![](https://seccdn.libravatar.org/avatar/e2145bc5cf53dda95c308a3c75e8fef3.jpg?s=120&d=mm&r=g)
Hello community, here is the log from the commit of package openal-soft for openSUSE:Factory checked in at Thu May 7 12:14:53 CEST 2009. -------- --- openal-soft/openal-soft.changes 2009-01-09 13:58:34.000000000 +0100 +++ /mounts/work_src_done/STABLE/openal-soft/openal-soft.changes 2009-05-06 16:22:04.000000000 +0200 @@ -1,0 +2,24 @@ +Wed May 6 16:09:16 CEST 2009 - lnussel@suse.de + +- new version 1.7.411 + * New table-based panning algorithm, allowing the center channel to be included in the mix + * Speaker arrangements are now configurable + * Added a new PortAudio backend + * Some changes to the ALSA device list + Standard enumeration will now only list a single ALSA playback device (for "default"), and there should be no more name clashes preventing a device with the same name from being used + * Low-pass filters now affect multi-channel sources + * Corrections for 6.1 channel placements + * Multi-channel sources are now re-mixed when using a different output mode + This prevents source channels from being lost if there isn't a matching output channel (eg. 5.1 sources on stereo output) + * Multi-channel source gains are now correctly clamped to the source's min/max gains + * The air absorption calculation now uses the correct distance + * The source room rolloff factor can now be set up to 10 + * Updated reverb code that better follows the reverb parameters +- add pulseaudio backend from git head and enable by default + +------------------------------------------------------------------- +Fri Apr 17 10:13:54 CEST 2009 - lnussel@suse.de + +- add shlib policy conform provides for libraries + +------------------------------------------------------------------- calling whatdependson for head-i586 Old: ---- 0001-install-pkgconfig-file-to-LIB_INSTALL_DIR.diff openal-soft-1.6.372.tar.bz2 New: ---- openal-soft-1.7.411-pulse.diff openal-soft-1.7.411.tar.bz2 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ Other differences: ------------------ ++++++ openal-soft.spec ++++++ --- /var/tmp/diff_new_pack.y17009/_old 2009-05-07 12:14:05.000000000 +0200 +++ /var/tmp/diff_new_pack.y17009/_new 2009-05-07 12:14:05.000000000 +0200 @@ -1,5 +1,5 @@ # -# spec file for package openal-soft (Version 1.6.372) +# spec file for package openal-soft (Version 1.7.411) # # Copyright (c) 2009 SUSE LINUX Products GmbH, Nuernberg, Germany. # @@ -17,9 +17,18 @@ # norootforbuild +%if 0%{?suse_version} > 1030 +%bcond_without pulseaudio +%else +%if 0%{?fedora_version} > 9 +%bcond_without pulseaudio +%else +%bcond_with pulseaudio +%endif +%endif Name: openal-soft -Version: 1.6.372 +Version: 1.7.411 Release: 1 License: LGPL v2.0 or later BuildRoot: %{_tmppath}/%{name}-%{version}-build @@ -34,11 +43,15 @@ Conflicts: openal <= 0.0.8 Provides: openal = 0.0.9 %if 0%{?mandriva_version} >= 2006 -BuildRequires: -alsa-plugins libalsa-oss-devel +BuildRequires: -alsa-plugins libalsa-devel %else BuildRequires: alsa-devel %endif -Patch0: 0001-install-pkgconfig-file-to-LIB_INSTALL_DIR.diff +%if %{with pulseaudio} +BuildRequires: pulseaudio-devel +%endif +# obsolete with next version +Patch0: openal-soft-1.7.411-pulse.diff %description OpenAL is an audio library designed in the spirit of OpenGL--machine @@ -80,6 +93,7 @@ License: LGPL v2.0 or later Summary: OpenAL Soft Group: System/Libraries +Provides: libopenal0 = %version-%release %if 0%{?suse_version} > 1010 Recommends: openal-soft %endif @@ -102,6 +116,7 @@ License: LGPL v2.0 or later Summary: OpenAL Soft Group: System/Libraries +Provides: libopenal1 = %version-%release %if 0%{?suse_version} > 1010 Recommends: openal-soft %endif @@ -142,6 +157,9 @@ make DESTDIR=$RPM_BUILD_ROOT install install -d %buildroot/etc/openal install -m644 libopenal.so.0 %buildroot%{_libdir} +# +# override driver ordering to prefer pulse +echo "drivers = pulse,alsa,oss,wave" > %buildroot/etc/openal/alsoft.conf %post -n libopenal0-soft -p /sbin/ldconfig @@ -158,6 +176,7 @@ %defattr(-,root,root) %doc alsoftrc.sample %dir /etc/openal +%config(noreplace) %attr(0644,root,root) /etc/openal/alsoft.conf %{_bindir}/openal-info %files -n libopenal1-soft @@ -179,6 +198,24 @@ %{_includedir}/AL/alext.h %changelog +* Wed May 06 2009 lnussel@suse.de +- new version 1.7.411 + * New table-based panning algorithm, allowing the center channel to be included in the mix + * Speaker arrangements are now configurable + * Added a new PortAudio backend + * Some changes to the ALSA device list + Standard enumeration will now only list a single ALSA playback device (for "default"), and there should be no more name clashes preventing a device with the same name from being used + * Low-pass filters now affect multi-channel sources + * Corrections for 6.1 channel placements + * Multi-channel sources are now re-mixed when using a different output mode + This prevents source channels from being lost if there isn't a matching output channel (eg. 5.1 sources on stereo output) + * Multi-channel source gains are now correctly clamped to the source's min/max gains + * The air absorption calculation now uses the correct distance + * The source room rolloff factor can now be set up to 10 + * Updated reverb code that better follows the reverb parameters +- add pulseaudio backend from git head and enable by default +* Fri Apr 17 2009 lnussel@suse.de +- add shlib policy conform provides for libraries * Fri Jan 09 2009 lnussel@suse.de - new version 1.6.372 * Channel volumes are now ramped from source position changes and when starting playback, to help prevent pops and clicks ++++++ openal-soft-1.7.411-pulse.diff ++++++
From b2b9161f7dca3be983c7876df5210b78350d1645 Mon Sep 17 00:00:00 2001 From: Chris Robinson <chris.kcat@gmail.com> Date: Thu, 16 Apr 2009 05:17:42 -0700 Subject: [PATCH] Add a PulseAudio backend
--- Alc/ALc.c | 3 + Alc/pulseaudio.c | 475 +++++++++++++++++++++++++++++++++++++++++++++ CMakeLists.txt | 18 ++- OpenAL32/Include/alMain.h | 1 + alsoftrc.sample | 5 +- config.h.in | 3 + 6 files changed, 503 insertions(+), 2 deletions(-) create mode 100644 Alc/pulseaudio.c diff --git a/Alc/ALc.c b/Alc/ALc.c index 574de76..f3ff580 100644 --- a/Alc/ALc.c +++ b/Alc/ALc.c @@ -71,6 +71,9 @@ static struct { #ifdef HAVE_PORTAUDIO { "port", alc_pa_init, EmptyFuncs }, #endif +#ifdef HAVE_PULSEAUDIO + { "pulse", alc_pulse_init, EmptyFuncs }, +#endif { "wave", alc_wave_init, EmptyFuncs }, diff --git a/Alc/pulseaudio.c b/Alc/pulseaudio.c new file mode 100644 index 0000000..6ba0255 --- /dev/null +++ b/Alc/pulseaudio.c @@ -0,0 +1,475 @@ +/** + * OpenAL cross platform audio library + * Copyright (C) 2009 by Konstantinos Natsakis <konstantinos.natsakis@gmail.com> + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library 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 + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + * Or go to http://www.gnu.org/copyleft/lgpl.html + */ + +#include "config.h" + +#include "alMain.h" + +#include <pulse/pulseaudio.h> + +#if PA_API_VERSION == 11 +#define PA_STREAM_ADJUST_LATENCY 0x2000U +static inline int PA_STREAM_IS_GOOD(pa_stream_state_t x) +{ + return (x == PA_STREAM_CREATING || x == PA_STREAM_READY); +} +static inline int PA_CONTEXT_IS_GOOD(pa_context_state_t x) +{ + return (x == PA_CONTEXT_CONNECTING || x == PA_CONTEXT_AUTHORIZING || + x == PA_CONTEXT_SETTING_NAME || x == PA_CONTEXT_READY); +} +#define PA_STREAM_IS_GOOD PA_STREAM_IS_GOOD +#define PA_CONTEXT_IS_GOOD PA_CONTEXT_IS_GOOD +#elif PA_API_VERSION != 12 +#error Invalid PulseAudio API version +#endif + +typedef struct { + ALCdevice *device; + + ALCenum format; + ALCuint samples; + ALCuint frequency; + ALCuint frame_size; + + RingBuffer *ring; + + pa_buffer_attr attr; + pa_sample_spec spec; + + char path_name[PATH_MAX]; + const char *context_name; + const char *stream_name; + + pa_threaded_mainloop *loop; + + pa_stream *stream; + pa_context *context; +} pulse_data; + +static char *pulse_device; +static char *pulse_capture_device; + +// PulseAudio Event Callbacks {{{ +static void stream_state_callback(pa_stream *stream, void *pdata) //{{{ +{ + pulse_data *data = pdata; + + switch(pa_stream_get_state(stream)) + { + case PA_STREAM_READY: + AL_PRINT("%s: %s ready!\n", data->context_name, data->stream_name); + break; + + case PA_STREAM_FAILED: + AL_PRINT("%s: %s: Connection failed: %s\n", data->context_name, + data->stream_name, pa_strerror(pa_context_errno(data->context))); + break; + + case PA_STREAM_TERMINATED: + AL_PRINT("%s: %s terminated!\n", data->context_name, data->stream_name); + break; + + default: + break; + } + + pa_threaded_mainloop_signal(data->loop, 1); +} //}}} + +static void context_state_callback(pa_context *context, void *pdata) //{{{ +{ + pulse_data *data = pdata; + + switch(pa_context_get_state(context)) + { + case PA_CONTEXT_READY: + AL_PRINT("%s ready!\n", data->context_name); + break; + + case PA_CONTEXT_FAILED: + AL_PRINT("%s: Connection failed: %s\n", data->context_name, + pa_strerror(pa_context_errno(context))); + break; + + case PA_CONTEXT_TERMINATED: + AL_PRINT("%s terminated!\n", data->context_name); + break; + + default: + break; + } + + pa_threaded_mainloop_signal(data->loop, 1); +} //}}} +//}}} + +// PulseAudio I/O Callbacks //{{{ +static void stream_write_callback(pa_stream *stream, size_t len, void *pdata) //{{{ +{ + ALCdevice *Device = pdata; + void *buf = pa_xmalloc0(len); + + SuspendContext(NULL); + aluMixData(Device->Context, buf, len, Device->Format); + ProcessContext(NULL); + + pa_stream_write(stream, buf, len, pa_xfree, 0, PA_SEEK_RELATIVE); +} //}}} + +static void stream_read_callback(pa_stream *stream, size_t length, void *pdata) //{{{ +{ + ALCdevice *Device = pdata; + pulse_data *data = Device->ExtraData; + const void *buf; + + if(pa_stream_peek(stream, &buf, &length) < 0) + { + AL_PRINT("pa_stream_peek() failed: %s\n", + pa_strerror(pa_context_errno(data->context))); + return; + } + + assert(buf); + assert(length); + + length /= data->frame_size; + + if(data->samples < length) + AL_PRINT("stream_read_callback: buffer overflow!\n"); + + WriteRingBuffer(data->ring, buf, (length<data->samples) ? length : data->samples); + + pa_stream_drop(stream); +} //}}} +//}}} + +static ALCboolean pulse_open(ALCdevice *device, ALCchar *device_name, ALCenum format, ALCuint samples, ALCuint frequency) //{{{ +{ + pulse_data *data = pa_xmalloc0(sizeof(pulse_data)); + + data->device = device; + data->format = format; + data->samples = samples; + data->frequency = frequency; + data->frame_size = aluBytesFromFormat(format) * aluChannelsFromFormat(format); + + if(pa_get_binary_name(data->path_name, sizeof(data->path_name))) + data->context_name = pa_path_get_filename(data->path_name); + else + data->context_name = "OpenAL Soft"; + + if(!