Copy the KDE 3.5 branch to branches/trinity for new KDE 3.5 features.

BUG:215923


git-svn-id: svn://anonsvn.kde.org/home/kde/branches/trinity/kdeaccessibility@1054174 283d02a7-25f6-0310-bc7c-ecb5cbfe19da

6 files changed, 2174 insertions, 0 deletions

diff --git a/kttsd/players/alsaplayer/Makefile.am b/kttsd/players/alsaplayer/Makefile.am
new file mode 100644
index 0000000..9653629
--- /dev/null
+++ b/kttsd/players/alsaplayer/Makefile.am
@@ -0,0 +1,30 @@
+# Include paths.
+INCLUDES = \
+	-I$(top_srcdir)/kttsd/libkttsd -I$(top_builddir)/kttsd/libkttsd \
+	$(all_includes)
+KDE_CXXFLAGS = -D_LARGEFILE64_SOURCE -D_GNU_SOURCE
+
+# Let automoc handle all of the metsource files (moc).
+METASOURCES = AUTO
+
+#########################################################################
+# LIBRARY SECTION
+#########################################################################
+# This is the library that gets installed.  It's name is used for all
+# of the other Makefile.am variables.
+kde_module_LTLIBRARIES = libkttsd_alsaplugin.la
+
+libkttsd_alsaplugin_la_SOURCES = \
+    alsaplugin.cpp \
+	alsaplayer.cpp
+libkttsd_alsaplugin_la_LDFLAGS = $(KDE_PLUGIN) $(all_libraries) $(LIBASOUND) -no-undefined
+libkttsd_alsaplugin_la_LIBADD = $(top_builddir)/kttsd/libkttsd/libkttsd.la
+
+# Header files that should not be installed.
+noinst_HEADERS = \
+	alsaplayer.h \
+	formats.h
+
+# This library is installed as a plugin.
+services_DATA = kttsd_alsaplugin.desktop
+servicesdir = $(kde_servicesdir)
diff --git a/kttsd/players/alsaplayer/alsaplayer.cpp b/kttsd/players/alsaplayer/alsaplayer.cpp
new file mode 100644
index 0000000..d3eabe8
--- /dev/null
+++ b/kttsd/players/alsaplayer/alsaplayer.cpp
@@ -0,0 +1,1729 @@
+/***************************************************** vim:set ts=4 sw=4 sts=4:
+  ALSA player.
+  -------------------
+  Copyright:
+  (C) 2005 by Gary Cramblitt <[email protected]>
+  Portions based on aplay.c in alsa-utils
+  Copyright (c) by Jaroslav Kysela <[email protected]>
+  Based on vplay program by Michael Beck
+  -------------------
+  Original author: Gary Cramblitt <[email protected]>
+
+  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 Street, Fifth Floor, Boston, MA  02110-1301, USA.
+ ******************************************************************************/
+
+// #include <sys/wait.h>
+// System includes.
+#include <config.h>
+#if TIME_WITH_SYS_TIME
+# include <sys/time.h>
+# include <time.h>
+#else
+# if HAVE_SYS_TIME_H
+#  include <sys/time.h>
+# else
+#  include <time.h>
+# endif
+#endif
+
+// Qt includes.
+#include <qdir.h>
+#include <qapplication.h>
+#include <qcstring.h>
+
+// KDE includes.
+#include <kdebug.h>
+#include <kconfig.h>
+#include <kstandarddirs.h>
+#include <kmessagebox.h>
+#include <klocale.h>
+
+// AlsaPlayer includes.
+#include "alsaplayer.h"
+
+#if !defined(__GNUC__) || __GNUC__ >= 3
+#define ERR(...) do {\
+    QString dbgStr;\
+    QString s = dbgStr.sprintf( "%s:%d: ERROR ", __FUNCTION__, __LINE__); \
+    s += dbgStr.sprintf( __VA_ARGS__); \
+    kdDebug() << timestamp() << "AlsaPlayer::" << s << endl; \
+} while (0)
+#else
+#define ERR(args...) do {\
+    QString dbgStr;\
+    QString s = dbgStr.sprintf( "%s:%d: ERROR ", __FUNCTION__, __LINE__); \
+    s += dbgStr.sprintf( ##args ); \
+    kdDebug() << timestamp() << "AlsaPlayer::" << s << endl; \
+} while (0)
+#endif
+
+#if !defined(__GNUC__) || __GNUC__ >= 3
+#define MSG(...) do {\
+    if (m_debugLevel >= 1) {\
+        QString dbgStr; \
+        QString s = dbgStr.sprintf( "%s:%d: ", __FUNCTION__, __LINE__); \
+        s += dbgStr.sprintf( __VA_ARGS__); \
+        kdDebug() << timestamp() << "AlsaPlayer::" << s << endl; \
+    }; \
+} while (0)
+#else
+#define MSG(args...) do {\
+    if (m_debugLevel >= 1) {\
+        QString dbgStr; \
+        QString s = dbgStr.sprintf( "%s:%d: ", __FUNCTION__, __LINE__); \
+        s += dbgStr.sprintf( ##args ); \
+        kdDebug() << timestamp() << "AlsaPlayer::" << s << endl; \
+    }; \
+} while (0)
+#endif
+
+#if !defined(__GNUC__) || __GNUC__ >= 3
+#define DBG(...) do {\
+    if (m_debugLevel >= 2) {\
+        QString dbgStr; \
+        QString s = dbgStr.sprintf( "%s:%d: ", __FUNCTION__, __LINE__); \
+        s += dbgStr.sprintf( __VA_ARGS__); \
+        kdDebug() << timestamp() << "AlsaPlayer::" << s << endl; \
+    }; \
+} while (0)
+#else
+#define DBG(args...) do {\
+    if (m_debugLevel >= 2) {\
+        QString dbgStr; \
+        QString s = dbgStr.sprintf( "%s:%d: ", __FUNCTION__, __LINE__); \
+        s += dbgStr.sprintf( ##args ); \
+        kdDebug() << timestamp() << "AlsaPlayer::" << s << endl; \
+    }; \
+} while (0)
+#endif
+
+QString AlsaPlayer::timestamp() const
+{
+    time_t t;
+    struct timeval tv;
+    char *tstr;
+    t = time(NULL);
+    tstr = strdup(ctime(&t));
+    tstr[strlen(tstr)-1] = 0;
+    gettimeofday(&tv,NULL);
+    QString ts;
+    ts.sprintf(" %s [%d] ",tstr, (int) tv.tv_usec);
+    free(tstr);
+    return ts;
+}
+
+////////////////////////////////////////////////////////////////////////////////
+// public methods
+////////////////////////////////////////////////////////////////////////////////
+
+AlsaPlayer::AlsaPlayer(QObject* parent, const char* name, const QStringList& args) : 
+    Player(parent, name, args),
+    m_currentVolume(1.0),
+    m_pcmName("default"),
+    m_defPeriodSize(128),
+    m_defPeriods(8),
+    m_debugLevel(1),
+    m_simulatedPause(false)
+{
+    init();
+}
+
+AlsaPlayer::~AlsaPlayer()
+{
+    if (running()) {
+        stop();
+        wait();
+    }
+}
+
+//void AlsaPlayer::play(const FileHandle &file)
+void AlsaPlayer::startPlay(const QString &file)
+{
+    if (running()) {
+        if (paused()) {
+            if (canPause)
+                snd_pcm_pause(handle, false);
+            else
+                m_simulatedPause = false;
+        }
+        return;
+    }
+    audiofile.setName(file);
+    audiofile.open(IO_ReadOnly);
+    fd = audiofile.handle();
+    // Start thread running.
+    start();
+}
+
+/*virtual*/ void AlsaPlayer::run()
+{
+    QString pName = m_pcmName.section(" ", 0, 0);
+    DBG("pName = %s", pName.ascii());
+    pcm_name = qstrdup(pName.ascii());
+    int err;
+    snd_pcm_info_t *info;
+
+    m_simulatedPause = false;
+
+    snd_pcm_info_alloca(&info);
+
+    err = snd_output_stdio_attach(&log, stderr, 0);
+    assert(err >= 0);
+
+    rhwdata.format = DEFAULT_FORMAT;
+    rhwdata.rate = DEFAULT_SPEED;
+    rhwdata.channels = 1;
+
+    err = snd_pcm_open(&handle, pcm_name, stream, open_mode);
+    if (err < 0) {
+        ERR("audio open error on pcm device %s: %s", pcm_name, snd_strerror(err));
+        return;
+    }
+
+    if ((err = snd_pcm_info(handle, info)) < 0) {
+        ERR("info error: %s", snd_strerror(err));
+        return;
+    }
+
+    chunk_size = 1024;
+    hwdata = rhwdata;
+
+    audioBuffer.resize(1024);
+    // audiobuf = (char *)malloc(1024);
+    audiobuf = audioBuffer.data();
+    if (audiobuf == NULL) {
+        ERR("not enough memory");
+        return;
+    }
+
+    if (mmap_flag) {
+        writei_func = snd_pcm_mmap_writei;
+        readi_func = snd_pcm_mmap_readi;
+        writen_func = snd_pcm_mmap_writen;
+        readn_func = snd_pcm_mmap_readn;
+    } else {
+        writei_func = snd_pcm_writei;
+        readi_func = snd_pcm_readi;
+        writen_func = snd_pcm_writen;
+        readn_func = snd_pcm_readn;
+    }
+
+    playback(fd);
+    cleanup();
+    return;
+}
+
+void AlsaPlayer::pause()
+{
+    if (running()) {
+        DBG("Pause requested");
+        m_mutex.lock();
+        if (handle) {
+            // Some hardware can pause; some can't.  canPause is set in set_params.
+            if (canPause) {
+                m_simulatedPause = false;
+                snd_pcm_pause(handle, true);
+                m_mutex.unlock();
+            } else {
+                // Set a flag and cause wait_for_poll to sleep.  When resumed, will get
+                // an underrun.
