rpms/ffmpeg/F-10 ffmpeg-r15348.patch, NONE,
1.1 ffmpeg-r16791-16793.patch, NONE,
1.1 ffmpeg-r17163-17311.patch, NONE, 1.1 ffmpeg-snapshot.sh, 1.2,
1.3 ffmpeg.spec, 1.11, 1.12
Dominik Mierzejewski
rathann at rpmfusion.org
Mon Oct 26 01:39:28 CET 2009
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Author: rathann
Update of /cvs/free/rpms/ffmpeg/F-10
In directory se02.es.rpmfusion.net:/tmp/cvs-serv8658
Modified Files:
ffmpeg-snapshot.sh ffmpeg.spec
Added Files:
ffmpeg-r15348.patch ffmpeg-r16791-16793.patch
ffmpeg-r17163-17311.patch
Log Message:
* Mon Oct 26 2009 Dominik Mierzejewski <rpm at greysector.net> - 0.4.9-0.56.20080908
- backport audio resampler fixes (bug #426)
- add a requirement on exact EVR between main package and -libs
so that "yum update ffmpeg" works as expected
- install presets (backport)
- use bindir instead of prefix/bin
ffmpeg-r15348.patch:
--- NEW FILE ffmpeg-r15348.patch ---
Index: ffpresets/libx264-fastfirstpass.ffpreset
===================================================================
--- ffpresets/libx264-fastfirstpass.ffpreset (revision 0)
+++ ffpresets/libx264-fastfirstpass.ffpreset (revision 15348)
@@ -0,0 +1,21 @@
+coder=1
+flags=+loop
+cmp=+chroma
+partitions=-parti8x8-parti4x4-partp8x8-partp4x4-partb8x8
+me=dia
+subq=1
+me_range=16
+g=250
+keyint_min=25
+sc_threshold=40
+i_qfactor=0.71
+b_strategy=1
+qcomp=0.6
+qmin=10
+qmax=51
+qdiff=4
+refs=1
+directpred=1
+bidir_refine=0
+trellis=0
+flags2=-bpyramid-wpred-brdo-mixed_refs-dct8x8+fastpskip
Index: ffpresets/libx264-default.ffpreset
===================================================================
--- ffpresets/libx264-default.ffpreset (revision 0)
+++ ffpresets/libx264-default.ffpreset (revision 15348)
@@ -0,0 +1,18 @@
+coder=1
+flags=+loop
+cmp=+chroma
+partitions=+parti8x8+parti4x4+partp8x8+partb8x8
+me=hex
+subq=5
+me_range=16
+g=250
+keyint_min=25
+sc_threshold=40
+i_qfactor=0.71
+b_strategy=1
+qcomp=0.6
+qmin=10
+qmax=51
+qdiff=4
+directpred=1
+flags2=+fastpskip
Index: ffpresets/libx264-max.ffpreset
===================================================================
--- ffpresets/libx264-max.ffpreset (revision 0)
+++ ffpresets/libx264-max.ffpreset (revision 15348)
@@ -0,0 +1,22 @@
+coder=1
+flags=+loop
+cmp=+chroma
+partitions=+parti8x8+parti4x4+partp8x8+partp4x4+partb8x8
+me=tesa
+subq=7
+me_range=32
+g=250
+keyint_min=25
+sc_threshold=40
+i_qfactor=0.71
+b_strategy=1
+qcomp=0.6
+qmin=10
+qmax=51
+qdiff=4
+bf=16
+refs=16
+directpred=3
+bidir_refine=1
+trellis=2
+flags2=+bpyramid+wpred+brdo+mixed_refs+dct8x8-fastpskip
Index: ffpresets/libx264-hq.ffpreset
===================================================================
--- ffpresets/libx264-hq.ffpreset (revision 0)
+++ ffpresets/libx264-hq.ffpreset (revision 15348)
@@ -0,0 +1,22 @@
+coder=1
+flags=+loop
+cmp=+chroma
+partitions=+parti8x8+parti4x4+partp8x8+partb8x8
+me=umh
+subq=7
+me_range=16
+g=250
+keyint_min=25
+sc_threshold=40
+i_qfactor=0.71
+b_strategy=1
+qcomp=0.6
+qmin=10
+qmax=51
+qdiff=4
+bf=16
+refs=4
+directpred=3
+bidir_refine=1
+trellis=1
+flags2=+bpyramid+wpred+brdo+mixed_refs+dct8x8+fastpskip
Index: ffpresets/libx264-normal.ffpreset
===================================================================
--- ffpresets/libx264-normal.ffpreset (revision 0)
+++ ffpresets/libx264-normal.ffpreset (revision 15348)
