Merge remote-tracking branch 'newdev/master'
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
commit
d4a50a2100
27
configure
vendored
27
configure
vendored
@ -960,6 +960,7 @@ CONFIG_LIST="
|
||||
rtpdec
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runtime_cpudetect
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shared
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sinewin
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small
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sram
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static
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@ -1238,8 +1239,8 @@ mdct_select="fft"
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rdft_select="fft"
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# decoders / encoders / hardware accelerators
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aac_decoder_select="mdct rdft"
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aac_encoder_select="mdct"
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aac_decoder_select="mdct rdft sinewin"
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aac_encoder_select="mdct sinewin"
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aac_latm_decoder_select="aac_decoder aac_latm_parser"
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ac3_decoder_select="mdct ac3_parser"
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ac3_encoder_select="mdct ac3dsp"
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@ -1247,12 +1248,12 @@ ac3_fixed_encoder_select="ac3dsp"
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alac_encoder_select="lpc"
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amrnb_decoder_select="lsp"
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amrwb_decoder_select="lsp"
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atrac1_decoder_select="mdct"
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atrac1_decoder_select="mdct sinewin"
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atrac3_decoder_select="mdct"
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binkaudio_dct_decoder_select="mdct rdft dct"
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binkaudio_rdft_decoder_select="mdct rdft"
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cavs_decoder_select="golomb"
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cook_decoder_select="mdct"
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cook_decoder_select="mdct sinewin"
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cscd_decoder_suggest="zlib"
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dca_decoder_select="mdct"
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dnxhd_encoder_select="aandct"
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@ -1315,8 +1316,8 @@ msmpeg4v2_decoder_select="h263_decoder"
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msmpeg4v2_encoder_select="h263_encoder"
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msmpeg4v3_decoder_select="h263_decoder"
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msmpeg4v3_encoder_select="h263_encoder"
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nellymoser_decoder_select="mdct"
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nellymoser_encoder_select="mdct"
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nellymoser_decoder_select="mdct sinewin"
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nellymoser_encoder_select="mdct sinewin"
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png_decoder_select="zlib"
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png_encoder_select="zlib"
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qcelp_decoder_select="lsp"
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@ -1343,7 +1344,7 @@ tiff_decoder_suggest="zlib"
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tiff_encoder_suggest="zlib"
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truehd_decoder_select="mlp_decoder"
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tscc_decoder_select="zlib"
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twinvq_decoder_select="mdct lsp"
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twinvq_decoder_select="mdct lsp sinewin"
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vc1_decoder_select="h263_decoder"
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vc1_crystalhd_decoder_select="crystalhd"
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vc1_dxva2_hwaccel_deps="dxva2api_h DXVA_PictureParameters_wDecodedPictureIndex"
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@ -1356,12 +1357,12 @@ vp6_decoder_select="huffman"
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vp6a_decoder_select="vp6_decoder"
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vp6f_decoder_select="vp6_decoder"
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vp8_decoder_select="h264pred"
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wmapro_decoder_select="mdct"
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wmav1_decoder_select="mdct"
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||||
wmav1_encoder_select="mdct"
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wmav2_decoder_select="mdct"
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wmav2_encoder_select="mdct"
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wmavoice_decoder_select="lsp rdft dct mdct"
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wmapro_decoder_select="mdct sinewin"
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wmav1_decoder_select="mdct sinewin"
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||||
wmav1_encoder_select="mdct sinewin"
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wmav2_decoder_select="mdct sinewin"
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||||
wmav2_encoder_select="mdct sinewin"
|
||||
wmavoice_decoder_select="lsp rdft dct mdct sinewin"
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||||
wmv1_decoder_select="h263_decoder"
|
||||
wmv1_encoder_select="h263_encoder"
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wmv2_decoder_select="h263_decoder"
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|
@ -622,11 +622,43 @@ Synchronize read on input.
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||||
@section Advanced options
|
||||
|
||||
@table @option
|
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@item -map @var{input_stream_id}[:@var{sync_stream_id}]
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||||
Set stream mapping from input streams to output streams.
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Just enumerate the input streams in the order you want them in the output.
|
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@var{sync_stream_id} if specified sets the input stream to sync
|
||||
against.
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||||
@item -map @var{input_file_id}.@var{input_stream_id}[:@var{sync_file_id}.@var{sync_stream_id}]
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||||
|
||||
Designate an input stream as a source for the output file. Each input
|
||||
stream is identified by the input file index @var{input_file_id} and
|
||||
the input stream index @var{input_stream_id} within the input
|
||||
file. Both indexes start at 0. If specified,
|
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@var{sync_file_id}.@var{sync_stream_id} sets which input stream
|
||||
is used as a presentation sync reference.
|
||||
|
||||
The @code{-map} options must be specified just after the output file.
|
||||
If any @code{-map} options are used, the number of @code{-map} options
|
||||
on the command line must match the number of streams in the output
|
||||
file. The first @code{-map} option on the command line specifies the
|
||||
source for output stream 0, the second @code{-map} option specifies
|
||||
the source for output stream 1, etc.
|
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|
||||
For example, if you have two audio streams in the first input file,
|
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these streams are identified by "0.0" and "0.1". You can use
|
||||
@code{-map} to select which stream to place in an output file. For
|
||||
example:
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@example
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||||
ffmpeg -i INPUT out.wav -map 0.1
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||||
@end example
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||||
will map the input stream in @file{INPUT} identified by "0.1" to
|
||||
the (single) output stream in @file{out.wav}.
|
||||
|
||||
For example, to select the stream with index 2 from input file
|
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@file{a.mov} (specified by the identifier "0.2"), and stream with
|
||||
index 6 from input @file{b.mov} (specified by the identifier "1.6"),
|
||||
and copy them to the output file @file{out.mov}:
|
||||
@example
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||||
ffmpeg -i a.mov -i b.mov -vcodec copy -acodec copy out.mov -map 0.2 -map 1.6
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@end example
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|
||||
To add more streams to the output file, you can use the
|
||||
@code{-newaudio}, @code{-newvideo}, @code{-newsubtitle} options.
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|
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@item -map_meta_data @var{outfile}[,@var{metadata}]:@var{infile}[,@var{metadata}]
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Deprecated, use @var{-map_metadata} instead.