+                m_simulatedPause = true;
+                m_mutex.unlock();
+            }
+        }
+    }
+}
+
+void AlsaPlayer::stop()
+{
+    if (running()) {
+        DBG("STOP! Locking mutex");
+        m_mutex.lock();
+        m_simulatedPause = false;
+        if (handle) {
+            /* This constant is arbitrary */
+            char buf = 42;
+            DBG("Request for stop, device state is %s",
+                snd_pcm_state_name(snd_pcm_state(handle)));
+            write(alsa_stop_pipe[1], &buf, 1);
+        }
+        DBG("unlocking mutex");
+        m_mutex.unlock();
+        /* Wait for thread to exit */
+        DBG("waiting for thread to exit");
+        wait();
+        DBG("cleaning up");
+    }
+    cleanup();
+}
+
+/*
+ * Stop playback, cleanup and exit thread.
+ */
+void AlsaPlayer::stopAndExit()
+{
+    // if (handle) snd_pcm_drop(handle);
+    cleanup();
+    exit();
+}
+
+void AlsaPlayer::setVolume(float volume)
+{
+    m_currentVolume = volume;
+}
+
+float AlsaPlayer::volume() const
+{
+    return m_currentVolume;
+}
+
+/////////////////////////////////////////////////////////////////////////////////
+// player status functions
+/////////////////////////////////////////////////////////////////////////////////
+
+bool AlsaPlayer::playing() const
+{
+    bool result = false;
+    if (running()) {
+        m_mutex.lock();
+        if (handle) {
+            if (canPause) {
+                snd_pcm_status_t *status;
+                snd_pcm_status_alloca(&status);
+                int res;
+                if ((res = snd_pcm_status(handle, status)) < 0)
+                    ERR("status error: %s", snd_strerror(res));
+                else {
+                    result = (SND_PCM_STATE_RUNNING == snd_pcm_status_get_state(status))
+                        || (SND_PCM_STATE_DRAINING == snd_pcm_status_get_state(status));
+                    DBG("state = %s", snd_pcm_state_name(snd_pcm_status_get_state(status)));
+                }
+            } else
+                result = !m_simulatedPause;
+        }
+        m_mutex.unlock();
+    }
+    return result;
+}
+
+bool AlsaPlayer::paused() const
+{
+    bool result = false;
+    if (running()) {
+        m_mutex.lock();
+        if (handle) {
+            if (canPause) {
+                snd_pcm_status_t *status;
+                snd_pcm_status_alloca(&status);
+                int res;
+                if ((res = snd_pcm_status(handle, status)) < 0)
+                    ERR("status error: %s", snd_strerror(res));
+                else {
+                    result = (SND_PCM_STATE_PAUSED == snd_pcm_status_get_state(status));
+                    DBG("state = %s", snd_pcm_state_name(snd_pcm_status_get_state(status)));
+                }
+            } else
+                result = m_simulatedPause;
+        }
+        m_mutex.unlock();
+    }
+    return result;
+}
+
+int AlsaPlayer::totalTime() const
+{
+    int total = 0;
+    int rate = hwdata.rate;
+    int channels = hwdata.channels;
+    if (rate > 0 && channels > 0) {
+        total = int((double(pbrec_count) / rate) / channels);
+        // DBG("pbrec_count = %i rate =%i channels = %i", pbrec_count, rate, channels);
+        // DBG("totalTime = %i", total);
+    }
+    return total;
+}
+
+int AlsaPlayer::currentTime() const
+{
+    int current = 0;
+    int rate = hwdata.rate;
+    int channels = hwdata.channels;
+    if (rate > 0 && channels > 0) {
+        current = int((double(fdcount) / rate) / channels);
+        // DBG("fdcount = %i rate = %i channels = %i", fdcount, rate, channels);
+        // DBG("currentTime = %i", current);
+    }
+    return current;
+}
+
+int AlsaPlayer::position() const
+{
+    // TODO: Make this more accurate by adding frames that have been so-far
+    // played within the Alsa ring buffer.
+    return pbrec_count > 0 ? int(double(fdcount) * 1000 / pbrec_count + .5) : 0;
+}
+
+/////////////////////////////////////////////////////////////////////////////////
+// player seek functions
+/////////////////////////////////////////////////////////////////////////////////
+
+void AlsaPlayer::seek(int /*seekTime*/)
+{
+    // TODO:
+}
+
+void AlsaPlayer::seekPosition(int /*position*/)
+{
+    // TODO:
+}
+
+/*
+ * Returns a list of PCM devices.
+ * This function fills the specified list with ALSA hardware soundcards found on the system.
+ * It uses plughw:xx instead of hw:xx for specifiers, because hw:xx are not practical to
+ * use (e.g. they require a resampler/channel mixer in the application).
+ */
+QStringList AlsaPlayer::getPluginList( const QCString& /*classname*/ )
+{
+    int err = 0;
+    int card = -1, device = -1;
+    snd_ctl_t *handle;
+    snd_ctl_card_info_t *info;
+    snd_pcm_info_t *pcminfo;
+    snd_ctl_card_info_alloca(&info);
+    snd_pcm_info_alloca(&pcminfo);
+    QStringList result;
+
+    result.append("default");
+    for (;;) {
+        err = snd_card_next(&card);
+        if (err < 0 || card < 0) break;
+        if (card >= 0) {
+            char name[32];
+            sprintf(name, "hw:%i", card);
+            if ((err = snd_ctl_open(&handle, name, 0)) < 0) continue;
+            if ((err = snd_ctl_card_info(handle, info)) < 0) {
+                snd_ctl_close(handle);
+                continue;
+            }
+            for (int devCnt=0;;++devCnt) {
+                err = snd_ctl_pcm_next_device(handle, &device);
+                if (err < 0 || device < 0) break;
+
+                snd_pcm_info_set_device(pcminfo, device);
+                snd_pcm_info_set_subdevice(pcminfo, 0);
+                snd_pcm_info_set_stream(pcminfo, SND_PCM_STREAM_PLAYBACK);
+                if ((err = snd_ctl_pcm_info(handle, pcminfo)) < 0) continue;
+                QString infoName = " ";
+                infoName += snd_ctl_card_info_get_name(info);
+                infoName += " (";
+                infoName += snd_pcm_info_get_name(pcminfo);
+                infoName += ")";
+                if (0 == devCnt) {
+                    QString pcmName = QString("default:%1").arg(card);
+                    result.append(pcmName + infoName);
+                }
+                QString pcmName = QString("plughw:%1,%2").arg(card).arg(device);
+                result.append(pcmName + infoName);
+            }
+            snd_ctl_close(handle);
+        }
+    }
+    return result;
+}
+
+// QStringList AlsaPlayer::getPluginList( const QCString& /*classname*/ )
+// {
+//     QStringList assumed("default");
+//     snd_config_t *conf;
+//     int err = snd_config_update();
+//     if (err < 0) {
+//         ERR("snd_config_update: %s", snd_strerror(err));
+//         return assumed;
+//     }
+//     err = snd_config_search(snd_config, "pcm", &conf);
+//     if (err < 0) return QStringList();
+//     snd_config_iterator_t it = snd_config_iterator_first(conf);
+//     snd_config_iterator_t itEnd = snd_config_iterator_end(conf);
+//     const char* id;
+//     snd_config_t *entry;
+//     QStringList result;
+//     snd_ctl_card_info_t *info;
+//     snd_ctl_card_info_alloca(&info);
+//     snd_pcm_info_t *pcminfo;
+//     snd_pcm_info_alloca(&pcminfo);
+//     while (it != itEnd) {
+//         entry = snd_config_iterator_entry(it);
+//         err = snd_config_get_id(entry, &id);
+//         if (err >= 0) {
+//             if (QString(id) != "default")
+//             {
+//                 int card = -1;
+//                 while (snd_card_next(&card) >= 0 && card >= 0) {
+//                     char name[32];
+//                     sprintf(name, "%s:%d", id, card);
+//                     DBG("Checking %s", name);
+//                     snd_ctl_t *handle;
+//                     if ((err = snd_ctl_open(&handle, name, SND_CTL_NONBLOCK)) >= 0) {
+//                         if ((err = snd_ctl_card_info(handle, info)) >= 0) {
+//                             int dev = -1;
+//                             snd_pcm_stream_t stream = SND_PCM_STREAM_PLAYBACK;
+//                             while (snd_ctl_pcm_next_device(handle, &dev) >= 0 && dev >= 0) {
+//                                 snd_pcm_info_set_device(pcminfo, dev);
+//                                 snd_pcm_info_set_subdevice(pcminfo, 0);
+//                                 snd_pcm_info_set_stream(pcminfo, stream);
+//                                 if ((err = snd_ctl_pcm_info(handle, pcminfo)) >= 0) {
+//                                     QString pluginName = name;
+//                                     pluginName += ",";
+//                                     pluginName += QString::number(dev);
+//                                     pluginName += " ";
+//                                     pluginName += snd_ctl_card_info_get_name(info);
+//                                     pluginName += ",";