@@ -0,0 +1,22 @@
+coder=1
+flags=+loop
+cmp=+chroma
+partitions=+parti8x8+parti4x4+partp8x8+partb8x8
+me=hex
+subq=6
+me_range=16
+g=250
+keyint_min=25
+sc_threshold=40
+i_qfactor=0.71
+b_strategy=1
+qcomp=0.6
+qmin=10
+qmax=51
+qdiff=4
+bf=16
+refs=2
+directpred=3
+bidir_refine=1
+trellis=0
+flags2=+bpyramid+wpred+dct8x8+fastpskip
ffmpeg-r16791-16793.patch:
--- NEW FILE ffmpeg-r16791-16793.patch ---
Index: configure
===================================================================
--- configure (revision 16790)
+++ configure (revision 16793)
@@ -64,6 +64,7 @@
echo " --disable-logging do not log configure debug information"
echo " --prefix=PREFIX install in PREFIX [$prefix]"
echo " --bindir=DIR install binaries in DIR [PREFIX/bin]"
+ echo " --datadir=DIR install data files in DIR [PREFIX/share/ffmpeg]"
echo " --libdir=DIR install libs in DIR [PREFIX/lib]"
echo " --shlibdir=DIR install shared libs in DIR [PREFIX/lib]"
echo " --incdir=DIR install includes in DIR [PREFIX/include]"
@@ -264,6 +265,10 @@
echo "$@" | tr ABCDEFGHIJKLMNOPQRSTUVWXYZ abcdefghijklmnopqrstuvwxyz
}
+c_escape(){
+ echo "$*" | sed 's/["\\]/\\\0/g'
+}
+
set_all(){
value=$1
shift
@@ -914,6 +919,7 @@
PATHS_LIST='
bindir
+ datadir
incdir
libdir
mandir
@@ -1113,6 +1119,7 @@
# installation paths
prefix_default="/usr/local"
bindir_default='${prefix}/bin'
+datadir_default='${prefix}/share/ffmpeg'
incdir_default='${prefix}/include'
libdir_default='${prefix}/lib'
mandir_default='${prefix}/share/man'
@@ -2258,6 +2265,7 @@
echo "#ifndef FFMPEG_CONFIG_H" >> $TMPH
echo "#define FFMPEG_CONFIG_H" >> $TMPH
echo "#define FFMPEG_CONFIGURATION \"$FFMPEG_CONFIGURATION\"" >> $TMPH
+echo "#define FFMPEG_DATADIR \"$(eval c_escape $datadir)\"" >> $TMPH
echo "FFMPEG_CONFIGURATION=$FFMPEG_CONFIGURATION" >> config.mak
echo "prefix=$prefix" >> config.mak
@@ -2265,6 +2273,7 @@
echo "SHLIBDIR=\$(DESTDIR)$shlibdir" >> config.mak
echo "INCDIR=\$(DESTDIR)$incdir" >> config.mak
echo "BINDIR=\$(DESTDIR)$bindir" >> config.mak
+echo "DATADIR=\$(DESTDIR)$datadir" >> config.mak
echo "MANDIR=\$(DESTDIR)$mandir" >> config.mak
echo "CC=$cc" >> config.mak
echo "YASM=$yasmexe" >> config.mak
Index: Makefile
===================================================================
--- Makefile (revision 16790)
+++ Makefile (revision 16793)
@@ -24,6 +24,8 @@
FFLIBS := avdevice avformat avcodec avutil
+DATA_FILES := $(wildcard $(SRC_DIR)/ffpresets/*.ffpreset)
+
include common.mak
FF_LDFLAGS := $(FFLDFLAGS)
@@ -35,7 +37,7 @@
INSTALL_TARGETS-$(CONFIG_VHOOK) += install-vhook
ifneq ($(PROGS),)
-INSTALL_TARGETS-yes += install-progs
+INSTALL_TARGETS-yes += install-progs install-data
INSTALL_TARGETS-$(BUILD_DOC) += install-man
endif
INSTALL_PROGS_TARGETS-$(BUILD_SHARED) = install-libs
@@ -129,6 +131,10 @@
install -d "$(BINDIR)"
install -c -m 755 $(PROGS) "$(BINDIR)"
+install-data: $(DATA_FILES)
+ install -d "$(DATADIR)"
+ install -m 644 $(DATA_FILES) "$(DATADIR)"
+
install-man: $(MANPAGES)
install -d "$(MANDIR)/man1"
install -m 644 $(MANPAGES) "$(MANDIR)/man1"
@@ -137,11 +143,14 @@
install -d "$(SHLIBDIR)/vhook"
install -m 755 $(HOOKS) "$(SHLIBDIR)/vhook"
-uninstall: uninstall-progs uninstall-man uninstall-vhook
+uninstall: uninstall-progs uninstall-data uninstall-man uninstall-vhook
uninstall-progs:
rm -f $(addprefix "$(BINDIR)/", $(ALLPROGS))
