U16_BE ) ||
-+         ( !IS_BIG_ENDIAN && fmt == FMT_U16_LE ) )
-+    {
-+        if ( channels == 1 )
-+            return convert_resample_mono_u16ne;
-+        else
-+            return convert_resample_stereo_u16ne;
-+    }
-+    if ( (  IS_BIG_ENDIAN && fmt == FMT_S16_BE ) ||
-+         ( !IS_BIG_ENDIAN && fmt == FMT_S16_LE ) )
-+    {
-+        if ( channels == 1 )
-+            return convert_resample_mono_s16ne;
-+        else
-+            return convert_resample_stereo_s16ne;
-+    }
-+    if ( ( !IS_BIG_ENDIAN && fmt == FMT_U16_BE ) ||
-+         (  IS_BIG_ENDIAN && fmt == FMT_U16_LE ) )
-+    {
-+        if ( channels == 1 )
-+            return convert_resample_mono_u16ae;
-+        else
-+            return convert_resample_stereo_u16ae;
-+    }
-+    if ( ( !IS_BIG_ENDIAN && fmt == FMT_S16_BE ) ||
-+         (  IS_BIG_ENDIAN && fmt == FMT_S16_LE ) )
-+    {
-+        if ( channels == 1 )
-+            return convert_resample_mono_s16ae;
-+        else
-+            return convert_resample_stereo_s16ae;
-+    }
-+    if ( fmt == FMT_U8 )
-+    {
-+        if ( channels == 1 )
-+            return convert_resample_mono_u8;
-+        else
-+            return convert_resample_stereo_u8;
-+    }
-+    if ( fmt == FMT_S8 )
-+    {
-+        if ( channels == 1 )
-+            return convert_resample_mono_s8;
-+        else
-+            return convert_resample_stereo_s8;
-+    }
-+    //g_warning( "Resample function not available"
-+    //      "Format %d.", fmt );
-+    return NULL;
-+}
-diff -urN last.fm-1.1.3.orig/src/alsaplayback/xconvert.h last.fm-1.1.3/src/alsaplayback/xconvert.h
---- last.fm-1.1.3.orig/src/alsaplayback/xconvert.h	1969-12-31 16:00:00.000000000 -0800
-+++ last.fm-1.1.3/src/alsaplayback/xconvert.h	2007-03-18 20:57:28.000000000 -0700
-@@ -0,0 +1,39 @@
-+/*
-+ *  Copyright (C) 2003  Haavard Kvaalen <havardk@xmms.org>
-+ *
-+ *  Licensed under GNU LGPL version 2.
-+ */
-+
-+#ifdef __cplusplus
-+extern "C" {
-+#endif
-+
-+typedef enum
-+{
-+    FMT_U8, FMT_S8, FMT_U16_LE, FMT_U16_BE, FMT_U16_NE, FMT_S16_LE, FMT_S16_BE, FMT_S16_NE
-+}
-+AFormat;
-+
-+struct xmms_convert_buffers;
-+
-+struct xmms_convert_buffers* xmms_convert_buffers_new(void);
-+/*
-+ * Free the data assosiated with the buffers, without destroying the
-+ * context.  The context can be reused.
-+ */
-+void xmms_convert_buffers_free(struct xmms_convert_buffers* buf);
-+void xmms_convert_buffers_destroy(struct xmms_convert_buffers* buf);
-+
-+
-+typedef int (*convert_func_t)(struct xmms_convert_buffers* buf, void **data, int length);
-+typedef int (*convert_channel_func_t)(struct xmms_convert_buffers* buf, void **data, int length);
-+typedef int (*convert_freq_func_t)(struct xmms_convert_buffers* buf, void **data, int length, int ifreq, int ofreq);
-+
-+
-+convert_func_t xmms_convert_get_func(AFormat output, AFormat input);
-+convert_channel_func_t xmms_convert_get_channel_func(AFormat fmt, int output, int input);
-+convert_freq_func_t xmms_convert_get_frequency_func(AFormat fmt, int channels);
-+
-+#ifdef __cplusplus
-+}
-+#endif
-diff -urN last.fm-1.1.3.orig/src/container.cpp last.fm-1.1.3/src/container.cpp
---- last.fm-1.1.3.orig/src/container.cpp	2007-03-18 20:57:23.000000000 -0700
-+++ last.fm-1.1.3/src/container.cpp	2007-03-18 20:57:28.000000000 -0700
-@@ -517,6 +517,7 @@
-     connect( m_iInput,      SIGNAL( newData( QByteArray ) ),  m_iTranscode, SLOT( dataAvailable( QByteArray ) ) );
-     connect( m_iTranscode,  SIGNAL( newData( QByteArray ) ),  m_iPlayback,  SLOT( dataAvailable( QByteArray ) ) );
-     connect( m_iPlayback,   SIGNAL( needData() ),             m_iInput,     SLOT( requestData() ) );
-+    connect( m_iPlayback,   SIGNAL( stop() ),                 m_iInput,     SLOT( stopStreaming() ) );
- 
-     // Auto updater
-     connect( &m_updater, SIGNAL( updateCheckDone( bool, bool, QString ) ),
-diff -urN last.fm-1.1.3.orig/src/libLastFMTools/containerutils.cpp last.fm-1.1.3/src/libLastFMTools/containerutils.cpp
---- last.fm-1.1.3.orig/src/libLastFMTools/containerutils.cpp	2007-03-18 20:57:23.000000000 -0700
-+++ last.fm-1.1.3/src/libLastFMTools/containerutils.cpp	2007-03-18 20:57:28.000000000 -0700
-@@ -288,7 +288,7 @@
- {
-     if ( g_playbackService == NULL )
-     {