+//                                     pluginName += snd_pcm_info_get_name(pcminfo);
+//                                     result.append(pluginName);
+//                                     // DBG(pluginName);
+//                                 }
+//                             }
+//                         }
+//                         snd_ctl_close(handle);
+//                     }
+//                 }
+//                 if (card == -1) result.append(id);
+//             } else result.append(id);
+//         }
+//         it = snd_config_iterator_next(it);
+//     }
+//     snd_config_update_free_global();
+//     return result;
+// }
+
+void AlsaPlayer::setSinkName(const QString& sinkName) { m_pcmName = sinkName; }
+
+/////////////////////////////////////////////////////////////////////////////////
+// private
+/////////////////////////////////////////////////////////////////////////////////
+
+void AlsaPlayer::init()
+{
+    pcm_name = 0;
+    handle = 0;
+    canPause = false;
+    timelimit = 0;
+    file_type = FORMAT_DEFAULT;
+    sleep_min = 0;
+    // open_mode = 0;
+    open_mode = SND_PCM_NONBLOCK;
+    stream = SND_PCM_STREAM_PLAYBACK;
+    mmap_flag = 0;
+    interleaved = 1;
+    audiobuf = NULL;
+    chunk_size = 0;
+    period_time = 0;
+    buffer_time = 0;
+    avail_min = -1;
+    start_delay = 0;
+    stop_delay = 0;
+    buffer_pos = 0;
+    log = 0;
+    fd = -1;
+    pbrec_count = LLONG_MAX;
+    alsa_stop_pipe[0] = 0;
+    alsa_stop_pipe[1] = 0;
+    alsa_poll_fds = 0;
+    m_simulatedPause = false;
+}
+
+void AlsaPlayer::cleanup()
+{
+    DBG("cleaning up");
+    m_mutex.lock();
+    if (pcm_name) free(pcm_name);
+    if (fd >= 0) audiofile.close();
+    if (handle) {
+        snd_pcm_drop(handle);
+        snd_pcm_close(handle);
+    }
+    if (alsa_stop_pipe[0]) close(alsa_stop_pipe[0]);
+    if (alsa_stop_pipe[1]) close(alsa_stop_pipe[1]);
+    if (audiobuf) audioBuffer.resize(0);
+    if (alsa_poll_fds) alsa_poll_fds_barray.resize(0);
+    if (log) snd_output_close(log);
+    snd_config_update_free_global();
+    init();
+    m_mutex.unlock();
+}
+
+/*
+ * Safe read (for pipes)
+ */
+
+ssize_t AlsaPlayer::safe_read(int fd, void *buf, size_t count)
+{
+    ssize_t result = 0;
+    ssize_t res;
+
+    while (count > 0) {
+        if ((res = read(fd, buf, count)) == 0)
+            break;
+        if (res < 0)
+            return result > 0 ? result : res;
+        count -= res;
+        result += res;
+        buf = (char *)buf + res;
+    }
+    return result;
+}
+
+/*
+ * Test, if it is a .VOC file and return >=0 if ok (this is the length of rest)
+ *                                       < 0 if not 
+ */
+int AlsaPlayer::test_vocfile(void *buffer)
+{
+    VocHeader *vp = (VocHeader*)buffer;
+
+    if (!memcmp(vp->magic, VOC_MAGIC_STRING, 20)) {
+        vocminor = LE_SHORT(vp->version) & 0xFF;
+        vocmajor = LE_SHORT(vp->version) / 256;
+        if (LE_SHORT(vp->version) != (0x1233 - LE_SHORT(vp->coded_ver)))
+            return -2;    /* coded version mismatch */
+        return LE_SHORT(vp->headerlen) - sizeof(VocHeader);    /* 0 mostly */
+    }
+    return -1;        /* magic string fail */
+}
+
+/*
+ * helper for test_wavefile
+ */
+
+size_t AlsaPlayer::test_wavefile_read(int fd, char *buffer, size_t *size, size_t reqsize, int line)
+{
+    if (*size >= reqsize)
+        return *size;
+    if ((size_t)safe_read(fd, buffer + *size, reqsize - *size) != reqsize - *size) {
+        ERR("read error (called from line %i)", line);
+        stopAndExit();
+    }
+    return *size = reqsize;
+}
+
+#define check_wavefile_space(buffer, len, blimit) \
+    if (len > blimit) { \
+        blimit = len; \
+        if ((buffer = (char*)realloc(buffer, blimit)) == NULL) { \
+            ERR("not enough memory"); \
+            stopAndExit(); \
+        } \
+    }
+
+/*
+ * test, if it's a .WAV file, > 0 if ok (and set the speed, stereo etc.)
+ *                            == 0 if not
+ * Value returned is bytes to be discarded.
+ */
+ssize_t AlsaPlayer::test_wavefile(int fd, char *_buffer, size_t size)
+{
+    WaveHeader *h = (WaveHeader *)_buffer;
+    char *buffer = NULL;
+    size_t blimit = 0;
+    WaveFmtBody *f;
+    WaveChunkHeader *c;
+    u_int type;
+    u_int len;
+
+    if (size < sizeof(WaveHeader))
+        return -1;
+    if (h->magic != WAV_RIFF || h->type != WAV_WAVE)
+        return -1;
+    if (size > sizeof(WaveHeader)) {
+        check_wavefile_space(buffer, size - sizeof(WaveHeader), blimit);
+        memcpy(buffer, _buffer + sizeof(WaveHeader), size - sizeof(WaveHeader));
+    }
+    size -= sizeof(WaveHeader);
+    while (1) {
+        check_wavefile_space(buffer, sizeof(WaveChunkHeader), blimit);
+        test_wavefile_read(fd, buffer, &size, sizeof(WaveChunkHeader), __LINE__);
+        c = (WaveChunkHeader*)buffer;
+        type = c->type;
+        len = LE_INT(c->length);
+        len += len % 2;
+        if (size > sizeof(WaveChunkHeader))
+            memmove(buffer, buffer + sizeof(WaveChunkHeader), size - sizeof(WaveChunkHeader));
+        size -= sizeof(WaveChunkHeader);
+        if (type == WAV_FMT)
+            break;
+        check_wavefile_space(buffer, len, blimit);
+        test_wavefile_read(fd, buffer, &size, len, __LINE__);
+        if (size > len)
+            memmove(buffer, buffer + len, size - len);
+        size -= len;
+    }
+
+    if (len < sizeof(WaveFmtBody)) {
+        ERR("unknown length of 'fmt ' chunk (read %u, should be %u at least)", len, (u_int)sizeof(WaveFmtBody));
+        stopAndExit();
+    }
+    check_wavefile_space(buffer, len, blimit);
+    test_wavefile_read(fd, buffer, &size, len, __LINE__);
+    f = (WaveFmtBody*) buffer;
+    if (LE_SHORT(f->format) != WAV_PCM_CODE) {
+        ERR("can't play not PCM-coded WAVE-files");
+        stopAndExit();
+    }
+    if (LE_SHORT(f->modus) < 1) {
+        ERR("can't play WAVE-files with %d tracks", LE_SHORT(f->modus));
+        stopAndExit();
+    }
+    hwdata.channels = LE_SHORT(f->modus);
+    switch (LE_SHORT(f->bit_p_spl)) {
+    case 8:
+        if (hwdata.format != DEFAULT_FORMAT &&
+            hwdata.format != SND_PCM_FORMAT_U8)
+            MSG("Warning: format is changed to U8");
+        hwdata.format = SND_PCM_FORMAT_U8;
+        break;
+    case 16:
+        if (hwdata.format != DEFAULT_FORMAT &&
+            hwdata.format != SND_PCM_FORMAT_S16_LE)
+            MSG("Warning: format is changed to S16_LE");
+        hwdata.format = SND_PCM_FORMAT_S16_LE;
+        break;
+    case 24:
+        switch (LE_SHORT(f->byte_p_spl) / hwdata.channels) {
+        case 3:
+            if (hwdata.format != DEFAULT_FORMAT &&
+                hwdata.format != SND_PCM_FORMAT_S24_3LE)
+                MSG("Warning: format is changed to S24_3LE");
+            hwdata.format = SND_PCM_FORMAT_S24_3LE;
+            break;
+        case 4:
+            if (hwdata.format != DEFAULT_FORMAT &&
+                hwdata.format != SND_PCM_FORMAT_S24_LE)
+                MSG("Warning: format is changed to S24_LE");
+            hwdata.format = SND_PCM_FORMAT_S24_LE;
+            break;
+        default:
+            ERR("can't play WAVE-files with sample %d bits in %d bytes wide (%d channels)", LE_SHORT(f->bit_p_spl), LE_SHORT(f->byte_p_spl), hwdata.channels);
+            stopAndExit();
+        }
+        break;
+    case 32:
+        hwdata.format = SND_PCM_FORMAT_S32_LE;
+        break;
+    default:
+        ERR("can't play WAVE-files with sample %d bits wide", LE_SHORT(f->bit_p_spl));
+        stopAndExit();
+    }
+    hwdata.rate = LE_INT(f->sample_fq);
+
+    if (size > len)
+        memmove(buffer, buffer + len, size - len);
+    size -= len;
+
+    while (1) {
+        u_int type, len;
+
+        check_wavefile_space(buffer, sizeof(WaveChunkHeader), blimit);
+        test_wavefile_read(fd, buffer, &size, sizeof(WaveChunkHeader), __LINE__);
+        c = (WaveChunkHeader*)buffer;
+        type = c->type;
+        len = LE_INT(c->length);
+        if (size > sizeof(WaveChunkHeader))
+            memmove(buffer, buffer + sizeof(WaveChunkHeader), size - sizeof(WaveChunkHeader));
+        size -= sizeof(WaveChunkHeader);
+        if (type == WAV_DATA) {
+            if (len < pbrec_count && len < 0x7ffffffe)
+                pbrec_count = len;
+            if (size > 0)
+                memcpy(_buffer, buffer, size);
+            free(buffer);
+            return size;
+        }
+        len += len % 2;
+        check_wavefile_space(buffer, len, blimit);
+        test_wavefile_read(fd, buffer, &size, len, __LINE__);
+        if (size > len)
+            memmove(buffer, buffer + len, size - len);
+        size -= len;
+    }
+
+    /* shouldn't be reached */
+    return -1;
+}
+
+/*
+ * Test for AU file.