+uninstall-data:
+ rm -rf "$(DATADIR)"
+
uninstall-man:
rm -f $(addprefix "$(MANDIR)/man1/",$(ALLMANPAGES))
ffmpeg-r17163-17311.patch:
--- NEW FILE ffmpeg-r17163-17311.patch ---
Index: ffmpeg.c
===================================================================
--- ffmpeg.c (revision 17162)
+++ ffmpeg.c (revision 17163)
@@ -555,12 +555,12 @@
ost->audio_resample = 1;
if (ost->audio_resample && !ost->resample) {
- if (dec->sample_fmt != SAMPLE_FMT_S16) {
- fprintf(stderr, "Audio resampler only works with 16 bits per sample, patch welcome.\n");
- av_exit(1);
- }
- ost->resample = audio_resample_init(enc->channels, dec->channels,
- enc->sample_rate, dec->sample_rate);
+ if (dec->sample_fmt != SAMPLE_FMT_S16)
+ fprintf(stderr, "Warning, using s16 intermediate sample format for resampling\n");
+ ost->resample = av_audio_resample_init(enc->channels, dec->channels,
+ enc->sample_rate, dec->sample_rate,
+ enc->sample_fmt, dec->sample_fmt,
+ 16, 10, 0, 0.8);
if (!ost->resample) {
fprintf(stderr, "Can not resample %d channels @ %d Hz to %d channels @ %d Hz\n",
dec->channels, dec->sample_rate,
@@ -570,7 +570,7 @@
}
#define MAKE_SFMT_PAIR(a,b) ((a)+SAMPLE_FMT_NB*(b))
- if (dec->sample_fmt!=enc->sample_fmt &&
+ if (!ost->audio_resample && dec->sample_fmt!=enc->sample_fmt &&
MAKE_SFMT_PAIR(enc->sample_fmt,dec->sample_fmt)!=ost->reformat_pair) {
if (!audio_out2)
audio_out2 = av_malloc(audio_out_size);
@@ -647,7 +647,7 @@
size_out = size;
}
- if (dec->sample_fmt!=enc->sample_fmt) {
+ if (!ost->audio_resample && dec->sample_fmt!=enc->sample_fmt) {
const void *ibuf[6]= {buftmp};
void *obuf[6]= {audio_out2};
int istride[6]= {isize};
Index: libavcodec/resample.c
===================================================================
--- libavcodec/resample.c (revision 17162)
+++ libavcodec/resample.c (revision 17163)
@@ -25,16 +25,32 @@
*/
#include "avcodec.h"
+#include "audioconvert.h"
+#include "opt.h"
struct AVResampleContext;
+static const char *context_to_name(void *ptr)
+{
+ return "audioresample";
+}
+
+static const AVOption options[] = {{NULL}};
+static const AVClass audioresample_context_class = { "ReSampleContext", context_to_name, options };
+
struct ReSampleContext {
+ const AVClass *av_class;
struct AVResampleContext *resample_context;
short *temp[2];
int temp_len;
float ratio;
/* channel convert */
int input_channels, output_channels, filter_channels;
+ AVAudioConvert *convert_ctx[2];
+ enum SampleFormat sample_fmt[2]; ///< input and output sample format
+ unsigned sample_size[2]; ///< size of one sample in sample_fmt
+ short *buffer[2]; ///< buffers used for conversion to S16
+ unsigned buffer_size[2]; ///< sizes of allocated buffers
};
/* n1: number of samples */
@@ -126,8 +142,12 @@
}
}
-ReSampleContext *audio_resample_init(int output_channels, int input_channels,
- int output_rate, int input_rate)
+ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
+ int output_rate, int input_rate,
+ enum SampleFormat sample_fmt_out,
+ enum SampleFormat sample_fmt_in,
+ int filter_length, int log2_phase_count,
+ int linear, double cutoff)
{
ReSampleContext *s;
@@ -153,6 +173,34 @@
if (s->output_channels < s->filter_channels)
s->filter_channels = s->output_channels;
+ s->sample_fmt [0] = sample_fmt_in;
+ s->sample_fmt [1] = sample_fmt_out;