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|
2
ffmpeg.c
2
ffmpeg.c
@ -4214,7 +4214,7 @@ static const OptionDef options[] = {
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{ "f", HAS_ARG, {(void*)opt_format}, "force format", "fmt" },
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{ "i", HAS_ARG, {(void*)opt_input_file}, "input file name", "filename" },
|
||||
{ "y", OPT_BOOL, {(void*)&file_overwrite}, "overwrite output files" },
|
||||
{ "map", HAS_ARG | OPT_EXPERT, {(void*)opt_map}, "set input stream mapping", "file:stream[:syncfile:syncstream]" },
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{ "map", HAS_ARG | OPT_EXPERT, {(void*)opt_map}, "set input stream mapping", "file.stream[:syncfile.syncstream]" },
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{ "map_meta_data", HAS_ARG | OPT_EXPERT, {(void*)opt_map_meta_data}, "DEPRECATED set meta data information of outfile from infile",
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"outfile[,metadata]:infile[,metadata]" },
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{ "map_metadata", HAS_ARG | OPT_EXPERT, {(void*)opt_map_metadata}, "set metadata information of outfile from infile",
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|
@ -43,6 +43,7 @@ OBJS-$(CONFIG_LSP) += lsp.o
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||||
OBJS-$(CONFIG_MDCT) += mdct.o
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RDFT-OBJS-$(CONFIG_HARDCODED_TABLES) += sin_tables.o
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OBJS-$(CONFIG_RDFT) += rdft.o $(RDFT-OBJS-yes)
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OBJS-$(CONFIG_SINEWIN) += sinewin.o
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OBJS-$(CONFIG_VAAPI) += vaapi.o
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OBJS-$(CONFIG_VDPAU) += vdpau.o
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|
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@ -50,14 +51,14 @@ OBJS-$(CONFIG_VDPAU) += vdpau.o
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OBJS-$(CONFIG_A64MULTI_ENCODER) += a64multienc.o elbg.o
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OBJS-$(CONFIG_A64MULTI5_ENCODER) += a64multienc.o elbg.o
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OBJS-$(CONFIG_AAC_DECODER) += aacdec.o aactab.o aacsbr.o aacps.o \
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aacadtsdec.o mpeg4audio.o
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aacadtsdec.o mpeg4audio.o kbdwin.o
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OBJS-$(CONFIG_AAC_ENCODER) += aacenc.o aaccoder.o \
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aacpsy.o aactab.o \
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psymodel.o iirfilter.o \
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mpeg4audio.o
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mpeg4audio.o kbdwin.o
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OBJS-$(CONFIG_AASC_DECODER) += aasc.o msrledec.o
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OBJS-$(CONFIG_AC3_DECODER) += ac3dec.o ac3dec_data.o ac3.o
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OBJS-$(CONFIG_AC3_ENCODER) += ac3enc_float.o ac3tab.o ac3.o
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OBJS-$(CONFIG_AC3_ENCODER) += ac3enc_float.o ac3tab.o ac3.o kbdwin.o
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OBJS-$(CONFIG_AC3_FIXED_ENCODER) += ac3enc_fixed.o ac3tab.o ac3.o
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OBJS-$(CONFIG_ALAC_DECODER) += alac.o
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OBJS-$(CONFIG_ALAC_ENCODER) += alacenc.o
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@ -694,7 +695,7 @@ $(SUBDIR)%_tablegen$(HOSTEXESUF): $(SUBDIR)%_tablegen.c $(SUBDIR)%_tablegen.h $(
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$(HOSTCC) $(HOSTCFLAGS) $(HOSTLDFLAGS) -o $@ $(filter %.c,$^) $(HOSTLIBS)
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|
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GEN_HEADERS = cbrt_tables.h aacps_tables.h aac_tables.h dv_tables.h \
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mdct_tables.h mpegaudio_tables.h motionpixels_tables.h \
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sinewin_tables.h mpegaudio_tables.h motionpixels_tables.h \
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pcm_tables.h qdm2_tables.h
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GEN_HEADERS := $(addprefix $(SUBDIR), $(GEN_HEADERS))
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@ -706,7 +707,7 @@ $(SUBDIR)aacdec.o: $(SUBDIR)cbrt_tables.h
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$(SUBDIR)aacps.o: $(SUBDIR)aacps_tables.h
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$(SUBDIR)aactab.o: $(SUBDIR)aac_tables.h
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$(SUBDIR)dv.o: $(SUBDIR)dv_tables.h
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$(SUBDIR)mdct.o: $(SUBDIR)mdct_tables.h
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$(SUBDIR)sinewin.o: $(SUBDIR)sinewin_tables.h
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$(SUBDIR)mpegaudiodec.o: $(SUBDIR)mpegaudio_tables.h
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$(SUBDIR)mpegaudiodec_float.o: $(SUBDIR)mpegaudio_tables.h
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$(SUBDIR)motionpixels.o: $(SUBDIR)motionpixels_tables.h
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|
@ -87,6 +87,8 @@
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#include "fft.h"
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#include "fmtconvert.h"
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#include "lpc.h"
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#include "kbdwin.h"
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#include "sinewin.h"
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#include "aac.h"
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#include "aactab.h"
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@ -1750,7 +1752,7 @@ static void windowing_and_mdct_ltp(AACContext *ac, float *out,
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ac->dsp.vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
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memset(in + 1024 + 576, 0, 448 * sizeof(float));
|
||||
}
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ff_mdct_calc(&ac->mdct_ltp, out, in);
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ac->mdct_ltp.mdct_calc(&ac->mdct_ltp, out, in);
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}
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|
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/**