+ */
+
+int AlsaPlayer::test_au(int fd, char *buffer)
+{
+    AuHeader *ap = (AuHeader*)buffer;
+
+    if (ap->magic != AU_MAGIC)
+        return -1;
+    if (BE_INT(ap->hdr_size) > 128 || BE_INT(ap->hdr_size) < 24)
+        return -1;
+    pbrec_count = BE_INT(ap->data_size);
+    switch (BE_INT(ap->encoding)) {
+    case AU_FMT_ULAW:
+        if (hwdata.format != DEFAULT_FORMAT &&
+            hwdata.format != SND_PCM_FORMAT_MU_LAW)
+            MSG("Warning: format is changed to MU_LAW");
+        hwdata.format = SND_PCM_FORMAT_MU_LAW;
+        break;
+    case AU_FMT_LIN8:
+        if (hwdata.format != DEFAULT_FORMAT &&
+            hwdata.format != SND_PCM_FORMAT_U8)
+            MSG("Warning: format is changed to U8");
+        hwdata.format = SND_PCM_FORMAT_U8;
+        break;
+    case AU_FMT_LIN16:
+        if (hwdata.format != DEFAULT_FORMAT &&
+            hwdata.format != SND_PCM_FORMAT_S16_BE)
+            MSG("Warning: format is changed to S16_BE");
+        hwdata.format = SND_PCM_FORMAT_S16_BE;
+        break;
+    default:
+        return -1;
+    }
+    hwdata.rate = BE_INT(ap->sample_rate);
+    if (hwdata.rate < 2000 || hwdata.rate > 256000)
+        return -1;
+    hwdata.channels = BE_INT(ap->channels);
+    if (hwdata.channels < 1 || hwdata.channels > 128)
+        return -1;
+    if ((size_t)safe_read(fd, buffer + sizeof(AuHeader), BE_INT(ap->hdr_size) - sizeof(AuHeader)) != BE_INT(ap->hdr_size) - sizeof(AuHeader)) {
+        ERR("read error");
+        stopAndExit();
+    }
+    return 0;
+}
+
+void AlsaPlayer::set_params(void)
+{
+    snd_pcm_hw_params_t *hwparams;
+    snd_pcm_uframes_t period_size;
+    int err;
+    int dir;
+    unsigned int rate;
+    unsigned int periods;
+
+    snd_pcm_hw_params_alloca(&hwparams);
+    err = snd_pcm_hw_params_any(handle, hwparams);
+    if (err < 0) {
+        ERR("Broken configuration for this PCM: no configurations available");
+        stopAndExit();
+    }
+
+    /* Create the pipe for communication about stop requests. */
+    if (pipe(alsa_stop_pipe)) {
+        ERR("Stop pipe creation failed (%s)", strerror(errno));
+        stopAndExit();
+    }
+
+    /* Find how many descriptors we will get for poll(). */
+    alsa_fd_count = snd_pcm_poll_descriptors_count(handle);
+    if (alsa_fd_count <= 0){
+        ERR("Invalid poll descriptors count returned from ALSA.");
+        stopAndExit();
+    }
+
+    /* Create and fill in struct pollfd *alsa_poll_fds with ALSA descriptors. */
+    // alsa_poll_fds = (pollfd *)malloc ((alsa_fd_count + 1) * sizeof(struct pollfd));
+    alsa_poll_fds_barray.resize((alsa_fd_count + 1) * sizeof(struct pollfd));
+    alsa_poll_fds = (pollfd *)alsa_poll_fds_barray.data();
+    assert(alsa_poll_fds);
+    if ((err = snd_pcm_poll_descriptors(handle, alsa_poll_fds, alsa_fd_count)) < 0) {
+        ERR("Unable to obtain poll descriptors for playback: %s", snd_strerror(err));
+        stopAndExit();
+    }
+
+    /* Create a new pollfd structure for requests by alsa_stop(). */
+    struct pollfd alsa_stop_pipe_pfd;
+    alsa_stop_pipe_pfd.fd = alsa_stop_pipe[0];
+    alsa_stop_pipe_pfd.events = POLLIN;
+    alsa_stop_pipe_pfd.revents = 0;
+
+    /* Join this our own pollfd to the ALSAs ones. */
+    alsa_poll_fds[alsa_fd_count] = alsa_stop_pipe_pfd;
+    ++alsa_fd_count;
+
+    if (mmap_flag) {
+        snd_pcm_access_mask_t *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);
+        err = snd_pcm_hw_params_set_access_mask(handle, hwparams, mask);
+    } else if (interleaved)
+        err = snd_pcm_hw_params_set_access(handle, hwparams,
+                           SND_PCM_ACCESS_RW_INTERLEAVED);
+    else
+        err = snd_pcm_hw_params_set_access(handle, hwparams,
+                           SND_PCM_ACCESS_RW_NONINTERLEAVED);
+    if (err < 0) {
+        ERR("Error setting access type: %s", snd_strerror(err));
+        stopAndExit();
+    }
+    err = snd_pcm_hw_params_set_format(handle, hwparams, hwdata.format);
+    if (err < 0) {
+        ERR("Error setting sample format to %i: %s", hwdata.format, snd_strerror(err));
+        stopAndExit();
+    }
+    err = snd_pcm_hw_params_set_channels(handle, hwparams, hwdata.channels);
+    if (err < 0) {
+        ERR("Error setting channel count to %i: %s", hwdata.channels, snd_strerror(err));
+        stopAndExit();
+    }
+
+#if 0
+    err = snd_pcm_hw_params_set_periods_min(handle, hwparams, 2);
+    assert(err >= 0);
+#endif
+    rate = hwdata.rate;
+#if SND_LIB_MAJOR >= 1
+    err = snd_pcm_hw_params_set_rate_near(handle, hwparams, &hwdata.rate, 0);
+#else
+    err = snd_pcm_hw_params_set_rate_near(handle, hwparams, hwdata.rate, 0);
+#endif
+    assert(err >= 0);
+    if ((float)rate * 1.05 < hwdata.rate || (float)rate * 0.95 > hwdata.rate) {
+        MSG("Warning: rate is not accurate (requested = %iHz, got = %iHz)", rate, hwdata.rate);
+        MSG("         please, try the plug plugin (-Dplug:%s)", snd_pcm_name(handle));
+    }
+
+    period_size = m_defPeriodSize;
+    dir = 1;
+#if SND_LIB_MAJOR >= 1
+    err = snd_pcm_hw_params_set_period_size_near(handle, hwparams, &period_size, &dir);
+#else
+    err = snd_pcm_hw_params_set_period_size_near(handle, hwparams, period_size, &dir);
+#endif
+    if (err < 0) {
+        MSG("Setting period_size to %lu failed, but continuing: %s", period_size, snd_strerror(err));
+    }
+
+    periods = m_defPeriods;
+    dir = 1;
+#if SND_LIB_MAJOR >= 1
+    err = snd_pcm_hw_params_set_periods_near(handle, hwparams, &periods, &dir);
+#else
+    err = snd_pcm_hw_params_set_periods_near(handle, hwparams, periods, &dir);
+#endif
+    if (err < 0)
+        MSG("Unable to set number of periods to %i, but continuing: %s", periods, snd_strerror(err));
+
+    /* Install hw parameters. */
+    err = snd_pcm_hw_params(handle, hwparams);
+    if (err < 0) {
+        MSG("Unable to install hw params: %s", snd_strerror(err));
+        snd_pcm_hw_params_dump(hwparams, log);
+        stopAndExit();
+    }
+
+    /* Determine if device can pause. */
+    canPause = (1 == snd_pcm_hw_params_can_pause(hwparams));
+
+    /* Get final buffer size and calculate the chunk size we will pass to device. */
+#if SND_LIB_MAJOR >= 1
+    snd_pcm_hw_params_get_buffer_size(hwparams, &buffer_size);
+#else
+    buffer_size = snd_pcm_hw_params_get_buffer_size(hwparams);
+#endif
+    chunk_size = periods * period_size;
+
+    if (0 == chunk_size) {
+        ERR("Invalid periods or period_size.  Cannot continue.");
+        stopAndExit();
+    }
+
+    if (chunk_size == buffer_size)
+        MSG("WARNING: Shouldn't use chunk_size equal to buffer_size (%lu).  Continuing anyway.", chunk_size);
+
+    DBG("Final buffer_size = %lu, chunk_size = %lu, periods = %i, period_size = %lu, canPause = %i",
+        buffer_size, chunk_size, periods, period_size, canPause);
+
+    if (m_debugLevel >= 2)
+        snd_pcm_dump(handle, log);
+
+    bits_per_sample = snd_pcm_format_physical_width(hwdata.format);
+    bits_per_frame = bits_per_sample * hwdata.channels;
+    chunk_bytes = chunk_size * bits_per_frame / 8;
+    audioBuffer.resize(chunk_bytes);
+    audiobuf = audioBuffer.data();
+    if (audiobuf == NULL) {
+        ERR("not enough memory");
+        stopAndExit();
+    }
+}
+
+#ifndef timersub
+#define    timersub(a, b, result) \
+do { \
+    (result)->tv_sec = (a)->tv_sec - (b)->tv_sec; \
+    (result)->tv_usec = (a)->tv_usec - (b)->tv_usec; \
+    if ((result)->tv_usec < 0) { \
+        --(result)->tv_sec; \
+        (result)->tv_usec += 1000000; \
+    } \
+} while (0)
+#endif
+
+/* I/O error handler */
+void AlsaPlayer::xrun()
+{
+    snd_pcm_status_t *status;
+    int res;
+
+    snd_pcm_status_alloca(&status);
+    if ((res = snd_pcm_status(handle, status))<0) {
+        ERR("status error: %s", snd_strerror(res));
+        stopAndExit();
+    }
+    if (SND_PCM_STATE_XRUN == snd_pcm_status_get_state(status)) {
+        struct timeval now, diff, tstamp;
+        gettimeofday(&now, 0);
+        snd_pcm_status_get_trigger_tstamp(status, &tstamp);
+        timersub(&now, &tstamp, &diff);
+        MSG("%s!!! (at least %.3f ms long)",
+            stream == SND_PCM_STREAM_PLAYBACK ? "underrun" : "overrun",
+            diff.tv_sec * 1000 + diff.tv_usec / 1000.0);
+        if (m_debugLevel >= 2) {
+            DBG("Status:");
+            snd_pcm_status_dump(status, log);
+        }
+        if ((res = snd_pcm_prepare(handle))<0) {
+            ERR("xrun: prepare error: %s", snd_strerror(res));
+            stopAndExit();
+        }
+        return;        /* ok, data should be accepted again */
+    } if (SND_PCM_STATE_DRAINING == snd_pcm_status_get_state(status)) {
+        if (m_debugLevel >= 2) {
+            DBG("Status(DRAINING):");
+            snd_pcm_status_dump(status, log);
+        }
+        if (stream == SND_PCM_STREAM_CAPTURE) {
+            MSG("capture stream format change? attempting recover...");
+            if ((res = snd_pcm_prepare(handle))<0) {
+                ERR("xrun(DRAINING): prepare error: %s", snd_strerror(res));
+                stopAndExit();
+            }
+            return;
+        }
+    }
+    if (m_debugLevel >= 2) {
+        DBG("Status(R/W):");
+        snd_pcm_status_dump(status, log);
+    }
+    ERR("read/write error, state = %s", snd_pcm_state_name(snd_pcm_status_get_state(status)));
+    stopAndExit();
+}
+
+/* I/O suspend handler */
+void AlsaPlayer::suspend(void)
+{
+    int res;
+
+    MSG("Suspended. Trying resume. ");
+    while ((res = snd_pcm_resume(handle)) == -EAGAIN)
+        sleep(1);    /* wait until suspend flag is released */
+    if (res < 0) {
+        MSG("Failed. Restarting stream. ");
+        if ((res = snd_pcm_prepare(handle)) < 0) {
+            ERR("suspend: prepare error: %s", snd_strerror(res));
+            stopAndExit();
+        }
+    }
+    MSG("Suspend done.");
+}
+
+/* peak handler */
+void AlsaPlayer::compute_max_peak(char *data, size_t count)
+{
+    signed int val, max, max_peak = 0, perc;
+    size_t ocount = count;
+
+    switch (bits_per_sample) {
+    case 8: {
+        signed char *valp = (signed char *)data;
+        signed char mask = snd_pcm_format_silence(hwdata.format);
+        while (count-- > 0) {
+            val = *valp++ ^ mask;
+            val = abs(val);
+            if (max_peak < val)
+                max_peak = val;
+        }
+        break;
+    }
+    case 16: {
+        signed short *valp = (signed short *)data;
+        signed short mask = snd_pcm_format_silence_16(hwdata.format);
+        count /= 2;
+        while (count-- > 0) {
+            val = *valp++ ^ mask;
+            val = abs(val);
+            if (max_peak < val)
+                max_peak = val;
+        }
+        break;
+    }
+    case 32: {
+        signed int *valp = (signed int *)data;
+        signed int mask = snd_pcm_format_silence_32(hwdata.format);
+        count /= 4;
+        while (count-- > 0) {
+            val = *valp++ ^ mask;
+            val = abs(val);
+            if (max_peak < val)
+                max_peak = val;
+        }
+        break;
+    }
+    default:
+        break;
+    }
+    max = 1 << (bits_per_sample-1);
+    if (max <= 0)
+        max = 0x7fffffff;
+    DBG("Max peak (%li samples): %05i (0x%04x) ", (long)ocount, max_peak, max_peak);
+    if (bits_per_sample > 16)
+        perc = max_peak / (max / 100);
+    else
+        perc = max_peak * 100 / max;
+    for (val = 0; val < 20; val++)
+        if (val <= perc / 5)
+            kdDebug() << '#';
+        else
+            kdDebug() << ' ';
+    DBG(" %i%%", perc);
+}
+
+/*
+ *  Write to the ALSA pcm.