+ s->sample_size[0] = av_get_bits_per_sample_format(s->sample_fmt[0])>>3;
+ s->sample_size[1] = av_get_bits_per_sample_format(s->sample_fmt[1])>>3;
+
+ if (s->sample_fmt[0] != SAMPLE_FMT_S16) {
+ if (!(s->convert_ctx[0] = av_audio_convert_alloc(SAMPLE_FMT_S16, 1,
+ s->sample_fmt[0], 1, NULL, 0))) {
+ av_log(s, AV_LOG_ERROR,
+ "Cannot convert %s sample format to s16 sample format\n",
+ avcodec_get_sample_fmt_name(s->sample_fmt[0]));
+ av_free(s);
+ return NULL;
+ }
+ }
+
+ if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
+ if (!(s->convert_ctx[1] = av_audio_convert_alloc(s->sample_fmt[1], 1,
+ SAMPLE_FMT_S16, 1, NULL, 0))) {
+ av_log(s, AV_LOG_ERROR,
+ "Cannot convert s16 sample format to %s sample format\n",
+ avcodec_get_sample_fmt_name(s->sample_fmt[1]));
+ av_audio_convert_free(s->convert_ctx[0]);
+ av_free(s);
+ return NULL;
+ }
+ }
+
/*
* AC-3 output is the only case where filter_channels could be greater than 2.
* input channels can't be greater than 2, so resample the 2 channels and then
@@ -162,11 +210,25 @@
s->filter_channels = 2;
#define TAPS 16
- s->resample_context= av_resample_init(output_rate, input_rate, TAPS, 10, 0, 0.8);
+ s->resample_context= av_resample_init(output_rate, input_rate,
+ filter_length, log2_phase_count, linear, cutoff);
+ s->av_class= &audioresample_context_class;
+
return s;
}
+#if LIBAVCODEC_VERSION_MAJOR < 53
+ReSampleContext *audio_resample_init(int output_channels, int input_channels,
+ int output_rate, int input_rate)
+{
+ return av_audio_resample_init(output_channels, input_channels,
+ output_rate, input_rate,
+ SAMPLE_FMT_S16, SAMPLE_FMT_S16,
+ TAPS, 10, 0, 0.8);
+}
+#endif
+
/* resample audio. 'nb_samples' is the number of input samples */
/* XXX: optimize it ! */
int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
@@ -175,6 +237,7 @@
short *bufin[2];
short *bufout[2];
short *buftmp2[2], *buftmp3[2];
+ short *output_bak = NULL;
int lenout;
if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) {
@@ -183,6 +246,52 @@
return nb_samples;
}
+ if (s->sample_fmt[0] != SAMPLE_FMT_S16) {
+ int istride[1] = { s->sample_size[0] };
+ int ostride[1] = { 2 };
+ const void *ibuf[1] = { input };
+ void *obuf[1];
+ unsigned input_size = nb_samples*s->input_channels*2;
+
+ if (!s->buffer_size[0] || s->buffer_size[0] < input_size) {
+ av_free(s->buffer[0]);
+ s->buffer_size[0] = input_size;
+ s->buffer[0] = av_malloc(s->buffer_size[0]);
+ if (!s->buffer[0]) {
+ av_log(s, AV_LOG_ERROR, "Could not allocate buffer\n");
+ return 0;
+ }
+ }
+
+ obuf[0] = s->buffer[0];
+
+ if (av_audio_convert(s->convert_ctx[0], obuf, ostride,
+ ibuf, istride, nb_samples*s->input_channels) < 0) {
+ av_log(s, AV_LOG_ERROR, "Audio sample format conversion failed\n");
+ return 0;
+ }
+
+ input = s->buffer[0];
+ }
+
+ lenout= 4*nb_samples * s->ratio + 16;
+
+ if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
+ output_bak = output;
+
+ if (!s->buffer_size[1] || s->buffer_size[1] < lenout) {
+ av_free(s->buffer[1]);
+ s->buffer_size[1] = lenout;
+ s->buffer[1] = av_malloc(s->buffer_size[1]);
+ if (!s->buffer[1]) {
+ av_log(s, AV_LOG_ERROR, "Could not allocate buffer\n");
+ return 0;
+ }
+ }
+
+ output = s->buffer[1];
+ }
+
/* XXX: move those malloc to resample init code */
for(i=0; i<s->filter_channels; i++){
bufin[i]= av_malloc( (nb_samples + s->temp_len) * sizeof(short) );