--        QObject* obj = loadService( "playback_rtaudio" );

-+        QObject* obj = loadService( "playback_alsaaudio" );

-         g_playbackService = qobject_cast<PlaybackInterface*>( obj );

-     }
-     
diff --git a/media-sound/lastfmplayer/files/digest-lastfmplayer-1.1.3-r2 b/media-sound/lastfmplayer/files/digest-lastfmplayer-1.1.3-r2
deleted file mode 100644
index 2f5b667e9411..000000000000
--- a/media-sound/lastfmplayer/files/digest-lastfmplayer-1.1.3-r2
+++ /dev/null
@@ -1,3 +0,0 @@
-MD5 237531e1b30f75c4c0c07e7e39e9a250 last.fm-1.1.3.src.tar.bz2 856755
-RMD160 bfbddb23975dec3186251ebb3e1dcd4d0ad4ff9f last.fm-1.1.3.src.tar.bz2 856755
-SHA256 e05f9b1e177f6c4801295d33246e891690447c594b1138a292779dcef7d85cec last.fm-1.1.3.src.tar.bz2 856755
diff --git a/media-sound/lastfmplayer/files/digest-lastfmplayer-1.3.0.58 b/media-sound/lastfmplayer/files/digest-lastfmplayer-1.3.0.58
deleted file mode 100644
index 41d137ddbb11..000000000000
--- a/media-sound/lastfmplayer/files/digest-lastfmplayer-1.3.0.58
+++ /dev/null
@@ -1,3 +0,0 @@
-MD5 ba4c05af37006815a55eaa508169b6c2 last.fm-1.3.0.58.src.tar.bz2 1410799
-RMD160 0406e128c2f62af5bb2f90c135e34e91770dfcd7 last.fm-1.3.0.58.src.tar.bz2 1410799
-SHA256 e8d3fa8359a9d62548cafa65e1bcbf20e65dc3cc151d423bcabca1ae305ec4c9 last.fm-1.3.0.58.src.tar.bz2 1410799
diff --git a/media-sound/lastfmplayer/files/lastfmplayer-1.1.3-mp3transcode.patch b/media-sound/lastfmplayer/files/lastfmplayer-1.1.3-mp3transcode.patch
deleted file mode 100644
index ef6f2dd5a0a5..000000000000
--- a/media-sound/lastfmplayer/files/lastfmplayer-1.1.3-mp3transcode.patch
+++ /dev/null
@@ -1,24 +0,0 @@
---- src/mp3transcode/mp3transcode.cpp.old	2007-06-08 11:20:49.000000000 +0200
-+++ src/mp3transcode/mp3transcode.cpp	2007-06-08 11:58:05.000000000 +0200
-@@ -19,6 +19,7 @@
-  *   51 Franklin Steet, Fifth Floor, Boston, MA  02111-1307, USA.          *
-  ***************************************************************************/
- 
-+#include <qconfig.h> // for Q_BYTE_ORDER
- #include <QtGui>
- 
- #include "mp3transcode.h"
-@@ -64,8 +65,13 @@
-         {
-             for ( int i = 0; i < ( size / 2 ); i++ )
-             {
-+#if Q_BYTE_ORDER == Q_LITTLE_ENDIAN
-                 outputBuffer.append( tempBuffer[i * 2] );
-                 outputBuffer.append( tempBuffer[i * 2 + 1] );
-+#else // Q_BIG_ENDIAN
-+                outputBuffer.append( tempBuffer[i * 2 + 1] );
-+                outputBuffer.append( tempBuffer[i * 2] );
-+#endif // Q_BYTE_ORDER
-             }
- 
-             result = decodeMP3( &mpeg, NULL, 0, tempBuffer, sizeof( tempBuffer ), &size );
diff --git a/media-sound/lastfmplayer/files/lastfmplayer-1.1.3-pic.patch b/media-sound/lastfmplayer/files/lastfmplayer-1.1.3-pic.patch
deleted file mode 100644
index 5ad22ae98023..000000000000