+ */
+
+ssize_t AlsaPlayer::pcm_write(char *data, size_t count)
+{
+    ssize_t r;
+    ssize_t result = 0;
+
+    if (sleep_min == 0 && count < chunk_size) {
+        DBG("calling snd_pcm_format_set_silence");
+        snd_pcm_format_set_silence(hwdata.format, data + count * bits_per_frame / 8, (chunk_size - count) * hwdata.channels);
+        count = chunk_size;
+    }
+    while (count > 0) {
+        DBG("calling writei_func, count = %i", count);
+        r = writei_func(handle, data, count);
+        DBG("writei_func returned %i", r);
+        if (-EAGAIN == r || (r >= 0 && (size_t)r < count)) {
+            DBG("r = %i calling snd_pcm_wait", r);
+            snd_pcm_wait(handle, 100);
+        } else if (-EPIPE == r) {
+            xrun();
+        } else if (-ESTRPIPE == r) {
+            suspend();
+        } else if (-EBUSY == r){
+            MSG("WARNING: sleeping while PCM BUSY");
+            usleep(1000);
+            continue;
+        } else if (r < 0) {
+            ERR("write error: %s", snd_strerror(r));
+            stopAndExit();
+        }
+        if (r > 0) {
+            if (m_debugLevel >= 1)
+                compute_max_peak(data, r * hwdata.channels);
+            result += r;
+            count -= r;
+            data += r * bits_per_frame / 8;
+        }
+        /* Report current state */
+        DBG("PCM state before polling: %s",
+            snd_pcm_state_name(snd_pcm_state(handle)));
+
+        int err = wait_for_poll(0);
+        if (err < 0) {
+            ERR("Wait for poll() failed");
+            return -1;
+        }
+        else if (err == 1){
+            MSG("Playback stopped");
+            /* Drop the playback on the sound device (probably
+               still in progress up till now) */
+            err = snd_pcm_drop(handle);
+            if (err < 0) {
+                ERR("snd_pcm_drop() failed: %s", snd_strerror(err));
+                return -1;
+            }
+            return -1;
+        }
+    }
+    return result;
+}
+
+/*
+ *  ok, let's play a .voc file
+ */
+
+ssize_t AlsaPlayer::voc_pcm_write(u_char *data, size_t count)
+{
+    ssize_t result = count, r;
+    size_t size;
+
+    while (count > 0) {
+        size = count;
+        if (size > chunk_bytes - buffer_pos)
+            size = chunk_bytes - buffer_pos;
+        memcpy(audiobuf + buffer_pos, data, size);
+        data += size;
+        count -= size;
+        buffer_pos += size;
+        if ((size_t)buffer_pos == chunk_bytes) {
+            if ((size_t)(r = pcm_write(audiobuf, chunk_size)) != chunk_size)
+                return r;
+            buffer_pos = 0;
+        }
+    }
+    return result;
+}
+
+void AlsaPlayer::voc_write_silence(unsigned x)
+{
+    unsigned l;
+    char *buf;
+
+    QByteArray buffer(chunk_bytes);
+    // buf = (char *) malloc(chunk_bytes);
+    buf = buffer.data();
+    if (buf == NULL) {
+        ERR("can't allocate buffer for silence");
+        return;        /* not fatal error */
+    }
+    snd_pcm_format_set_silence(hwdata.format, buf, chunk_size * hwdata.channels);
+    while (x > 0) {
+        l = x;
+        if (l > chunk_size)
+            l = chunk_size;
+        if (voc_pcm_write((u_char*)buf, l) != (ssize_t)l) {
+            ERR("write error");
+            stopAndExit();
+        }
+        x -= l;
+    }
+    // free(buf);
+}
+
+void AlsaPlayer::voc_pcm_flush(void)
+{
+    if (buffer_pos > 0) {
+        size_t b;
+        if (sleep_min == 0) {
+            if (snd_pcm_format_set_silence(hwdata.format, audiobuf + buffer_pos, chunk_bytes - buffer_pos * 8 / bits_per_sample) < 0)
+                MSG("voc_pcm_flush - silence error");
+            b = chunk_size;
+        } else {
+            b = buffer_pos * 8 / bits_per_frame;
+        }
+        if (pcm_write(audiobuf, b) != (ssize_t)b)
+            ERR("voc_pcm_flush error");
+    }
+    snd_pcm_drain(handle);
+}
+
+void AlsaPlayer::voc_play(int fd, int ofs, const char* name)
+{
+    int l;
+    VocBlockType *bp;
+    VocVoiceData *vd;
+    VocExtBlock *eb;
+    size_t nextblock, in_buffer;
+    u_char *data, *buf;
+    char was_extended = 0, output = 0;
+    u_short *sp, repeat = 0;
+    size_t silence;
+    off64_t filepos = 0;
+
+#define COUNT(x)    nextblock -= x; in_buffer -= x; data += x
+#define COUNT1(x)    in_buffer -= x; data += x
+
+    QByteArray buffer(64 * 1024);
+    // data = buf = (u_char *)malloc(64 * 1024);
+    data = buf = (u_char*)buffer.data();
+    buffer_pos = 0;
+    if (data == NULL) {
+        ERR("malloc error");
+        stopAndExit();
+    }
+    MSG("Playing Creative Labs Channel file '%s'...", name);
+    /* first we waste the rest of header, ugly but we don't need seek */
+    while (ofs > (ssize_t)chunk_bytes) {
+        if ((size_t)safe_read(fd, buf, chunk_bytes) != chunk_bytes) {
+            ERR("read error");
+            stopAndExit();
+        }
+        ofs -= chunk_bytes;
+    }
+    if (ofs) {
+        if (safe_read(fd, buf, ofs) != ofs) {
+            ERR("read error");
+            stopAndExit();
+        }
+    }
+    hwdata.format = DEFAULT_FORMAT;
+    hwdata.channels = 1;
+    hwdata.rate = DEFAULT_SPEED;
+    set_params();
+
+    in_buffer = nextblock = 0;
+    while (1) {
+          Fill_the_buffer:    /* need this for repeat */
+        if (in_buffer < 32) {
+            /* move the rest of buffer to pos 0 and fill the buf up */
+            if (in_buffer)
+                memcpy(buf, data, in_buffer);
+            data = buf;
+            if ((l = safe_read(fd, buf + in_buffer, chunk_bytes - in_buffer)) > 0)
+                in_buffer += l;
+            else if (!in_buffer) {
+                /* the file is truncated, so simulate 'Terminator' 
+                   and reduce the datablock for safe landing */
+                nextblock = buf[0] = 0;
+                if (l == -1) {
+//                    perror(name);
+                    stopAndExit();
+                }
+            }
+        }
+        while (!nextblock) {    /* this is a new block */
+            if (in_buffer < sizeof(VocBlockType))
+                goto __end;
+            bp = (VocBlockType *) data;
+            COUNT1(sizeof(VocBlockType));
+            nextblock = VOC_DATALEN(bp);
+            if (output)
+                MSG(" ");  /* write /n after ASCII-out */
+            output = 0;
+            switch (bp->type) {
+            case 0:
+#if 0
+                MSG("Terminator");
+#endif
+                return;        /* VOC-file stop */
+            case 1:
+                vd = (VocVoiceData *) data;
+                COUNT1(sizeof(VocVoiceData));
+                /* we need a SYNC, before we can set new SPEED, STEREO ... */
+
+                if (!was_extended) {
+                    hwdata.rate = (int) (vd->tc);
+                    hwdata.rate = 1000000 / (256 - hwdata.rate);
+#if 0
+                    MSG("Channel data %d Hz", dsp_speed);
+#endif
+                    if (vd->pack) {        /* /dev/dsp can't it */
+                        ERR("can't play packed .voc files");
+                        return;
+                    }
+                    if (hwdata.channels == 2)        /* if we are in Stereo-Mode, switch back */
+                        hwdata.channels = 1;
+                } else {    /* there was extended block */
+                    hwdata.channels = 2;
+                    was_extended = 0;
+                }
+                set_params();
+                break;
+            case 2:    /* nothing to do, pure data */
+#if 0
+                MSG("Channel continuation");
+#endif
+                break;
+            case 3:    /* a silence block, no data, only a count */
+                sp = (u_short *) data;
+                COUNT1(sizeof(u_short));
+                hwdata.rate = (int) (*data);
+                COUNT1(1);
+                hwdata.rate = 1000000 / (256 - hwdata.rate);
+                set_params();
+                silence = (((size_t) * sp) * 1000) / hwdata.rate;
+#if 0
+                MSG("Silence for %d ms", (int) silence);
+#endif
+                voc_write_silence(*sp);
+                break;
+            case 4:    /* a marker for syncronisation, no effect */
+                sp = (u_short *) data;
+                COUNT1(sizeof(u_short));
+#if 0
+                MSG("Marker %d", *sp);
+#endif
+                break;
+            case 5:    /* ASCII text, we copy to stderr */
+                output = 1;
+#if 0
+                MSG("ASCII - text :");
+#endif
+                break;
+            case 6:    /* repeat marker, says repeatcount */
+                /* my specs don't say it: maybe this can be recursive, but