@@ -191,7 +300,6 @@
}
/* make some zoom to avoid round pb */
- lenout= 4*nb_samples * s->ratio + 16;
bufout[0]= av_malloc( lenout * sizeof(short) );
bufout[1]= av_malloc( lenout * sizeof(short) );
@@ -233,6 +341,19 @@
ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
}
+ if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
+ int istride[1] = { 2 };
+ int ostride[1] = { s->sample_size[1] };
+ const void *ibuf[1] = { output };
+ void *obuf[1] = { output_bak };
+
+ if (av_audio_convert(s->convert_ctx[1], obuf, ostride,
+ ibuf, istride, nb_samples1*s->output_channels) < 0) {
+ av_log(s, AV_LOG_ERROR, "Audio sample format convertion failed\n");
+ return 0;
+ }
+ }
+
for(i=0; i<s->filter_channels; i++)
av_free(bufin[i]);
@@ -246,5 +367,9 @@
av_resample_close(s->resample_context);
av_freep(&s->temp[0]);
av_freep(&s->temp[1]);
+ av_freep(&s->buffer[0]);
+ av_freep(&s->buffer[1]);
+ av_audio_convert_free(s->convert_ctx[0]);
+ av_audio_convert_free(s->convert_ctx[1]);
av_free(s);
}
Index: libavcodec/avcodec.h
===================================================================
--- libavcodec/avcodec.h (revision 17162)
+++ libavcodec/avcodec.h (revision 17163)
@@ -2443,8 +2443,36 @@
typedef struct ReSampleContext ReSampleContext;
-ReSampleContext *audio_resample_init(int output_channels, int input_channels,
- int output_rate, int input_rate);
+#if LIBAVCODEC_VERSION_MAJOR < 53
+/**
+ * @deprecated Use av_audio_resample_init() instead.
+ */
+attribute_deprecated ReSampleContext *audio_resample_init(int output_channels, int input_channels,
+ int output_rate, int input_rate);
+#endif
+/**
+ * Initializes audio resampling context
+ *
+ * @param output_channels number of output channels
+ * @param input_channels number of input channels
+ * @param output_rate output sample rate
+ * @param input_rate input sample rate
+ * @param sample_fmt_out requested output sample format
+ * @param sample_fmt_in input sample format
+ * @param filter_length length of each FIR filter in the filterbank relative to the cutoff freq
+ * @param log2_phase_count log2 of the number of entries in the polyphase filterbank
+ * @param linear If 1 then the used FIR filter will be linearly interpolated
+ between the 2 closest, if 0 the closest will be used
+ * @param cutoff cutoff frequency, 1.0 corresponds to half the output sampling rate
+ * @return allocated ReSampleContext, NULL if error occured
+ */
+ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
+ int output_rate, int input_rate,
+ enum SampleFormat sample_fmt_out,
+ enum SampleFormat sample_fmt_in,
+ int filter_length, int log2_phase_count,
+ int linear, double cutoff);
+
int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples);
void audio_resample_close(ReSampleContext *s);
Index: ffmpeg-snapshot.sh
===================================================================
RCS file: /cvs/free/rpms/ffmpeg/F-10/ffmpeg-snapshot.sh,v
retrieving revision 1.2
retrieving revision 1.3
diff -u -r1.2 -r1.3
--- ffmpeg-snapshot.sh 5 Sep 2008 15:17:52 -0000 1.2
+++ ffmpeg-snapshot.sh 26 Oct 2009 00:39:28 -0000 1.3
@@ -13,10 +13,13 @@
unset CDPATH
pwd=$(pwd)
svn=$(date +%Y%m%d)
+svn=20080908
cd "$tmp"
svn checkout -r {$svn} svn://svn.mplayerhq.hu/ffmpeg/trunk ffmpeg-$svn
-cd ffmpeg-$svn
+cd ffmpeg-$svn/libswscale
+svn update -r {$svn}
+cd ..