--- a/media-sound/lastfmplayer/files/lastfmplayer-1.1.3-pic.patch
+++ /dev/null
@@ -1,11 +0,0 @@
---- src/Makefile	2007-02-11 19:39
-+++ src/Makefile	2007-02-11 19:40
-@@ -828,7 +828,7 @@
- 	$(CXX) -c $(CXXFLAGS) $(INCPATH) -o build/main.o main.cpp
- 
- build/Loqqer.o: Loqqer.cpp Loqqer.h
--	$(CXX) -c $(CXXFLAGS) $(INCPATH) -o build/Loqqer.o Loqqer.cpp
-+	$(CXX) -c $(CXXFLAGS) -fPIC $(INCPATH) -o build/Loqqer.o Loqqer.cpp
- 
- build/container.o: container.cpp container.h \
- 		playerlistener.h \
diff --git a/media-sound/lastfmplayer/files/lastfmplayer-amd64.patch b/media-sound/lastfmplayer/files/lastfmplayer-amd64.patch
deleted file mode 100644
index bff8283df96a..000000000000
--- a/media-sound/lastfmplayer/files/lastfmplayer-amd64.patch
+++ /dev/null
@@ -1,31 +0,0 @@
---- src/Makefile	2006-07-28 00:46:38.325333128 -0400
-+++ src/Makefile	2006-07-28 00:46:45.732207112 -0400
-@@ -748,7 +748,7 @@
- 	$(CXX) -c $(CXXFLAGS) $(INCPATH) -o build/main.o main.cpp
- 
- build/Loqqer.o: Loqqer.cpp Loqqer.h
--	$(CXX) -c $(CXXFLAGS) $(INCPATH) -o build/Loqqer.o Loqqer.cpp
-+	$(CXX) -c $(CXXFLAGS) -fPIC $(INCPATH) -o build/Loqqer.o Loqqer.cpp
- 
- build/container.o: container.cpp container.h \
- 		playerlistener.h \
---- src/wizardselectupdatespage.cpp	2006-07-28 14:32:19.099658720 -0400
-+++ src/wizardselectupdatespage.cpp	2006-07-28 15:29:14.400454080 -0400
-@@ -86,7 +86,7 @@
-         item->setCheckState(nCheck == 1 ? Qt::Checked : Qt::Unchecked);
- 
-         // Store its pointer along with string
--        item->setData(1, reinterpret_cast<unsigned int>(&current));
-+        item->setData(1, reinterpret_cast<unsigned long long>(&current));
-     }
- }
- 
-@@ -122,7 +122,7 @@
-         {
-             CComponentInfo* pComp =
-                 reinterpret_cast<CComponentInfo*>(
--                    ui.updatesList->item(i)->data(1).toUInt());
-+                    ui.updatesList->item(i)->data(1).toULongLong());
-             vecChecked.push_back(pComp);
-         }
-     }
diff --git a/media-sound/lastfmplayer/files/qt4.patch b/media-sound/lastfmplayer/files/qt4.patch
deleted file mode 100644
index 84e0316e3b8f..000000000000
--- a/media-sound/lastfmplayer/files/qt4.patch
+++ /dev/null
@@ -1,11 +0,0 @@
---- last.fm-1.1.3/configure.orig	2007-05-14 21:50:43.000000000 +0200
-+++ last.fm-1.1.3/configure	2007-05-14 21:50:49.000000000 +0200
-@@ -19,7 +19,7 @@
- # QMake version 2.01a
- # Using Qt version 4.2.2 in /opt/qt/4.2.2/lib
- 
--qmake -v | grep '^Using Qt version 4.2' --color=never &> /dev/null
-+qmake -v | grep '^Using Qt version 4' --color=never &> /dev/null
- 
- if [[ $? > 0 ]]; then
- 	      ################################################################################
diff --git a/media-sound/lastfmplayer/lastfmplayer-1.1.3-r2.ebuild b/media-sound/lastfmplayer/lastfmplayer-1.1.3-r2.ebuild