(data->ring = CreateRingBuffer(data->frame_size, data->samples))) + { + pa_xfree(data); + return ALC_FALSE; + } + + device->ExtraData = data; + device->szDeviceName = device_name; + + data->attr.minreq = -1; + data->attr.prebuf = -1; + data->attr.maxlength = -1; + + if(device->IsCaptureDevice) + { + data->attr.tlength = -1; + data->attr.fragsize = data->frame_size * data->samples / 2; + data->stream_name = "Capture Stream"; + } + else + { + data->attr.tlength = data->frame_size * (device->UpdateSize&~3); + data->attr.fragsize = -1; + data->stream_name = "Playback Stream"; + } + + data->spec.rate = data->frequency; + data->spec.channels = aluChannelsFromFormat(data->format); + + switch(aluBytesFromFormat(data->format)) + { + case 1: + data->spec.format = PA_SAMPLE_U8; + break; + case 2: + data->spec.format = PA_SAMPLE_S16NE; + break; + default: + AL_PRINT("Unknown format: %x\n", data->format); + goto out2; + } + + if(pa_sample_spec_valid(&data->spec) == 0) + { + AL_PRINT("Invalid sample format\n"); + goto out2; + } + + if(!(data->loop = pa_threaded_mainloop_new())) + { + AL_PRINT("pa_threaded_mainloop_new() failed!\n"); + goto out2; + } + + if(pa_threaded_mainloop_start(data->loop) < 0) + { + AL_PRINT("pa_threaded_mainloop_start() failed\n"); + goto out3; + } + + pa_threaded_mainloop_lock(data->loop); + + data->context = pa_context_new(pa_threaded_mainloop_get_api(data->loop), data->context_name); + if(!data->context) + { + AL_PRINT("pa_context_new() failed: %s\n", + pa_strerror(pa_context_errno(data->context))); + + pa_threaded_mainloop_unlock(data->loop); + goto out3; + } + + pa_context_set_state_callback(data->context, context_state_callback, data); + + if(pa_context_connect(data->context, NULL, PA_CONTEXT_NOAUTOSPAWN, NULL) < 0) + { + AL_PRINT("Context did not connect: %s\n", + pa_strerror(pa_context_errno(data->context))); + + pa_context_unref(data->context); + pa_threaded_mainloop_unlock(data->loop); + + data->context = NULL; + goto out3; + } + + while(pa_context_get_state(data->context) != PA_CONTEXT_READY) + { + if(!PA_CONTEXT_IS_GOOD(pa_context_get_state(data->context))) + { + pa_context_unref(data->context); + pa_threaded_mainloop_unlock(data->loop); + + data->context = NULL; + goto out3; + } + + pa_threaded_mainloop_wait(data->loop); + pa_threaded_mainloop_accept(data->loop); + } + + data->stream = pa_stream_new(data->context, data->stream_name, &data->spec, NULL); + if(!data->stream) + { + AL_PRINT("pa_stream_new() failed: %s\n", + pa_strerror(pa_context_errno(data->context))); + + pa_threaded_mainloop_unlock(data->loop); + goto out4; + } + + pa_stream_set_state_callback(data->stream, stream_state_callback, data); + + if(device->IsCaptureDevice) + { + if(pa_stream_connect_record(data->stream, NULL, &data->attr, PA_STREAM_ADJUST_LATENCY) < 0) + { + AL_PRINT("Stream did not connect: %s\n", + pa_strerror(pa_context_errno(data->context))); + + pa_stream_unref(data->stream); + pa_threaded_mainloop_unlock(data->loop); + + data->stream = NULL; + goto out4; + } + } + else + { + pa_stream_set_write_callback(data->stream, stream_write_callback, device); + + if(pa_stream_connect_playback(data->stream, NULL, &data->attr, PA_STREAM_ADJUST_LATENCY, NULL, NULL) < 0) + { + AL_PRINT("Stream did not connect: %s\n", + pa_strerror(pa_context_errno(data->context))); + + pa_stream_unref(data->stream); + pa_threaded_mainloop_unlock(data->loop); + + data->stream = NULL; + goto out4; + } + } + + while(pa_stream_get_state(data->stream) != PA_STREAM_READY) + { + if(!PA_STREAM_IS_GOOD(pa_stream_get_state(data->stream))) + { + pa_stream_unref(data->stream); + pa_threaded_mainloop_unlock(data->loop); + + data->stream = NULL; + goto out4; + } + + pa_threaded_mainloop_wait(data->loop); + pa_threaded_mainloop_accept(data->loop); + } + + device->UpdateSize /= 4; + pa_threaded_mainloop_unlock(data->loop); + + return ALC_TRUE; + +out4: + pa_threaded_mainloop_lock(data->loop); + + pa_context_disconnect(data->context); + pa_context_unref(data->context); + + pa_threaded_mainloop_unlock(data->loop); +out3: + pa_threaded_mainloop_stop(data->loop); + pa_threaded_mainloop_free(data->loop); +out2: + device->ExtraData = NULL; + device->szDeviceName = NULL; + DestroyRingBuffer(data->ring); + + pa_xfree(data); + return ALC_FALSE; +} //}}} + +static void pulse_close(ALCdevice *device) //{{{ +{ + pulse_data *data = device->ExtraData; + + pa_threaded_mainloop_lock(data->loop); + + pa_stream_disconnect(data->stream); + pa_stream_unref(data->stream); + + pa_context_disconnect(data->context); + pa_context_unref(data->context); + + pa_threaded_mainloop_unlock(data->loop); + + pa_threaded_mainloop_stop(data->loop); + pa_threaded_mainloop_free(data->loop); + + device->ExtraData = NULL; + device->szDeviceName = NULL; + DestroyRingBuffer(data->ring); + + pa_xfree(data); +} //}}} +//}}} + +// OpenAL {{{ +static ALCboolean pulse_open_playback(ALCdevice *device, const ALCchar *device_name) //{{{ +{ + if(device_name) + { + if(strcmp(device_name, pulse_device) != 0) + return ALC_FALSE; + } + + return pulse_open(device, pulse_device, device->Format, 0, device->Frequency); +} //}}} + +static void pulse_close_playback(ALCdevice *device) //{{{ +{ + pulse_close(device); +} //}}} + +static ALCboolean pulse_open_capture(ALCdevice *device, const ALCchar *device_name, ALCuint frequency, ALCenum format, ALCsizei samples) //{{{ +{ + if(device_name) + { + if(strcmp(device_name, pulse_capture_device) != 0) + return ALC_FALSE; + } + + return pulse_open(device, pulse_capture_device, format, samples, frequency); +} //}}} + +static void pulse_close_capture(ALCdevice *device) //{{{ +{ + pulse_close(device); +} //}}} + +static void pulse_start_capture(ALCdevice *device) //{{{ +{ + pulse_data *data = device->ExtraData; + + pa_threaded_mainloop_lock(data->loop); + pa_stream_set_read_callback(data->stream, stream_read_callback, device); + pa_threaded_mainloop_unlock(data->loop); +} //}}} + +static void pulse_stop_capture(ALCdevice *device) //{{{ +{ + pulse_data *data = device->ExtraData; + + pa_threaded_mainloop_lock(data->loop); + pa_stream_set_read_callback(data->stream, NULL, NULL); + pa_threaded_mainloop_unlock(data->loop); +} //}}} + +static void pulse_capture_samples(ALCdevice *device, ALCvoid *buffer, ALCuint samples) //{{{ +{ + pulse_data *data = device->ExtraData; + ALCuint available = RingBufferSize(data->ring); + + if(available < samples) + SetALCError(ALC_INVALID_VALUE); + else + ReadRingBuffer(data->ring, buffer, samples); +} //}}} + +static ALCuint pulse_available_samples(ALCdevice *device) //{{{ +{ + pulse_data *data = device->ExtraData; + return RingBufferSize(data->ring); +} //}}} + +BackendFuncs pulse_funcs = { //{{{ + pulse_open_playback, + pulse_close_playback, + pulse_open_capture, + pulse_close_capture, + pulse_start_capture, + pulse_stop_capture, + pulse_capture_samples, + pulse_available_samples +}; //}}} + +void alc_pulse_init(BackendFuncs *func_list) //{{{ +{ + *func_list = pulse_funcs; + + pulse_device = AppendDeviceList("PulseAudio Software"); + AppendAllDeviceList(pulse_device); + + pulse_capture_device = AppendCaptureDeviceList("PulseAudio Capture"); +} //}}} +//}}} diff --git a/CMakeLists.txt b/CMakeLists.txt index 29350bc..0b59ab5 100644 --- a/CMakeLists.txt +++ b/CMakeLists.txt @@ -26,7 +26,8 @@ OPTION(OSS "Check for OSS backend" ON) OPTION(SOLARIS "Check for Solaris backend" ON) OPTION(DSOUND "Check for DirectSound backend" ON) OPTION(WINMM "Check for Windows Multimedia backend" ON) -OPTION(PORTAUDIO "Check for PortAudio backend" ON) +OPTION(PORTAUDIO "Check for PortAudio backend" ON) +OPTION(PULSEAUDIO "Check for PulseAudio backend" ON) OPTION(DLOPEN "Check for the dlopen API for loading optional libs" ON) @@ -340,6 +341,21 @@ IF(PORTAUDIO) ENDIF() ENDIF() +# Check PortAudio backend +IF(PULSEAUDIO) + CHECK_INCLUDE_FILE(pulse/pulseaudio.h HAVE_PULSE_PULSEAUDIO_H) + IF(HAVE_PULSE_PULSEAUDIO_H) + CHECK_LIBRARY_EXISTS(pulse pa_context_new "" HAVE_LIBPULSE) + IF(HAVE_LIBPULSE) + SET(HAVE_PULSEAUDIO 1) + SET(ALC_OBJS ${ALC_OBJS} Alc/pulseaudio.c) + SET(BACKENDS "${BACKENDS} PulseAudio \(linked\),") + + SET(EXTRA_LIBS pulse ${EXTRA_LIBS}) + ENDIF() + ENDIF() +ENDIF() + # This is always available SET(BACKENDS "${BACKENDS} WaveFile") diff --git a/OpenAL32/Include/alMain.h b/OpenAL32/Include/alMain.h index 68b176e..8791af2 100644 --- a/OpenAL32/Include/alMain.h +++ b/OpenAL32/Include/alMain.h @@ -152,6 +152,7 @@ void alcDSoundInit(BackendFuncs *func_list); void alcWinMMInit(BackendFuncs *FuncList); void alc_pa_init(BackendFuncs *func_list); void alc_wave_init(BackendFuncs *func_list); +void alc_pulse_init(BackendFuncs *func_list); struct ALCdevice_struct diff --git a/alsoftrc.sample b/alsoftrc.sample index 1e9cdce..91218ca 100644 --- a/alsoftrc.sample +++ b/alsoftrc.sample @@ -59,7 +59,7 @@ drivers = # Sets the backend driver list order, comma-seperated. Unknown # backends and duplicated names are ignored, and unlisted backends # won't be considered for use. An empty list means the default. # Default is: - # alsa,oss,solaris,dsound,winmm,port,wave + # alsa,oss,solaris,dsound,winmm,port,pulse,wave excludefx = # Sets which effects to exclude, preventing apps from using them. # This can help for apps that try to use effects which are too CPU @@ -142,6 +142,9 @@ device = -1 # Sets the device index for output. Negative values will use the periods = 4 # Sets the number of update buffers. Default is 4 +[pulse] # PulseAudio backend stuff + # Nothing yet... + [wave] # Wave File Writer stuff file = # Sets the filename of the wave file to write to. An empty name # prevents the backend from opening, even when explicitly requested. diff --git a/config.h.in b/config.h.in index 7f93267..93c13a6 100644 --- a/config.h.in +++ b/config.h.in @@ -22,6 +22,9 @@ /* Define if we have the PortAudio backend */ #cmakedefine HAVE_PORTAUDIO +/* Define if we have the PulseAudio backend */ +#cmakedefine HAVE_PULSEAUDIO + /* Define if we have dlfcn.h */ #cmakedefine HAVE_DLFCN_H -- 1.6.2.1 ++++++ openal-soft-1.6.372.tar.bz2 -> openal-soft-1.7.411.tar.bz2 ++++++ ++++ 2828 lines of diff (skipped) ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ Remember to have fun... -- To unsubscribe, e-mail: opensuse-commit+unsubscribe@opensuse.org For additional commands, e-mail: opensuse-commit+help@opensuse.org