./version.sh . version.h
find . -type d -name .svn -print0 | xargs -0r rm -rf
sed -i -e '/^\.PHONY: version\.h$/d' Makefile
Index: ffmpeg.spec
===================================================================
RCS file: /cvs/free/rpms/ffmpeg/F-10/ffmpeg.spec,v
retrieving revision 1.11
retrieving revision 1.12
diff -u -r1.11 -r1.12
--- ffmpeg.spec 8 Mar 2009 23:02:44 -0000 1.11
+++ ffmpeg.spec 26 Oct 2009 00:39:28 -0000 1.12
@@ -6,7 +6,7 @@
Summary: Digital VCR and streaming server
Name: ffmpeg
Version: 0.4.9
-Release: 0.55.%{svn}%{?dist}
+Release: 0.56.%{svn}%{?dist}
License: GPLv2+
Group: Applications/Multimedia
URL: http://ffmpeg.org/
@@ -25,8 +25,15 @@
Patch11: %{name}-r16846.patch
# backport av_find_nearest_q_idx for dvdstyler
Patch12: %{name}-r15415.patch
+# backport presets
+Patch13: %{name}-r15348.patch
+# backport audio resampling fix
+Patch14: %{name}-r17163-17311.patch
+# backport presets installation
+Patch15: %{name}-r16791-16793.patch
BuildRoot: %{_tmppath}/%{name}-%{version}-%{release}-root-%(%{__id_u} -n)
+Requires: %{name}-libs = %{version}-%{release}
%{?_with_amr:BuildRequires: amrnb-devel amrwb-devel}
BuildRequires: bzip2-devel
BuildRequires: faac-devel
@@ -84,6 +91,8 @@
%define ff_configure \
../configure \\\
--prefix=%{_prefix} \\\
+ --bindir=%{_bindir} \\\
+ --datadir=%{_datadir}/ffmpeg \\\
--incdir=%{_includedir}/ffmpeg \\\
--libdir=%{_libdir} \\\
--mandir=%{_mandir} \\\
@@ -123,6 +132,9 @@
%patch10 -p1
%patch11 -p1
%patch12 -p1
+%patch13 -p0
+%patch14 -p0
+%patch15 -p0
%build
mkdir generic
@@ -220,10 +232,10 @@
%files
%defattr(-,root,root,-)
%doc COPYING.GPL CREDITS Changelog README __doc/*.*
-# Note: as of 20070204, "configure" doesn't have --bindir.
-%{_prefix}/bin/ffmpeg
-%{_prefix}/bin/ffplay
-%{_prefix}/bin/ffserver
+%{_bindir}/ffmpeg
+%{_bindir}/ffplay
+%{_bindir}/ffserver
+%{_datadir}/ffmpeg
%{_mandir}/man1/ffmpeg.1*
%{_mandir}/man1/ffplay.1*
%{_mandir}/man1/ffserver.1*
@@ -262,6 +274,13 @@
%changelog
+* Mon Oct 26 2009 Dominik Mierzejewski <rpm at greysector.net> - 0.4.9-0.56.20080908
+- backport audio resampler fixes (bug #426)
+- add a requirement on exact EVR between main package and -libs
+ so that "yum update ffmpeg" works as expected
+- install presets (backport)
+- use bindir instead of prefix/bin
+
* Sun Mar 08 2009 Dominik Mierzejewski <rpm at greysector.net> - 0.4.9-0.55.20080908
- backport support for Dirac in Matroska
- add comments for all patches
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