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@ -1839,9 +1841,9 @@ static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
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// imdct
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if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
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for (i = 0; i < 1024; i += 128)
|
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ff_imdct_half(&ac->mdct_small, buf + i, in + i);
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ac->mdct_small.imdct_half(&ac->mdct_small, buf + i, in + i);
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} else
|
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ff_imdct_half(&ac->mdct, buf, in);
|
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ac->mdct.imdct_half(&ac->mdct, buf, in);
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||||
|
||||
/* window overlapping
|
||||
* NOTE: To simplify the overlapping code, all 'meaningless' short to long
|
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|
@ -34,6 +34,8 @@
|
||||
#include "put_bits.h"
|
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#include "dsputil.h"
|
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#include "mpeg4audio.h"
|
||||
#include "kbdwin.h"
|
||||
#include "sinewin.h"
|
||||
|
||||
#include "aac.h"
|
||||
#include "aactab.h"
|
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@ -250,7 +252,7 @@ static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s,
|
||||
for (i = 0; i < 1024; i++)
|
||||
sce->saved[i] = audio[i * chans];
|
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}
|
||||
ff_mdct_calc(&s->mdct1024, sce->coeffs, output);
|
||||
s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);
|
||||
} else {
|
||||
for (k = 0; k < 1024; k += 128) {
|
||||
for (i = 448 + k; i < 448 + k + 256; i++)
|
||||
@ -259,7 +261,7 @@ static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s,
|
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: audio[(i-1024)*chans];
|
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s->dsp.vector_fmul (output, output, k ? swindow : pwindow, 128);
|
||||
s->dsp.vector_fmul_reverse(output+128, output+128, swindow, 128);
|
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ff_mdct_calc(&s->mdct128, sce->coeffs + k, output);
|
||||
s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + k, output);
|
||||
}
|
||||
for (i = 0; i < 1024; i++)
|
||||
sce->saved[i] = audio[i * chans];
|
||||
|
@ -1155,7 +1155,7 @@ static void sbr_qmf_analysis(DSPContext *dsp, FFTContext *mdct, const float *in,
|
||||
}
|
||||
z[64+63] = z[32];
|
||||
|
||||
ff_imdct_half(mdct, z, z+64);
|
||||
mdct->imdct_half(mdct, z, z+64);
|
||||
for (k = 0; k < 32; k++) {
|
||||
W[1][i][k][0] = -z[63-k];
|
||||
W[1][i][k][1] = z[k];
|
||||
@ -1190,7 +1190,7 @@ static void sbr_qmf_synthesis(DSPContext *dsp, FFTContext *mdct,
|
||||
X[0][i][ n] = -X[0][i][n];
|
||||
X[0][i][32+n] = X[1][i][31-n];
|
||||
}
|
||||
ff_imdct_half(mdct, mdct_buf[0], X[0][i]);
|
||||
mdct->imdct_half(mdct, mdct_buf[0], X[0][i]);
|
||||
for (n = 0; n < 32; n++) {
|
||||
v[ n] = mdct_buf[0][63 - 2*n];
|
||||
v[63 - n] = -mdct_buf[0][62 - 2*n];
|
||||
@ -1199,8 +1199,8 @@ static void sbr_qmf_synthesis(DSPContext *dsp, FFTContext *mdct,
|
||||
for (n = 1; n < 64; n+=2) {
|
||||
X[1][i][n] = -X[1][i][n];
|
||||
}
|
||||
ff_imdct_half(mdct, mdct_buf[0], X[0][i]);
|
||||
ff_imdct_half(mdct, mdct_buf[1], X[1][i]);
|
||||
mdct->imdct_half(mdct, mdct_buf[0], X[0][i]);
|
||||
mdct->imdct_half(mdct, mdct_buf[1], X[1][i]);
|
||||
for (n = 0; n < 64; n++) {
|
||||
v[ n] = -mdct_buf[0][63 - n] + mdct_buf[1][ n ];
|
||||
v[127 - n] = mdct_buf[0][63 - n] + mdct_buf[1][ n ];
|
||||
|
@ -35,6 +35,7 @@
|
||||
#include "ac3_parser.h"
|
||||
#include "ac3dec.h"
|
||||
#include "ac3dec_data.h"
|
||||
#include "kbdwin.h"
|
||||
|
||||
/** Large enough for maximum possible frame size when the specification limit is ignored */
|
||||
#define AC3_FRAME_BUFFER_SIZE 32768
|
||||
@ -621,13 +622,13 @@ static inline void do_imdct(AC3DecodeContext *s, int channels)
|
||||
float *x = s->tmp_output+128;
|
||||
for(i=0; i<128; i++)
|
||||
x[i] = s->transform_coeffs[ch][2*i];
|
||||
ff_imdct_half(&s->imdct_256, s->tmp_output, x);
|
||||
s->imdct_256.imdct_half(&s->imdct_256, s->tmp_output, x);
|
||||
s->dsp.vector_fmul_window(s->output[ch-1], s->delay[ch-1], s->tmp_output, s->window, 128);
|
||||
for(i=0; i<128; i++)
|
||||
x[i] = s->transform_coeffs[ch][2*i+1];
|
||||
ff_imdct_half(&s->imdct_256, s->delay[ch-1], x);
|
||||
s->imdct_256.imdct_half(&s->imdct_256, s->delay[ch-1], x);
|
||||
} else {
|
||||
ff_imdct_half(&s->imdct_512, s->tmp_output, s->transform_coeffs[ch]);
|
||||
s->imdct_512.imdct_half(&s->imdct_512, s->tmp_output, s->transform_coeffs[ch]);
|
||||
s->dsp.vector_fmul_window(s->output[ch-1], s->delay[ch-1], s->tmp_output, s->window, 128);
|
||||
memcpy(s->delay[ch-1], s->tmp_output+128, 128*sizeof(float));
|
||||
}
|
||||
|
@ -28,6 +28,7 @@
|
||||
|
||||
#define CONFIG_AC3ENC_FLOAT 1
|
||||
#include "ac3enc.c"
|
||||
#include "kbdwin.h"
|
||||
|
||||
|
||||
/**
|
||||
@ -74,7 +75,7 @@ static av_cold int mdct_init(AVCodecContext *avctx, AC3MDCTContext *mdct,
|
||||
*/
|
||||
static void mdct512(AC3MDCTContext *mdct, float *out, float *in)
|
||||
{
|
||||
ff_mdct_calc(&mdct->fft, out, in);
|
||||
mdct->fft.mdct_calc(&mdct->fft, out, in);
|
||||
}
|
||||
|
||||
|
||||
|
@ -19,6 +19,7 @@
|
||||
*/
|
||||
|
||||
#include "libavcodec/fft.h"
|
||||
#include "libavcodec/rdft.h"
|
||||
#include "libavcodec/synth_filter.h"
|
||||
|
||||
void ff_fft_permute_neon(FFTContext *s, FFTComplex *z);
|
||||
|
@ -36,6 +36,7 @@
|
||||
#include "get_bits.h"
|
||||
#include "dsputil.h"
|
||||
#include "fft.h"
|
||||
#include "sinewin.h"
|
||||
|
||||
#include "atrac.h"
|
||||
#include "atrac1data.h"
|
||||
@ -99,7 +100,7 @@ static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits,
|
||||
for (i = 0; i < transf_size / 2; i++)
|
||||
FFSWAP(float, spec[i], spec[transf_size - 1 - i]);
|
||||
}
|
||||
ff_imdct_half(mdct_context, out, spec);
|
||||
mdct_context->imdct_half(mdct_context, out, spec);
|
||||
}
|
||||
|
||||
|
||||
|
@ -146,7 +146,7 @@ static void IMLT(ATRAC3Context *q, float *pInput, float *pOutput, int odd_band)
|
||||
/**
|
||||
* Reverse the odd bands before IMDCT, this is an effect of the QMF transform
|
||||
* or it gives better compression to do it this way.
|
||||
* FIXME: It should be possible to handle this in ff_imdct_calc
|
||||
* FIXME: It should be possible to handle this in imdct_calc
|
||||
* for that to happen a modification of the prerotation step of
|
||||
* all SIMD code and C code is needed.
|
||||
* Or fix the functions before so they generate a pre reversed spectrum.