+                   I don't think somebody use it */
+                repeat = *(u_short *) data;
+                COUNT1(sizeof(u_short));
+#if 0
+                MSG("Repeat loop %d times", repeat);
+#endif
+                if (filepos >= 0) {    /* if < 0, one seek fails, why test another */
+                    if ((filepos = lseek64(fd, 0, 1)) < 0) {
+                        ERR("can't play loops; %s isn't seekable", name);
+                        repeat = 0;
+                    } else {
+                        filepos -= in_buffer;    /* set filepos after repeat */
+                    }
+                } else {
+                    repeat = 0;
+                }
+                break;
+            case 7:    /* ok, lets repeat that be rewinding tape */
+                if (repeat) {
+                    if (repeat != 0xFFFF) {
+#if 0
+                        MSG("Repeat loop %d", repeat);
+#endif
+                        --repeat;
+                    }
+#if 0
+                    else
+                        MSG("Neverending loop");
+#endif
+                    lseek64(fd, filepos, 0);
+                    in_buffer = 0;    /* clear the buffer */
+                    goto Fill_the_buffer;
+                }
+#if 0
+                else
+                    MSG("End repeat loop");
+#endif
+                break;
+            case 8:    /* the extension to play Stereo, I have SB 1.0 :-( */
+                was_extended = 1;
+                eb = (VocExtBlock *) data;
+                COUNT1(sizeof(VocExtBlock));
+                hwdata.rate = (int) (eb->tc);
+                hwdata.rate = 256000000L / (65536 - hwdata.rate);
+                hwdata.channels = eb->mode == VOC_MODE_STEREO ? 2 : 1;
+                if (hwdata.channels == 2)
+                    hwdata.rate = hwdata.rate >> 1;
+                if (eb->pack) {        /* /dev/dsp can't it */
+                    ERR("can't play packed .voc files");
+                    return;
+                }
+#if 0
+                MSG("Extended block %s %d Hz",
+                     (eb->mode ? "Stereo" : "Mono"), dsp_speed);
+#endif
+                break;
+            default:
+                ERR("unknown blocktype %d. terminate.", bp->type);
+                return;
+            }    /* switch (bp->type) */
+        }        /* while (! nextblock)  */
+        /* put nextblock data bytes to dsp */
+        l = in_buffer;
+        if (nextblock < (size_t)l)
+            l = nextblock;
+        if (l) {
+            if (output) {
+                if (write(2, data, l) != l) {    /* to stderr */
+                    ERR("write error");
+                    stopAndExit();
+                }
+            } else {
+                if (voc_pcm_write(data, l) != l) {
+                    ERR("write error");
+                    stopAndExit();
+                }
+            }
+            COUNT(l);
+        }
+    }            /* while(1) */
+      __end:
+        voc_pcm_flush();
+        // free(buf);
+}
+/* that was a big one, perhaps somebody split it :-) */
+
+/* setting the globals for playing raw data */
+void AlsaPlayer::init_raw_data(void)
+{
+    hwdata = rhwdata;
+}
+
+/* calculate the data count to read from/to dsp */
+off64_t AlsaPlayer::calc_count(void)
+{
+    off64_t count;
+
+    if (timelimit == 0) {
+        count = pbrec_count;
+    } else {
+        count = snd_pcm_format_size(hwdata.format, hwdata.rate * hwdata.channels);
+        count *= (off64_t)timelimit;
+    }
+    return count < pbrec_count ? count : pbrec_count;
+}
+
+void AlsaPlayer::header(int /*rtype*/, const char* /*name*/)
+{
+//        fprintf(stderr, "%s %s '%s' : ",
+//            (stream == SND_PCM_STREAM_PLAYBACK) ? "Playing" : "Recording",
+//            fmt_rec_table[rtype].what,
+//            name);
+    QString channels;
+    if (hwdata.channels == 1)
+        channels = "Mono";
+    else if (hwdata.channels == 2)
+        channels = "Stereo";
+    else
+        channels = QString("Channels %1").arg(hwdata.channels);
+    DBG("Format: %s, Rate %d Hz, %s",
+        snd_pcm_format_description(hwdata.format),
+        hwdata.rate,
+        channels.ascii());
+}
+
+/* playing raw data */
+
+void AlsaPlayer::playback_go(int fd, size_t loaded, off64_t count, int rtype, const char *name)
+{
+    int l, r;
+    off64_t written = 0;
+    off64_t c;
+
+    if (m_debugLevel >= 1) header(rtype, name);
+    set_params();
+
+    while (loaded > chunk_bytes && written < count) {
+        if (pcm_write(audiobuf + written, chunk_size) <= 0)
+            return;
+        written += chunk_bytes;
+        loaded -= chunk_bytes;
+    }
+    if (written > 0 && loaded > 0)
+        memmove(audiobuf, audiobuf + written, loaded);
+
+    l = loaded;
+    while (written < count) {
+        do {
+            c = count - written;
+            if (c > chunk_bytes)
+                c = chunk_bytes;
+            c -= l;
+
+            if (c == 0)
+                break;
+            r = safe_read(fd, audiobuf + l, c);
+            if (r < 0) {
+//                perror(name);
+                stopAndExit();
+            }
+            fdcount += r;
+            if (r == 0)
+                break;
+            l += r;
+        } while (sleep_min == 0 && (size_t)l < chunk_bytes);
+        l = l * 8 / bits_per_frame;
+        DBG("calling pcm_write with %i frames.", l);
+        r = pcm_write(audiobuf, l);
+        DBG("pcm_write returned r = %i", r);
+        if (r < 0) return;
+        if (r != l)
+            break;
+        r = r * bits_per_frame / 8;
+        written += r;
+        l = 0;
+    }
+
+    DBG("Draining...");
+
+    /* We want the next "device ready" notification only when the buffer is completely empty. */
+    /* Do this by setting the avail_min to the buffer size. */
+    int err;
+    DBG("Getting swparams");
+    snd_pcm_sw_params_t *swparams;
+    snd_pcm_sw_params_alloca(&swparams);
+    err = snd_pcm_sw_params_current(handle, swparams);
+    if (err < 0) {
+        ERR("Unable to get current swparams: %s", snd_strerror(err));
+        return;
+    }
+    DBG("Setting avail min to %lu", buffer_size);
+    err = snd_pcm_sw_params_set_avail_min(handle, swparams, buffer_size);
+    if (err < 0) {
+        ERR("Unable to set avail min for playback: %s", snd_strerror(err));
+        return;
+    }
+    /* write the parameters to the playback device */
+    DBG("Writing swparams");
+    err = snd_pcm_sw_params(handle, swparams);
+    if (err < 0) {
+        ERR("Unable to set sw params for playback: %s", snd_strerror(err));
+        return;
+    }
+
+    DBG("Waiting for poll");
+    err = wait_for_poll(1);
+    if (err < 0) {
+        ERR("Wait for poll() failed");
+        return;
+    } else if (err == 1){
+        MSG("Playback stopped while draining");
+
+        /* Drop the playback on the sound device (probably
+           still in progress up till now) */
+        err = snd_pcm_drop(handle);
+        if (err < 0) {
+            ERR("snd_pcm_drop() failed: %s", snd_strerror(err));
+            return;
+        }
+    }
+    DBG("Draining completed");
+}
+
+/*
+ *  let's play or capture it (capture_type says VOC/WAVE/raw)
+ */
+
+void AlsaPlayer::playback(int fd)
+{
+    int ofs;
+    size_t dta;
+    ssize_t dtawave;
+
+    pbrec_count = LLONG_MAX;
+    fdcount = 0;
+
+    /* read the file header */
+    dta = sizeof(AuHeader);
+    if ((size_t)safe_read(fd, audiobuf, dta) != dta) {
+        ERR("read error");
+        stopAndExit();
+    }
+    if (test_au(fd, audiobuf) >= 0) {
+        rhwdata.format = hwdata.format;
+        pbrec_count = calc_count();
+        playback_go(fd, 0, pbrec_count, FORMAT_AU, name.ascii());
+        goto __end;
+    }
+    dta = sizeof(VocHeader);
+    if ((size_t)safe_read(fd, audiobuf + sizeof(AuHeader),
+         dta - sizeof(AuHeader)) != dta - sizeof(AuHeader)) {
+        ERR("read error");
+        stopAndExit();
+    }
+    if ((ofs = test_vocfile(audiobuf)) >= 0) {
+        pbrec_count = calc_count();
+        voc_play(fd, ofs, name.ascii());
+        goto __end;
+    }
+    /* read bytes for WAVE-header */
+    if ((dtawave = test_wavefile(fd, audiobuf, dta)) >= 0) {
+        pbrec_count = calc_count();
+        playback_go(fd, dtawave, pbrec_count, FORMAT_WAVE, name.ascii());
+    } else {
+        /* should be raw data */
+        init_raw_data();
+        pbrec_count = calc_count();
+        playback_go(fd, dta, pbrec_count, FORMAT_RAW, name.ascii());
+    }
+__end:
+    return;
+}
+
+/* Wait until ALSA is ready for more samples or stop() was called.
