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