|
||||
@ -156,7 +156,7 @@ static void IMLT(ATRAC3Context *q, float *pInput, float *pOutput, int odd_band)
|
||||
FFSWAP(float, pInput[i], pInput[255-i]);
|
||||
}
|
||||
|
||||
ff_imdct_calc(&q->mdct_ctx,pOutput,pInput);
|
||||
q->mdct_ctx.imdct_calc(&q->mdct_ctx,pOutput,pInput);
|
||||
|
||||
/* Perform windowing on the output. */
|
||||
dsp.vector_fmul(pOutput, pOutput, mdct_window, 512);
|
||||
|
@ -19,6 +19,8 @@
|
||||
#include "libavutil/mem.h"
|
||||
#include "avfft.h"
|
||||
#include "fft.h"
|
||||
#include "rdft.h"
|
||||
#include "dct.h"
|
||||
|
||||
/* FFT */
|
||||
|
||||
@ -101,7 +103,7 @@ RDFTContext *av_rdft_init(int nbits, enum RDFTransformType trans)
|
||||
|
||||
void av_rdft_calc(RDFTContext *s, FFTSample *data)
|
||||
{
|
||||
ff_rdft_calc(s, data);
|
||||
s->rdft_calc(s, data);
|
||||
}
|
||||
|
||||
void av_rdft_end(RDFTContext *s)
|
||||
@ -128,7 +130,7 @@ DCTContext *av_dct_init(int nbits, enum DCTTransformType inverse)
|
||||
|
||||
void av_dct_calc(DCTContext *s, FFTSample *data)
|
||||
{
|
||||
ff_dct_calc(s, data);
|
||||
s->dct_calc(s, data);
|
||||
}
|
||||
|
||||
void av_dct_end(DCTContext *s)
|
||||
|
@ -32,7 +32,8 @@
|
||||
#define ALT_BITSTREAM_READER_LE
|
||||
#include "get_bits.h"
|
||||
#include "dsputil.h"
|
||||
#include "fft.h"
|
||||
#include "dct.h"
|
||||
#include "rdft.h"
|
||||
#include "fmtconvert.h"
|
||||
#include "libavutil/intfloat_readwrite.h"
|
||||
|
||||
@ -223,11 +224,11 @@ static void decode_block(BinkAudioContext *s, short *out, int use_dct)
|
||||
|
||||
if (CONFIG_BINKAUDIO_DCT_DECODER && use_dct) {
|
||||
coeffs[0] /= 0.5;
|
||||
ff_dct_calc (&s->trans.dct, coeffs);
|
||||
s->trans.dct.dct_calc(&s->trans.dct, coeffs);
|
||||
s->dsp.vector_fmul_scalar(coeffs, coeffs, s->frame_len / 2, s->frame_len);
|
||||
}
|
||||
else if (CONFIG_BINKAUDIO_RDFT_DECODER)
|
||||
ff_rdft_calc(&s->trans.rdft, coeffs);
|
||||
s->trans.rdft.rdft_calc(&s->trans.rdft, coeffs);
|
||||
}
|
||||
|
||||
s->fmt_conv.float_to_int16_interleave(out, (const float **)s->coeffs_ptr,
|
||||
|
@ -54,6 +54,7 @@
|
||||
#include "bytestream.h"
|
||||
#include "fft.h"
|
||||
#include "libavutil/audioconvert.h"
|
||||
#include "sinewin.h"
|
||||
|
||||
#include "cookdata.h"
|
||||
|
||||
@ -753,7 +754,7 @@ static void imlt_gain(COOKContext *q, float *inbuffer,
|
||||
int i;
|
||||
|
||||
/* Inverse modified discrete cosine transform */
|
||||
ff_imdct_calc(&q->mdct_ctx, q->mono_mdct_output, inbuffer);
|
||||
q->mdct_ctx.imdct_calc(&q->mdct_ctx, q->mono_mdct_output, inbuffer);
|
||||
|
||||
q->imlt_window (q, buffer1, gains_ptr, previous_buffer);
|
||||
|
||||
|
@ -37,7 +37,7 @@ int main(int argc, char *argv[])
|
||||
double (*func)(double) = do_sin ? sin : cos;
|
||||
|
||||
printf("/* This file was generated by libavcodec/costablegen */\n");
|
||||
printf("#include \"libavcodec/fft.h\"\n");
|
||||
printf("#include \"libavcodec/%s\"\n", do_sin ? "rdft.h" : "fft.h");
|
||||
for (i = 4; i <= BITS; i++) {
|
||||
int m = 1 << i;
|
||||
double freq = 2*M_PI/m;
|
||||
|
@ -29,8 +29,7 @@
|
||||
|
||||
#include <math.h>
|
||||
#include "libavutil/mathematics.h"
|
||||
#include "fft.h"
|
||||
#include "x86/fft.h"
|
||||
#include "dct.h"
|
||||
|
||||
#define DCT32_FLOAT
|
||||
#include "dct32.c"
|
||||
@ -59,7 +58,7 @@ static void ff_dst_calc_I_c(DCTContext *ctx, FFTSample *data)
|
||||
}
|
||||
|
||||
data[n/2] *= 2;
|
||||
ff_rdft_calc(&ctx->rdft, data);
|
||||
ctx->rdft.rdft_calc(&ctx->rdft, data);
|
||||
|
||||
data[0] *= 0.5f;
|
||||
|
||||
@ -93,7 +92,7 @@ static void ff_dct_calc_I_c(DCTContext *ctx, FFTSample *data)
|
||||
data[n - i] = tmp1 + s;
|
||||
}
|
||||
|
||||
ff_rdft_calc(&ctx->rdft, data);
|
||||
ctx->rdft.rdft_calc(&ctx->rdft, data);
|
||||
data[n] = data[1];
|
||||
data[1] = next;
|
||||
|
||||
@ -121,7 +120,7 @@ static void ff_dct_calc_III_c(DCTContext *ctx, FFTSample *data)
|
||||
|
||||
data[1] = 2 * next;
|
||||
|
||||
ff_rdft_calc(&ctx->rdft, data);
|
||||
ctx->rdft.rdft_calc(&ctx->rdft, data);
|
||||
|
||||
for (i = 0; i < n / 2; i++) {
|
||||
float tmp1 = data[i ] * inv_n;
|
||||
@ -152,7 +151,7 @@ static void ff_dct_calc_II_c(DCTContext *ctx, FFTSample *data)
|
||||
data[n-i-1] = tmp1 - s;
|
||||
}
|
||||
|
||||
ff_rdft_calc(&ctx->rdft, data);
|
||||
ctx->rdft.rdft_calc(&ctx->rdft, data);
|
||||
|
||||
next = data[1] * 0.5;
|
||||
data[1] *= -1;
|
||||
@ -176,11 +175,6 @@ static void dct32_func(DCTContext *ctx, FFTSample *data)
|
||||
ctx->dct32(data, data);
|
||||
}
|
||||
|
||||
void ff_dct_calc(DCTContext *s, FFTSample *data)
|
||||
{
|
||||
s->dct_calc(s, data);
|
||||
}
|
||||
|
||||
av_cold int ff_dct_init(DCTContext *s, int nbits, enum DCTTransformType inverse)
|
||||
{
|
||||
int n = 1 << nbits;
|
||||
|
50
libavcodec/dct.h
Normal file
50
libavcodec/dct.h
Normal file
@ -0,0 +1,50 @@
|
||||
/*
|
||||
* (I)DCT Transforms
|
||||
* Copyright (c) 2009 Peter Ross <pross@xvid.org>
|
||||
* Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
|
||||
* Copyright (c) 2010 Vitor Sessak
|
||||
*
|
||||
* This file is part of FFmpeg.
|
||||
*
|
||||
* FFmpeg is free software; you can redistribute it and/or
|
||||
* modify it under the terms of the GNU Lesser General Public
|
||||
* License as published by the Free Software Foundation; either
|
||||
* version 2.1 of the License, or (at your option) any later version.