+   @return 0 if ALSA is ready for more input, +1 if a request to stop
+   the sound output was received and a negative value on error.  */
+int AlsaPlayer::wait_for_poll(int draining)
+{
+    unsigned short revents;
+    snd_pcm_state_t state;
+    int ret;
+
+    DBG("Waiting for poll");
+
+    /* Wait for certain events */
+    while (1) {
+        /* Simulated pause by not writing to alsa device, which will lead to an XRUN
+           when resumed. */
+        if (m_simulatedPause)
+            msleep(500);
+        else {
+
+            ret = poll(alsa_poll_fds, alsa_fd_count, -1);
+            DBG("activity on %d descriptors", ret);
+
+            /* Check for stop request from alsa_stop on the last file descriptors. */
+            if ((revents = alsa_poll_fds[alsa_fd_count-1].revents)) {
+                if (revents & POLLIN){
+                    DBG("stop requested");
+                    return 1;
+                }
+            }
+
+            /* Check the first count-1 descriptors for ALSA events */
+            snd_pcm_poll_descriptors_revents(handle, alsa_poll_fds, alsa_fd_count-1, &revents);
+
+            /* Ensure we are in the right state */
+            state = snd_pcm_state(handle);
+            DBG("State after poll returned is %s", snd_pcm_state_name(state));
+
+            if (SND_PCM_STATE_XRUN == state){
+                if (!draining){
+                    MSG("WARNING: Buffer underrun detected!");
+                    xrun();
+                    return 0;
+                }else{
+                    DBG("Playback terminated");
+                    return 0;
+                }
+            }
+
+            if (SND_PCM_STATE_SUSPENDED == state){
+                DBG("WARNING: Suspend detected!");
+                suspend();
+                return 0;
+            }
+
+            /* Check for errors */
+            if (revents & POLLERR) {
+                DBG("poll revents says POLLERR");
+                return -EIO;
+            }
+
+            /* Is ALSA ready for more input? */
+            if ((revents & POLLOUT)){
+                DBG("Ready for more input");
+                return 0;
+            }
+        }
+    }
+}
+
+#include "alsaplayer.moc"
+
+#undef DBG
+#undef MSG
+#undef ERR
+
+// vim: sw=4 ts=8 et
diff --git a/kttsd/players/alsaplayer/alsaplayer.h b/kttsd/players/alsaplayer/alsaplayer.h
new file mode 100644
index 0000000..acd465c
--- /dev/null
+++ b/kttsd/players/alsaplayer/alsaplayer.h
@@ -0,0 +1,187 @@
+/***************************************************** vim:set ts=4 sw=4 sts=4:
+  ALSA player.
+  -------------------
+  Copyright:
+  (C) 2005 by Gary Cramblitt <[email protected]>
+  -------------------
+  Original author: Gary Cramblitt <[email protected]>
+
+  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 Street, Fifth Floor, Boston, MA  02110-1301, USA.
+ ******************************************************************************/
+
+#ifndef ALSAPLAYER_H
+#define ALSAPLAYER_H
+
+// System includes.
+#include <alsa/asoundlib.h>
+#include <sys/poll.h>
+
+// Qt includes.
+#include <qstring.h>
+#include <qobject.h>
+#include <qthread.h>
+#include <qfile.h>
+#include <qmutex.h>
+
+// KDE includes.
+#include <config.h>
+#include <kdemacros.h>
+#include "kdeexportfix.h"
+#include <kurl.h>
+
+// AlsaPlayer includes.
+#include "formats.h"
+#include "player.h"
+
+#ifndef LLONG_MAX
+#define LLONG_MAX    9223372036854775807LL
+#endif
+
+#define DEFAULT_FORMAT		SND_PCM_FORMAT_U8
+#define DEFAULT_SPEED 		8000
+
+#define FORMAT_DEFAULT		-1
+#define FORMAT_RAW		0
+#define FORMAT_VOC		1
+#define FORMAT_WAVE		2
+#define FORMAT_AU		3
+
+static snd_pcm_sframes_t (*readi_func)(snd_pcm_t *handle, void *buffer, snd_pcm_uframes_t size);
+static snd_pcm_sframes_t (*writei_func)(snd_pcm_t *handle, const void *buffer, snd_pcm_uframes_t size);
+static snd_pcm_sframes_t (*readn_func)(snd_pcm_t *handle, void **bufs, snd_pcm_uframes_t size);
+static snd_pcm_sframes_t (*writen_func)(snd_pcm_t *handle, void **bufs, snd_pcm_uframes_t size);
+
+class KDE_EXPORT AlsaPlayer : public Player, QThread
+{
+    Q_OBJECT
+
+public:
+    AlsaPlayer(QObject* parent = 0, const char* name = 0, const QStringList& args=QStringList());
+    ~AlsaPlayer();
+
+    virtual void startPlay(const QString& file);
+    virtual void pause();
+    virtual void stop();
+
+    virtual void setVolume(float volume = 1.0);
+    virtual float volume() const;
+
+    virtual bool playing() const;
+    virtual bool paused() const;
+
+    virtual int totalTime() const;
+    virtual int currentTime() const;
+    virtual int position() const; // in this case not really the percent
+
+    virtual void seek(int seekTime);
+    virtual void seekPosition(int position);
+
+    virtual QStringList getPluginList( const QCString& classname );
+    virtual void setSinkName(const QString &sinkName);
+
+    virtual void setDebugLevel(uint level) { m_debugLevel = level; }
+    virtual void setPeriodSize(uint periodSize) { m_defPeriodSize = periodSize; }
+    virtual void setPeriods(uint periods) { m_defPeriods = periods; }
+
+protected:
+    virtual void run();
+
+private slots:
+
+private:
+    void init();
+    void cleanup();
+    void stopAndExit();
+    int wait_for_poll(int draining);
+
+    QString timestamp() const;
+
+    ssize_t safe_read(int fd, void *buf, size_t count);
+    int test_vocfile(void *buffer);
+    size_t test_wavefile_read(int fd, char *buffer, size_t *size, size_t reqsize, int line);
+    ssize_t test_wavefile(int fd, char *_buffer, size_t size);
+    int test_au(int fd, char *buffer);
+    void set_params(void);
+    void xrun();
+    void suspend(void);
+    void compute_max_peak(char *data, size_t count);
+    ssize_t pcm_write(char *data, size_t count);
+    ssize_t voc_pcm_write(u_char *data, size_t count);
+    void voc_write_silence(unsigned x);
+    void voc_pcm_flush(void);
+    void voc_play(int fd, int ofs, const char *name);
+    void init_raw_data(void);
+    off64_t calc_count(void);
+    void header(int rtype, const char *name);
+    void playback_go(int fd, size_t loaded, off64_t count, int rtype, const char *name);
+    void playback(int fd);
+
+    KURL m_currentURL;
+    float m_currentVolume;
+    QString m_pcmName;
+    char* pcm_name;
+    mutable QMutex m_mutex;
+
+    QFile audiofile;
+    QString name;
+    bool canPause;
+
+    snd_pcm_t *handle;
+    struct {
+        snd_pcm_format_t format;
+        unsigned int channels;
+        unsigned int rate;
+    } hwdata, rhwdata;
+    int timelimit;
+    int file_type;
+    unsigned int sleep_min;
+    int open_mode;
+    snd_pcm_stream_t stream;
+    int mmap_flag;
+    int interleaved;
+    QByteArray audioBuffer;
+    char *audiobuf;
+    snd_pcm_uframes_t chunk_size;
+    snd_pcm_uframes_t period_frames;
+    unsigned period_time;
+    unsigned buffer_time;
+    snd_pcm_uframes_t buffer_size;
+    int avail_min;
+    int start_delay;
+    int stop_delay;
+    int buffer_pos;
+    size_t bits_per_sample;
+    size_t bits_per_frame;
+    size_t chunk_bytes;
+    snd_output_t *log;
+    int fd;
+    off64_t pbrec_count;
+    off64_t fdcount;
+    int vocmajor;
+    int vocminor;
+
+    int alsa_stop_pipe[2];          /* Pipe for communication about stop requests*/
+    int alsa_fd_count;              /* Counter of descriptors to poll */
+    QByteArray alsa_poll_fds_barray;
+    struct pollfd *alsa_poll_fds;   /* Descriptors to poll */
+    unsigned int m_defPeriodSize;
+    unsigned int m_defPeriods;
+    unsigned int m_debugLevel;
+    bool m_simulatedPause;
+};
+
+#endif              // ALSAPLAYER_H
+
+// vim: sw=4 ts=8 et
diff --git a/kttsd/players/alsaplayer/alsaplugin.cpp b/kttsd/players/alsaplayer/alsaplugin.cpp
new file mode 100644
index 0000000..4bd700d
--- /dev/null
+++ b/kttsd/players/alsaplayer/alsaplugin.cpp
@@ -0,0 +1,29 @@
+/***************************************************** vim:set ts=4 sw=4 sts=4:
+  Generating the factories so ALSA can be used as an audio plug in.
+  -------------------
+  Copyright:
+  (C) 2005 by Gary Cramblitt <[email protected]>
+  -------------------
+  Original author: Gary Cramblitt <[email protected]>
+
+  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 Street, Fifth Floor, Boston, MA  02110-1301, USA.