deleted file mode 100644
index e19471c42a76..000000000000
--- a/media-sound/lastfmplayer/lastfmplayer-1.1.3-r2.ebuild
+++ /dev/null
@@ -1,81 +0,0 @@
-# Copyright 1999-2007 Gentoo Foundation
-# Distributed under the terms of the GNU General Public License v2
-# $Header: /var/cvsroot/gentoo-x86/media-sound/lastfmplayer/lastfmplayer-1.1.3-r2.ebuild,v 1.3 2007/07/02 15:15:06 peper Exp $
-
-inherit eutils versionator
-
-DESCRIPTION="The player allows you to listen to last.fm radio streams"
-HOMEPAGE="http://www.last.fm/help/player"
-MY_P="${P/lastfmplayer/last.fm}"
-SRC_URI="http://static.last.fm/client/Linux/${MY_P}.src.tar.bz2"
-
-LICENSE="GPL-2"
-SLOT="0"
-KEYWORDS="~amd64 ~ppc ~x86"
-IUSE=""
-RESTRICT="mirror"
-S="${WORKDIR}/${MY_P}"
-
-DEPEND=">=x11-libs/qt-4.2
-	media-libs/alsa-lib"
-
-src_unpack() {
-	unpack ${A}
-	cd ${S}
-
-	epatch ${FILESDIR}/13_alsa-r1.diff
-	epatch ${FILESDIR}/qt4.patch
-	epatch ${FILESDIR}/${P}-mp3transcode.patch
-}
-
-src_compile() {
-	./configure
-	emake -j1 qmake_all || die "emake qmake_all failed"
-	epatch ${FILESDIR}/${P}-pic.patch
-
-	emake -j1 || die "emake failed"
-}
-
-src_install() {
-	# Docs
-	dodoc ChangeLog README
-
-	# The root at which the player, data, and cache
-	# are to be installed
-	local destination="/opt/lastfm"
-	cd bin
-
-	# Make ${destination} writable by audio group
-	diropts -m0775 -g audio
-	dodir ${destination}
-
-	# Install the player
-	cp -R * ${D}/${destination}
-
-	# Make a folder such that album art cache works
-	diropts -m0775 -g audio
-	dodir ${destination}/cache
-	keepdir ${destination}/cache
-
-	# Icon, menu, protcol
-	make_wrapper lastfm ./last.fm ${destination} ${destination}
-	newicon data/icons/as.png lastfm.png
-	make_desktop_entry lastfm "Last.fm Player" lastfm.png
-
-	# make sure no hanging instance of lastfm is running
-	sed -i -e 's:exec:if ! ps aux | grep "^$USER.*last.fm.app" | grep Sl >/dev/null 2>\&1; then killall -u $USER last.fm.app >/dev/null 2>\&1; fi\nexec:' ${D}/usr/bin/lastfm
-
-	insinto /usr/share/services
-	doins ${FILESDIR}/lastfm.protocol
-}
-
-pkg_postinst() {
-	elog "To use the Last.fm player with a mozilla based browser:"
-	elog " 1. Go to about:config in the browser"
-	elog " 2. Right-click on the page"
-	elog " 3. Select New and then String"
-	elog " 4. For the name: network.protocol-handler.app.lastfm"
-	elog " 5. For the value: /usr/bin/lastfm"
-	elog
-	elog "If you experiance awkward fonts or widgets, try running qtconfig."
-}
diff --git a/media-sound/lastfmplayer/lastfmplayer-1.3.0.58.ebuild b/media-sound/lastfmplayer/lastfmplayer-1.3.0.58.ebuild
deleted file mode 100644
index 04f1f4291904..000000000000
--- a/media-sound/lastfmplayer/lastfmplayer-1.3.0.58.ebuild
+++ /dev/null
@@ -1,76 +0,0 @@
-# Copyright 1999-2007 Gentoo Foundation