|
||||
*
|
||||
* FFmpeg 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
|
||||
* Lesser General Public License for more details.
|
||||
*
|
||||
* You should have received a copy of the GNU Lesser General Public
|
||||
* License along with FFmpeg; if not, write to the Free Software
|
||||
* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
|
||||
*/
|
||||
|
||||
#ifndef AVCODEC_DCT_H
|
||||
#define AVCODEC_DCT_H
|
||||
|
||||
#include "rdft.h"
|
||||
|
||||
struct DCTContext {
|
||||
int nbits;
|
||||
int inverse;
|
||||
RDFTContext rdft;
|
||||
const float *costab;
|
||||
FFTSample *csc2;
|
||||
void (*dct_calc)(struct DCTContext *s, FFTSample *data);
|
||||
void (*dct32)(FFTSample *out, const FFTSample *in);
|
||||
};
|
||||
|
||||
/**
|
||||
* Set up DCT.
|
||||
* @param nbits size of the input array:
|
||||
* (1 << nbits) for DCT-II, DCT-III and DST-I
|
||||
* (1 << nbits) + 1 for DCT-I
|
||||
*
|
||||
* @note the first element of the input of DST-I is ignored
|
||||
*/
|
||||
int ff_dct_init(DCTContext *s, int nbits, enum DCTTransformType type);
|
||||
void ff_dct_end (DCTContext *s);
|
||||
|
||||
#endif
|
@ -27,6 +27,8 @@
|
||||
#include "libavutil/lfg.h"
|
||||
#include "libavutil/log.h"
|
||||
#include "fft.h"
|
||||
#include "dct.h"
|
||||
#include "rdft.h"
|
||||
#include <math.h>
|
||||
#include <unistd.h>
|
||||
#include <sys/time.h>
|
||||
@ -327,20 +329,20 @@ int main(int argc, char **argv)
|
||||
case TRANSFORM_MDCT:
|
||||
if (do_inverse) {
|
||||
imdct_ref((float *)tab_ref, (float *)tab1, fft_nbits);
|
||||
ff_imdct_calc(m, tab2, (float *)tab1);
|
||||
m->imdct_calc(m, tab2, (float *)tab1);
|
||||
err = check_diff((float *)tab_ref, tab2, fft_size, scale);
|
||||
} else {
|
||||
mdct_ref((float *)tab_ref, (float *)tab1, fft_nbits);
|
||||
|
||||
ff_mdct_calc(m, tab2, (float *)tab1);
|
||||
m->mdct_calc(m, tab2, (float *)tab1);
|
||||
|
||||
err = check_diff((float *)tab_ref, tab2, fft_size / 2, scale);
|
||||
}
|
||||
break;
|
||||
case TRANSFORM_FFT:
|
||||
memcpy(tab, tab1, fft_size * sizeof(FFTComplex));
|
||||
ff_fft_permute(s, tab);
|
||||
ff_fft_calc(s, tab);
|
||||
s->fft_permute(s, tab);
|
||||
s->fft_calc(s, tab);
|
||||
|
||||
fft_ref(tab_ref, tab1, fft_nbits);
|
||||
err = check_diff((float *)tab_ref, (float *)tab, fft_size * 2, 1.0);
|
||||
@ -357,7 +359,7 @@ int main(int argc, char **argv)
|
||||
memcpy(tab2, tab1, fft_size * sizeof(FFTSample));
|
||||
tab2[1] = tab1[fft_size_2].re;
|
||||
|
||||
ff_rdft_calc(r, tab2);
|
||||
r->rdft_calc(r, tab2);
|
||||
fft_ref(tab_ref, tab1, fft_nbits);
|
||||
for (i = 0; i < fft_size; i++) {
|
||||
tab[i].re = tab2[i];
|
||||
@ -369,7 +371,7 @@ int main(int argc, char **argv)
|
||||
tab2[i] = tab1[i].re;
|
||||
tab1[i].im = 0;
|
||||
}
|
||||
ff_rdft_calc(r, tab2);
|
||||
r->rdft_calc(r, tab2);
|
||||
fft_ref(tab_ref, tab1, fft_nbits);
|
||||
tab_ref[0].im = tab_ref[fft_size_2].re;
|
||||
err = check_diff((float *)tab_ref, (float *)tab2, fft_size, 1.0);
|
||||
@ -377,7 +379,7 @@ int main(int argc, char **argv)
|
||||
break;
|
||||
case TRANSFORM_DCT:
|
||||
memcpy(tab, tab1, fft_size * sizeof(FFTComplex));
|
||||
ff_dct_calc(d, tab);
|
||||
d->dct_calc(d, tab);
|
||||
if (do_inverse) {
|
||||
idct_ref(tab_ref, tab1, fft_nbits);
|
||||
} else {
|
||||
@ -402,22 +404,22 @@ int main(int argc, char **argv)
|
||||
switch (transform) {
|
||||
case TRANSFORM_MDCT:
|
||||
if (do_inverse) {
|
||||
ff_imdct_calc(m, (float *)tab, (float *)tab1);
|
||||
m->imdct_calc(m, (float *)tab, (float *)tab1);
|
||||
} else {
|
||||
ff_mdct_calc(m, (float *)tab, (float *)tab1);
|
||||
m->mdct_calc(m, (float *)tab, (float *)tab1);
|
||||
}
|
||||
break;
|
||||
case TRANSFORM_FFT:
|
||||
memcpy(tab, tab1, fft_size * sizeof(FFTComplex));
|
||||
ff_fft_calc(s, tab);
|
||||
s->fft_calc(s, tab);
|
||||
break;
|
||||
case TRANSFORM_RDFT:
|
||||
memcpy(tab2, tab1, fft_size * sizeof(FFTSample));
|
||||
ff_rdft_calc(r, tab2);
|
||||
r->rdft_calc(r, tab2);
|
||||
break;
|
||||
case TRANSFORM_DCT:
|
||||
memcpy(tab2, tab1, fft_size * sizeof(FFTSample));
|
||||
ff_dct_calc(d, tab2);
|
||||
d->dct_calc(d, tab2);
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
149
libavcodec/fft.h
149
libavcodec/fft.h
@ -39,7 +39,14 @@ struct FFTContext {
|
||||
/* pre/post rotation tables */
|
||||
FFTSample *tcos;
|
||||
FFTSample *tsin;
|
||||
/**
|
||||
* Do the permutation needed BEFORE calling fft_calc().
|
||||
*/
|
||||
void (*fft_permute)(struct FFTContext *s, FFTComplex *z);
|
||||
/**
|
||||
* Do a complex FFT with the parameters defined in ff_fft_init(). The
|
||||
* input data must be permuted before. No 1.0/sqrt(n) normalization is done.