+ ******************************************************************************/
+
+#include <kgenericfactory.h>
+
+#include "alsaplayer.h"
+
+K_EXPORT_COMPONENT_FACTORY( libkttsd_alsaplugin, KGenericFactory<AlsaPlayer>("kttsd_alsa") )
+
diff --git a/kttsd/players/alsaplayer/formats.h b/kttsd/players/alsaplayer/formats.h
new file mode 100644
index 0000000..2c4f724
--- /dev/null
+++ b/kttsd/players/alsaplayer/formats.h
@@ -0,0 +1,110 @@
+#ifndef FORMATS_H
+#define FORMATS_H		1
+
+#include <endian.h>
+#include <byteswap.h>
+
+/* Definitions for .VOC files */
+
+#define VOC_MAGIC_STRING	"Creative Voice File\x1A"
+#define VOC_ACTUAL_VERSION	0x010A
+#define VOC_SAMPLESIZE		8
+
+#define VOC_MODE_MONO		0
+#define VOC_MODE_STEREO		1
+
+#define VOC_DATALEN(bp)		((u_long)(bp->datalen) | \
+                         	((u_long)(bp->datalen_m) << 8) | \
+                         	((u_long)(bp->datalen_h) << 16) )
+
+typedef struct voc_header {
+	u_char magic[20];	/* must be MAGIC_STRING */
+	u_short headerlen;	/* Headerlength, should be 0x1A */
+	u_short version;	/* VOC-file version */
+	u_short coded_ver;	/* 0x1233-version */
+} VocHeader;
+
+typedef struct voc_blocktype {
+	u_char type;
+	u_char datalen;		/* low-byte    */
+	u_char datalen_m;	/* medium-byte */
+	u_char datalen_h;	/* high-byte   */
+} VocBlockType;
+
+typedef struct voc_voice_data {
+	u_char tc;
+	u_char pack;
+} VocVoiceData;
+
+typedef struct voc_ext_block {
+	u_short tc;
+	u_char pack;
+	u_char mode;
+} VocExtBlock;
+
+/* Definitions for Microsoft WAVE format */
+
+#if __BYTE_ORDER == __LITTLE_ENDIAN
+#define COMPOSE_ID(a,b,c,d)	((a) | ((b)<<8) | ((c)<<16) | ((d)<<24))
+#define LE_SHORT(v)		(v)
+#define LE_INT(v)		(v)
+#define BE_SHORT(v)		bswap_16(v)
+#define BE_INT(v)		bswap_32(v)
+#elif __BYTE_ORDER == __BIG_ENDIAN
+#define COMPOSE_ID(a,b,c,d)	((d) | ((c)<<8) | ((b)<<16) | ((a)<<24))
+#define LE_SHORT(v)		bswap_16(v)
+#define LE_INT(v)		bswap_32(v)
+#define BE_SHORT(v)		(v)
+#define BE_INT(v)		(v)
+#else
+#error "Wrong endian"
+#endif
+
+#define WAV_RIFF		COMPOSE_ID('R','I','F','F')
+#define WAV_WAVE		COMPOSE_ID('W','A','V','E')
+#define WAV_FMT			COMPOSE_ID('f','m','t',' ')
+#define WAV_DATA		COMPOSE_ID('d','a','t','a')
+#define WAV_PCM_CODE		1
+
+/* it's in chunks like .voc and AMIGA iff, but my source say there
+   are in only in this combination, so I combined them in one header;
+   it works on all WAVE-file I have
+ */
+typedef struct {
+	u_int magic;		/* 'RIFF' */
+	u_int length;		/* filelen */
+	u_int type;		/* 'WAVE' */
+} WaveHeader;
+
+typedef struct {
+	u_short format;		/* should be 1 for PCM-code */
+	u_short modus;		/* 1 Mono, 2 Stereo */
+	u_int sample_fq;	/* frequence of sample */
+	u_int byte_p_sec;
+	u_short byte_p_spl;	/* samplesize; 1 or 2 bytes */
+	u_short bit_p_spl;	/* 8, 12 or 16 bit */
+} WaveFmtBody;
+
+typedef struct {
+	u_int type;		/* 'data' */
+	u_int length;		/* samplecount */
+} WaveChunkHeader;
+
+/* Definitions for Sparc .au header */
+
+#define AU_MAGIC		COMPOSE_ID('.','s','n','d')
+
+#define AU_FMT_ULAW		1
+#define AU_FMT_LIN8		2
+#define AU_FMT_LIN16		3
+
+typedef struct au_header {
+	u_int magic;		/* '.snd' */
+	u_int hdr_size;		/* size of header (min 24) */
+	u_int data_size;	/* size of data */
+	u_int encoding;		/* see to AU_FMT_XXXX */
+	u_int sample_rate;	/* sample rate */
+	u_int channels;		/* number of channels (voices) */
+} AuHeader;
+
+#endif				/* FORMATS */
diff --git a/kttsd/players/alsaplayer/kttsd_alsaplugin.desktop b/kttsd/players/alsaplayer/kttsd_alsaplugin.desktop
new file mode 100644
index 0000000..36eec0b
--- /dev/null
+++ b/kttsd/players/alsaplayer/kttsd_alsaplugin.desktop
@@ -0,0 +1,89 @@
+[Desktop Entry]
+Name=KTTSD ALSA Plugin
+Name[br]=Lugent ALSA evit KTTSD
+Name[bs]=KTTSD ALSA dodatak
+Name[ca]=Connector ALSA pel KTTSD
+Name[cs]=KTTSD ALSA modul
+Name[da]=KTTSD ALSA plugin
+Name[de]=KTTSD ALSA-Modul
+Name[el]=KTTSD πρόσθετο ALSA
+Name[es]=Complemento ALSA de KTTSD
+Name[et]=KTTSD ALSA plugin
+Name[eu]=KTTSD-ren ALSA plugina
+Name[fa]=وصلۀ KTTSD ALSA
+Name[fi]=KTTSD ALSA -liitännäinen
+Name[fr]=Module ALSA pour KTTSD
+Name[ga]=Breiseán KTTSD ALSA
+Name[gl]=Plugin ALSA de KTTSD
+Name[hu]=KTTSD ALSA hangmodul
+Name[is]=KTTSD ALSA íforrit
+Name[it]=Plugin per ALSA di KTTSD
+Name[ja]=KTTSD ALSA プラグイン
+Name[ka]=KTTSD ALSA მოდული
+Name[km]= កម្មវិធី​ជំនួយ​ ALSA សម្រាប់ KTTSD
+Name[mk]=ALSA-приклучок за KTTSD
+Name[nb]=Alsas programtillegg for KTTSD
+Name[nds]=ALSA-Moduul för KTTSD
+Name[ne]=KTTSD ALSA प्लगइन
+Name[nl]=KTTSD ALSA-plugin
+Name[pa]=KTTSD ALSA ਪਲੱਗਿੰਨ
+Name[pl]=Wtyczka ALSA dla KTTSD
+Name[pt]='Plugin' ALSA do KTTSD
+Name[pt_BR]=Plug-in ALSA para o KTTSD
+Name[ru]=Модуль alsa для KTTSD
+Name[sk]=Modul KTTSD ALSA
+Name[sl]=Vstavek KTTSD za ALSA
+Name[sr]=ALSA прикључак за KTTSD
+Name[sr@Latn]=ALSA priključak za KTTSD
+Name[sv]=KTTSD-insticksprogram för ALSA
+Name[tg]=Модули KTTSD ALSA
+Name[tr]=KTTSD ALSA Eklentisi
+Name[uk]=Втулок ALSA для KTTSD
+Name[vi]=Trình bổ sung KTTSD ALSA
+Name[zh_TW]=KTTSd ALSA 外掛程式
+Comment=KTTSD ALSA audio plugin
+Comment[bg]=Аудио приставка на KTTSD за ALSA
+Comment[br]=Lugent klevet ALSA evit KTTSD
+Comment[bs]=KTTSD ALSA audio dodatak
+Comment[ca]=Connector d'àudio ALSA pel KTTSD
+Comment[cs]=KTTSD ALSA zvukový modul
+Comment[da]=KTTSD ALSA lyd-plugin
+Comment[de]=KTTSD ALSA-Audio-Modul
+Comment[el]=KTTSD πρόσθετο ήχου ALSA
+Comment[es]=Complemento de audio ALSA de KTTSD
+Comment[et]=KTTSD ALSA audioplugin
+Comment[eu]=KTTSD-ren ALSA audio-plugina
+Comment[fa]=وصلۀ صوتی KTTSD ALSA
+Comment[fi]=KTTSD ALSA ääniliitännäinen
+Comment[fr]=Module audio ALSA pour KTTSD
+Comment[ga]=Breiseán fuaime KTTSD ALSA
+Comment[gl]=Plugin de áudio ALSA para KTTSD
+Comment[hu]=KTTSD ALSA hangmodul
+Comment[is]=KTTSD ALSA hljóð íforrit
+Comment[it]=Plugin audio per ALSA di KTTSD
+Comment[ja]=KTTSD ALSA オーディオプラグイン
+Comment[ka]=KTTSD ALSA აუდიო მოდული
+Comment[km]= កម្មវិធី​ជំនួយអូឌីយ៉ូ​ ALSA សម្រាប់ KTTSD
+Comment[mk]=ALSA-аудиоприклучок за KTTSD
+Comment[nb]=Alsas lyd-programtillegg for KTTSD
+Comment[nds]=ALSA-Audiomoduul för KTTSD
+Comment[ne]=KTTSD ALSA अडियो प्लगइन
+Comment[nl]=KTTSD ALSA geluidsplugin
+Comment[pa]=KTTSD ALSA ਆਡੀਓ ਪਲੱਗਿੰਨ
+Comment[pl]=Wtyczka dźwięku ALSA dla KTTSD
+Comment[pt]='Plugin' de áudio ALSA do KTTSD
+Comment[pt_BR]=Plug-in de áudio ALSA para o KTTSD
+Comment[ru]=Модуль вывода звука KTTSD через alsa
+Comment[sk]=Modul KTTSD ALSA audio
+Comment[sl]=Vstavek KTTSD za zvok v ALSA
+Comment[sr]=ALSA аудио прикључак за KTTSD
+Comment[sr@Latn]=ALSA audio priključak za KTTSD
+Comment[sv]=KTTSD-ljudinsticksprogram för ALSA
+Comment[tg]=Модули KTTSD ALSA барои аудио
+Comment[tr]=KTTSD ALSA ses eklentisi
+Comment[uk]=Втулок аудіо ALSA для KTTSD
+Comment[vi]=Trình bổ sung âm thanh KTTSD ALSA
+Comment[zh_TW]=KTTSd ALSA 語音外掛程式
+Type=Service
+ServiceTypes=KTTSD/AudioPlugin
+X-KDE-Library=libkttsd_alsaplugin