-# Distributed under the terms of the GNU General Public License v2
-# $Header: /var/cvsroot/gentoo-x86/media-sound/lastfmplayer/lastfmplayer-1.3.0.58.ebuild,v 1.4 2007/07/04 00:58:27 josejx Exp $
-
-inherit eutils versionator
-
-DESCRIPTION="The player allows you to listen to last.fm radio streams"
-HOMEPAGE="http://www.last.fm/help/player"
-MY_P="${P/lastfmplayer/last.fm}"
-SRC_URI="http://static.last.fm/client/Linux/${MY_P}.src.tar.bz2"
-
-LICENSE="GPL-2"
-SLOT="0"
-KEYWORDS="~x86 ~ppc ~amd64"
-IUSE=""
-RESTRICT="mirror"
-S="${WORKDIR}/${MY_P}"
-
-DEPEND=">=x11-libs/qt-4.2
-	media-libs/alsa-lib"
-
-src_unpack() {
-	unpack ${A}
-	cd ${S}
-	epatch ${FILESDIR}/qt4.patch
-}
-
-src_compile() {
-	./configure
-	emake qmake_all || die "emake qmake_all failed"
-	emake || die "emake failed"
-}
-
-src_install() {
-	# Docs
-	dodoc ChangeLog README
-
-	# The root at which the player, data, and cache
-	# are to be installed
-	local destination="/opt/lastfm"
-	cd bin
-
-	# Make ${destination} writable by audio group
-	diropts -m0775 -g audio
-	dodir ${destination}
-
-	# Install the player
-	cp -R * ${D}/${destination}
-
-	# Make a folder such that album art cache works
-	diropts -m0775 -g audio
-	dodir ${destination}/cache
-	keepdir ${destination}/cache
-
-	# Icon, menu, protcol
-	make_wrapper lastfm ./last.fm ${destination} ${destination}
-	newicon data/icons/as.png lastfm.png
-	make_desktop_entry lastfm "Last.fm Player" lastfm.png
-
-	# make sure no hanging instance of lastfm is running
-	sed -i -e 's:exec:if ! ps aux | grep "^$USER.*last.fm.app" | grep Sl >/dev/null 2>\&1; then killall -u $USER last.fm.app >/dev/null 2>\&1; fi\nexec:' ${D}/usr/bin/lastfm
-
-	insinto /usr/share/services
-	doins ${FILESDIR}/lastfm.protocol
-}
-
-pkg_postinst() {
-	elog "To use the Last.fm player with a mozilla based browser:"
-	elog " 1. Go to about:config in the browser"
-	elog " 2. Right-click on the page"
-	elog " 3. Select New and then String"
-	elog " 4. For the name: network.protocol-handler.app.lastfm"
-	elog " 5. For the value: /usr/bin/lastfm"
-	elog
-	elog "If you experiance awkward fonts or widgets, try running qtconfig."
-}
diff --git a/media-sound/lastfmplayer/lastfmplayer-1.3.1.0.ebuild b/media-sound/lastfmplayer/lastfmplayer-1.3.1.0.ebuild
index b4f35e2e50f2..ac1805928e31 100644
--- a/media-sound/lastfmplayer/lastfmplayer-1.3.1.0.ebuild
+++ b/media-sound/lastfmplayer/lastfmplayer-1.3.1.0.ebuild
@@ -1,12 +1,13 @@
 # Copyright 1999-2007 Gentoo Foundation
 # Distributed under the terms of the GNU General Public License v2
-# $Header: /var/cvsroot/gentoo-x86/media-sound/lastfmplayer/lastfmplayer-1.3.1.0.ebuild,v 1.1 2007/07/27 20:06:20 peper Exp $
+# $Header: /var/cvsroot/gentoo-x86/media-sound/lastfmplayer/lastfmplayer-1.3.1.0.ebuild,v 1.2 2007/08/29 18:30:00 genstef Exp $
 