|
||||
*/
|
||||
void (*fft_calc)(struct FFTContext *s, FFTComplex *z);
|
||||
void (*imdct_calc)(struct FFTContext *s, FFTSample *output, const FFTSample *input);
|
||||
void (*imdct_half)(struct FFTContext *s, FFTSample *output, const FFTSample *input);
|
||||
@ -54,20 +61,13 @@ struct FFTContext {
|
||||
|
||||
#if CONFIG_HARDCODED_TABLES
|
||||
#define COSTABLE_CONST const
|
||||
#define SINTABLE_CONST const
|
||||
#define SINETABLE_CONST const
|
||||
#else
|
||||
#define COSTABLE_CONST
|
||||
#define SINTABLE_CONST
|
||||
#define SINETABLE_CONST
|
||||
#endif
|
||||
|
||||
#define COSTABLE(size) \
|
||||
COSTABLE_CONST DECLARE_ALIGNED(16, FFTSample, ff_cos_##size)[size/2]
|
||||
#define SINTABLE(size) \
|
||||
SINTABLE_CONST DECLARE_ALIGNED(16, FFTSample, ff_sin_##size)[size/2]
|
||||
#define SINETABLE(size) \
|
||||
SINETABLE_CONST DECLARE_ALIGNED(16, float, ff_sine_##size)[size]
|
||||
|
||||
extern COSTABLE(16);
|
||||
extern COSTABLE(32);
|
||||
extern COSTABLE(64);
|
||||
@ -89,20 +89,6 @@ extern COSTABLE_CONST FFTSample* const ff_cos_tabs[17];
|
||||
*/
|
||||
void ff_init_ff_cos_tabs(int index);
|
||||
|
||||
extern SINTABLE(16);
|
||||
extern SINTABLE(32);
|
||||
extern SINTABLE(64);
|
||||
extern SINTABLE(128);
|
||||
extern SINTABLE(256);
|
||||
extern SINTABLE(512);
|
||||
extern SINTABLE(1024);
|
||||
extern SINTABLE(2048);
|
||||
extern SINTABLE(4096);
|
||||
extern SINTABLE(8192);
|
||||
extern SINTABLE(16384);
|
||||
extern SINTABLE(32768);
|
||||
extern SINTABLE(65536);
|
||||
|
||||
/**
|
||||
* Set up a complex FFT.
|
||||
* @param nbits log2 of the length of the input array
|
||||
@ -115,131 +101,12 @@ void ff_fft_init_mmx(FFTContext *s);
|
||||
void ff_fft_init_arm(FFTContext *s);
|
||||
void ff_dct_init_mmx(DCTContext *s);
|
||||
|
||||
/**
|
||||
* Do the permutation needed BEFORE calling ff_fft_calc().
|
||||
*/
|
||||
static inline void ff_fft_permute(FFTContext *s, FFTComplex *z)
|
||||
{
|
||||
s->fft_permute(s, z);
|
||||
}
|
||||
/**
|
||||
* Do a complex FFT with the parameters defined in ff_fft_init(). The
|
||||
* input data must be permuted before. No 1.0/sqrt(n) normalization is done.
|
||||
*/
|
||||
static inline void ff_fft_calc(FFTContext *s, FFTComplex *z)
|
||||
{
|
||||
s->fft_calc(s, z);
|
||||
}
|
||||
void ff_fft_end(FFTContext *s);
|
||||
|
||||
/* MDCT computation */
|
||||
|
||||
static inline void ff_imdct_calc(FFTContext *s, FFTSample *output, const FFTSample *input)
|
||||
{
|
||||
s->imdct_calc(s, output, input);
|
||||
}
|
||||
static inline void ff_imdct_half(FFTContext *s, FFTSample *output, const FFTSample *input)
|
||||
{
|
||||
s->imdct_half(s, output, input);
|
||||
}
|
||||
|
||||
static inline void ff_mdct_calc(FFTContext *s, FFTSample *output,
|
||||
const FFTSample *input)
|
||||
{
|
||||
s->mdct_calc(s, output, input);
|
||||
}
|
||||
|
||||
/**
|
||||
* Maximum window size for ff_kbd_window_init.
|
||||
*/
|
||||
#define FF_KBD_WINDOW_MAX 1024
|
||||
|
||||
/**
|
||||
* Generate a Kaiser-Bessel Derived Window.
|
||||
* @param window pointer to half window
|
||||
* @param alpha determines window shape
|
||||
* @param n size of half window, max FF_KBD_WINDOW_MAX
|
||||
*/
|
||||
void ff_kbd_window_init(float *window, float alpha, int n);
|
||||
|
||||
/**
|
||||
* Generate a sine window.
|
||||
* @param window pointer to half window
|
||||
* @param n size of half window
|
||||
*/
|
||||
void ff_sine_window_init(float *window, int n);
|
||||
|
||||
/**
|
||||
* initialize the specified entry of ff_sine_windows
|
||||
*/
|
||||
void ff_init_ff_sine_windows(int index);
|
||||
extern SINETABLE( 32);
|
||||
extern SINETABLE( 64);
|
||||
extern SINETABLE( 128);
|
||||
extern SINETABLE( 256);
|
||||
extern SINETABLE( 512);
|
||||
extern SINETABLE(1024);
|
||||
extern SINETABLE(2048);
|
||||
extern SINETABLE(4096);
|
||||
extern SINETABLE_CONST float * const ff_sine_windows[13];
|
||||
|
||||
int ff_mdct_init(FFTContext *s, int nbits, int inverse, double scale);
|
||||
void ff_imdct_calc_c(FFTContext *s, FFTSample *output, const FFTSample *input);
|
||||
void ff_imdct_half_c(FFTContext *s, FFTSample *output, const FFTSample *input);
|
||||
void ff_mdct_calc_c(FFTContext *s, FFTSample *output, const FFTSample *input);
|
||||
void ff_mdct_end(FFTContext *s);
|
||||
|
||||
/* Real Discrete Fourier Transform */
|
||||
|
||||
struct RDFTContext {
|
||||
int nbits;
|
||||
int inverse;
|
||||
int sign_convention;
|
||||
|
||||
/* pre/post rotation tables */
|
||||
const FFTSample *tcos;
|
||||
SINTABLE_CONST FFTSample *tsin;
|
||||
FFTContext fft;
|
||||
void (*rdft_calc)(struct RDFTContext *s, FFTSample *z);
|
||||
};
|
||||
|
||||
/**
|
||||
* Set up a real FFT.
|
||||
* @param nbits log2 of the length of the input array
|
||||
* @param trans the type of transform
|
||||
*/
|
||||
int ff_rdft_init(RDFTContext *s, int nbits, enum RDFTransformType trans);
|
||||
void ff_rdft_end(RDFTContext *s);
|
||||
|
||||
void ff_rdft_init_arm(RDFTContext *s);
|
||||
|
||||
static av_always_inline void ff_rdft_calc(RDFTContext *s, FFTSample *data)
|
||||
{
|
||||
s->rdft_calc(s, data);
|
||||
}
|
||||
|
||||
/* Discrete Cosine Transform */
|
||||
|
||||
struct DCTContext {
|
||||
int nbits;
|
||||
int inverse;
|
||||
RDFTContext rdft;
|
||||
const float *costab;
|
||||
FFTSample *csc2;
|
||||
void (*dct_calc)(struct DCTContext *s, FFTSample *data);
|
||||
void (*dct32)(FFTSample *out, const FFTSample *in);
|
||||
};
|
||||
|
||||
/**
|
||||
* Set up DCT.
|
||||
* @param nbits size of the input array:
|
||||
* (1 << nbits) for DCT-II, DCT-III and DST-I
|
||||
* (1 << nbits) + 1 for DCT-I
|
||||
*
|
||||
* @note the first element of the input of DST-I is ignored
|
||||
*/
|
||||
int ff_dct_init(DCTContext *s, int nbits, enum DCTTransformType type);
|
||||
void ff_dct_calc(DCTContext *s, FFTSample *data);
|
||||
void ff_dct_end (DCTContext *s);
|
||||
|
||||
#endif /* AVCODEC_FFT_H */
|
||||
|
@ -41,6 +41,7 @@
|
||||
#include "dsputil.h"
|
||||
#include "fft.h"
|
||||
#include "libavutil/audioconvert.h"
|
||||
#include "sinewin.h"
|
||||
|
||||
#include "imcdata.h"
|
||||
|
||||
@ -564,8 +565,8 @@ static void imc_imdct256(IMCContext *q) {
|
||||
}
|
||||
|
||||
/* FFT */
|
||||
ff_fft_permute(&q->fft, q->samples);
|
||||
ff_fft_calc (&q->fft, q->samples);
|
||||
q->fft.fft_permute(&q->fft, q->samples);
|
||||
q->fft.fft_calc (&q->fft, q->samples);
|
||||
|
||||
/* postrotation, window and reorder */
|
||||
for(i = 0; i < COEFFS/2; i++){
|
||||
|
48
libavcodec/kbdwin.c
Normal file
48
libavcodec/kbdwin.c
Normal file
@ -0,0 +1,48 @@
|
||||
/*
|
||||
* This file is part of FFmpeg.
|
||||
*
|
||||
* FFmpeg is free software; you can redistribute it and/or
|
||||
* modify it under the terms of the GNU Lesser General Public
|
||||
* License as published by the Free Software Foundation; either
|
||||
* version 2.1 of the License, or (at your option) any later version.
|
||||
*
|
||||
* FFmpeg 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
|
||||
* Lesser General Public License for more details.
|
||||
*
|
||||
* You should have received a copy of the GNU Lesser General Public
|
||||
* License along with FFmpeg; if not, write to the Free Software
|
||||
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
|
||||
*/
|
||||
|
||||
#include <assert.h>
|
||||
#include <libavutil/mathematics.h>
|
||||
#include "libavutil/attributes.h"
|
||||
#include "kbdwin.h"
|
||||
|
||||
#define BESSEL_I0_ITER 50 // default: 50 iterations of Bessel I0 approximation
|
||||
|
||||
av_cold void ff_kbd_window_init(float *window, float alpha, int n)
|
||||
{
|
||||
int i, j;
|
||||
double sum = 0.0, bessel, tmp;
|
||||
double local_window[FF_KBD_WINDOW_MAX];
|
||||
double alpha2 = (alpha * M_PI / n) * (alpha * M_PI / n);
|
||||
|
||||
assert(n <= FF_KBD_WINDOW_MAX);
|
||||
|
||||
for (i = 0; i < n; i++) {
|
||||
tmp = i * (n - i) * alpha2;
|
||||
bessel = 1.0;
|
||||
for (j = BESSEL_I0_ITER; j > 0; j--)
|
||||
bessel = bessel * tmp / (j * j) + 1;
|
||||
sum += bessel;
|
||||
local_window[i] = sum;
|
||||
}
|
||||
|
||||
sum++;
|
||||
for (i = 0; i < n; i++)
|
||||
window[i] = sqrt(local_window[i] / sum);
|
||||
}
|
||||
|
35
libavcodec/kbdwin.h
Normal file
35
libavcodec/kbdwin.h
Normal file
@ -0,0 +1,35 @@
|
||||
/*
|
||||
* This file is part of FFmpeg.
|
||||
*
|
||||
* FFmpeg is free software; you can redistribute it and/or
|
||||
* modify it under the terms of the GNU Lesser General Public
|
||||
* License as published by the Free Software Foundation; either
|
||||
* version 2.1 of the License, or (at your option) any later version.
|
||||
*
|
||||
* FFmpeg 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
|
||||
* Lesser General Public License for more details.
|
||||
*
|
||||
* You should have received a copy of the GNU Lesser General Public
|
||||
* License along with FFmpeg; if not, write to the Free Software
|
||||
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
|
||||
*/
|
||||
|
||||
#ifndef AVCODEC_KBDWIN_H
|
||||
#define AVCODEC_KBDWIN_H
|
||||
|
||||
/**
|
||||
* Maximum window size for ff_kbd_window_init.
|
||||
*/
|
||||
#define FF_KBD_WINDOW_MAX 1024
|
||||
|
||||
/**
|
||||
* Generate a Kaiser-Bessel Derived Window.
|
||||
* @param window pointer to half window
|
||||
* @param alpha determines window shape
|
||||
* @param n size of half window, max FF_KBD_WINDOW_MAX
|
||||
*/
|
||||
void ff_kbd_window_init(float *window, float alpha, int n);
|
||||
|
||||
#endif
|
@ -30,33 +30,6 @@
|
||||
* MDCT/IMDCT transforms.
|
||||
*/
|
||||
|
||||
// Generate a Kaiser-Bessel Derived Window.
|
||||
#define BESSEL_I0_ITER 50 // default: 50 iterations of Bessel I0 approximation
|
||||
av_cold void ff_kbd_window_init(float *window, float alpha, int n)
|
||||
{
|
||||
int i, j;
|
||||
double sum = 0.0, bessel, tmp;
|
||||
double local_window[FF_KBD_WINDOW_MAX];
|
||||
double alpha2 = (alpha * M_PI / n) * (alpha * M_PI / n);
|
||||
|
||||
assert(n <= FF_KBD_WINDOW_MAX);
|
||||
|
||||
for (i = 0; i < n; i++) {
|
||||
tmp = i * (n - i) * alpha2;
|
||||
bessel = 1.0;
|
||||
for (j = BESSEL_I0_ITER; j > 0; j--)
|
||||
bessel = bessel * tmp / (j * j) + 1;
|
||||
sum += bessel;
|
||||
local_window[i] = sum;
|
||||
}
|
||||
|
||||
sum++;
|
||||
for (i = 0; i < n; i++)
|
||||
window[i] = sqrt(local_window[i] / sum);
|
||||
}
|
||||
|
||||
#include "mdct_tablegen.h"
|
||||
|
||||
/**
|
||||
* init MDCT or IMDCT computation.