-inherit eutils
+inherit eutils qt4
+
+MY_P="${P/lastfmplayer/last.fm}"
 
 DESCRIPTION="The player allows you to listen to last.fm radio streams"
 HOMEPAGE="http://www.last.fm/help/player"
-MY_P="${P/lastfmplayer/last.fm}"
 SRC_URI="http://static.last.fm/client/Linux/${MY_P}.src.tar.bz2"
 
 LICENSE="GPL-2"
@@ -14,20 +15,18 @@ SLOT="0"
 KEYWORDS="~x86 ~ppc ~amd64"
 IUSE=""
 RESTRICT="mirror"
-S="${WORKDIR}/${MY_P}"
 
-DEPEND=">=x11-libs/qt-4.2
+DEPEND="$(qt4_min_version 4.2)
 	media-libs/alsa-lib"
 
-src_unpack() {
-	unpack ${A}
-	cd "${S}"
-	epatch ${FILESDIR}/qt4.patch
-}
+S="${WORKDIR}/${MY_P}"
 
 src_compile() {
-	./configure
-	emake qmake_all || die "emake qmake_all failed"
+	eqmake4 LastFM.pro
+	for subdir in src src/libLastFmTools src/httpinput src/mp3transcode \
+			src/output/alsa-playback src/output/portAudio ; do
+		eqmake4 "${subdir}/${subdir##*/}.pro" -o ${subdir}/Makefile
+	done
 	emake || die "emake failed"
 }
 
@@ -45,7 +44,7 @@ src_install() {
 	dodir ${destination}
 
 	# Install the player
-	cp -R * ${D}/${destination}
+	cp -R * "${D}"/${destination}
 
 	# Make a folder such that album art cache works
 	diropts -m0775 -g audio
@@ -58,10 +57,12 @@ src_install() {
 	make_desktop_entry lastfm "Last.fm Player" lastfm.png
 
 	# make sure no hanging instance of lastfm is running
-	sed -i -e 's:exec:if ! ps aux | grep "^$USER.*last.fm.app" | grep Sl >/dev/null 2>\&1; then killall -u $USER last.fm.app >/dev/null 2>\&1; fi\nexec:' ${D}/usr/bin/lastfm
+	sed -i -e 's:exec:if ! ps aux | grep "^$USER.*last.fm.app" | grep Sl >/dev/null 2>\&1; \
+			then killall -u $USER last.fm.app >/dev/null 2>\&1; fi\nexec:' \
+			"${D}"/usr/bin/lastfm
 
 	insinto /usr/share/services
-	doins ${FILESDIR}/lastfm.protocol
+	doins "${FILESDIR}"/lastfm.protocol
 }
 
 pkg_postinst() {
@@ -72,5 +73,5 @@ pkg_postinst() {
 	elog " 4. For the name: network.protocol-handler.app.lastfm"
 	elog " 5. For the value: /usr/bin/lastfm"
 	elog
-	elog "If you experiance awkward fonts or widgets, try running qtconfig."
+	elog "If you experience awkward fonts or widgets, try running qtconfig."
 }