|
||||
*/
|
||||
@ -146,7 +119,7 @@ void ff_imdct_half_c(FFTContext *s, FFTSample *output, const FFTSample *input)
|
||||
in1 += 2;
|
||||
in2 -= 2;
|
||||
}
|
||||
ff_fft_calc(s, z);
|
||||
s->fft_calc(s, z);
|
||||
|
||||
/* post rotation + reordering */
|
||||
for(k = 0; k < n8; k++) {
|
||||
@ -213,7 +186,7 @@ void ff_mdct_calc_c(FFTContext *s, FFTSample *out, const FFTSample *input)
|
||||
CMUL(x[j].re, x[j].im, re, im, -tcos[n8 + i], tsin[n8 + i]);
|
||||
}
|
||||
|
||||
ff_fft_calc(s, x);
|
||||
s->fft_calc(s, x);
|
||||
|
||||
/* post rotation */
|
||||
for(i=0;i<n8;i++) {
|
||||
|
@ -33,7 +33,7 @@
|
||||
#include "avcodec.h"
|
||||
#include "get_bits.h"
|
||||
#include "dsputil.h"
|
||||
#include "fft.h"
|
||||
#include "dct.h"
|
||||
|
||||
#define CONFIG_AUDIO_NONSHORT 0
|
||||
|
||||
|
@ -39,6 +39,7 @@
|
||||
#include "dsputil.h"
|
||||
#include "fft.h"
|
||||
#include "fmtconvert.h"
|
||||
#include "sinewin.h"
|
||||
|
||||
#define ALT_BITSTREAM_READER_LE
|
||||
#include "get_bits.h"
|
||||
@ -121,7 +122,7 @@ static void nelly_decode_block(NellyMoserDecodeContext *s,
|
||||
memset(&aptr[NELLY_FILL_LEN], 0,
|
||||
(NELLY_BUF_LEN - NELLY_FILL_LEN) * sizeof(float));
|
||||
|
||||
ff_imdct_calc(&s->imdct_ctx, s->imdct_out, aptr);
|
||||
s->imdct_ctx.imdct_calc(&s->imdct_ctx, s->imdct_out, aptr);
|
||||
/* XXX: overlapping and windowing should be part of a more
|
||||
generic imdct function */
|
||||
overlap_and_window(s, s->state, aptr, s->imdct_out);
|
||||
|
@ -39,6 +39,7 @@
|
||||
#include "avcodec.h"
|
||||
#include "dsputil.h"
|
||||
#include "fft.h"
|
||||
#include "sinewin.h"
|
||||
|
||||
#define BITSTREAM_WRITER_LE
|
||||
#include "put_bits.h"
|
||||
@ -116,13 +117,13 @@ static void apply_mdct(NellyMoserEncodeContext *s)
|
||||
s->dsp.vector_fmul(s->in_buff, s->buf[s->bufsel], ff_sine_128, NELLY_BUF_LEN);
|
||||
s->dsp.vector_fmul_reverse(s->in_buff + NELLY_BUF_LEN, s->buf[s->bufsel] + NELLY_BUF_LEN, ff_sine_128,
|
||||
NELLY_BUF_LEN);
|
||||
ff_mdct_calc(&s->mdct_ctx, s->mdct_out, s->in_buff);
|
||||
s->mdct_ctx.mdct_calc(&s->mdct_ctx, s->mdct_out, s->in_buff);
|
||||
|
||||
s->dsp.vector_fmul(s->buf[s->bufsel] + NELLY_BUF_LEN, s->buf[s->bufsel] + NELLY_BUF_LEN,
|
||||
ff_sine_128, NELLY_BUF_LEN);
|
||||
s->dsp.vector_fmul_reverse(s->buf[s->bufsel] + 2 * NELLY_BUF_LEN, s->buf[1 - s->bufsel], ff_sine_128,
|
||||
NELLY_BUF_LEN);
|
||||
ff_mdct_calc(&s->mdct_ctx, s->mdct_out + NELLY_BUF_LEN, s->buf[s->bufsel] + NELLY_BUF_LEN);
|
||||
s->mdct_ctx.mdct_calc(&s->mdct_ctx, s->mdct_out + NELLY_BUF_LEN, s->buf[s->bufsel] + NELLY_BUF_LEN);
|
||||
}
|
||||
|
||||
static av_cold int encode_init(AVCodecContext *avctx)
|
||||
|
@ -1588,7 +1588,7 @@ static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
|
||||
int i;
|
||||
q->fft.complex[channel][0].re *= 2.0f;
|
||||
q->fft.complex[channel][0].im = 0.0f;
|
||||
ff_rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
|
||||
q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
|
||||
/* add samples to output buffer */
|
||||
for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++)
|
||||
q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain;
|
||||
|
@ -21,7 +21,7 @@
|
||||
#include <stdlib.h>
|
||||
#include <math.h>
|
||||
#include "libavutil/mathematics.h"
|
||||
#include "fft.h"
|
||||
#include "rdft.h"
|
||||
|
||||
/**
|
||||
* @file
|
||||
@ -65,8 +65,8 @@ static void ff_rdft_calc_c(RDFTContext* s, FFTSample* data)
|
||||
const FFTSample *tsin = s->tsin;
|
||||
|
||||
if (!s->inverse) {
|
||||
ff_fft_permute(&s->fft, (FFTComplex*)data);
|
||||
ff_fft_calc(&s->fft, (FFTComplex*)data);
|
||||
s->fft.fft_permute(&s->fft, (FFTComplex*)data);
|
||||
s->fft.fft_calc(&s->fft, (FFTComplex*)data);
|
||||
}
|
||||
/* i=0 is a special case because of packing, the DC term is real, so we
|
||||
are going to throw the N/2 term (also real) in with it. */
|
||||
@ -91,8 +91,8 @@ static void ff_rdft_calc_c(RDFTContext* s, FFTSample* data)
|
||||
if (s->inverse) {
|
||||
data[0] *= k1;
|
||||
data[1] *= k1;
|
||||
ff_fft_permute(&s->fft, (FFTComplex*)data);
|
||||
ff_fft_calc(&s->fft, (FFTComplex*)data);
|
||||
s->fft.fft_permute(&s->fft, (FFTComplex*)data);
|
||||
s->fft.fft_calc(&s->fft, (FFTComplex*)data);
|
||||
}
|
||||
}
|
||||
|
||||
|
74
libavcodec/rdft.h
Normal file
74
libavcodec/rdft.h
Normal file
@ -0,0 +1,74 @@
|
||||
/*
|
||||
* (I)RDFT transforms
|
||||
* Copyright (c) 2009 Alex Converse <alex dot converse at gmail dot com>
|
||||
*
|
||||
* This file is part of FFmpeg.
|
||||
*
|
||||
* FFmpeg is free software; you can redistribute it and/or
|
||||
* modify it under the terms of the GNU Lesser General Public
|
||||
* License as published by the Free Software Foundation; either
|
||||
* version 2.1 of the License, or (at your option) any later version.
|
||||
*
|
||||
* FFmpeg 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
|
||||
* Lesser General Public License for more details.
|
||||
*
|
||||
* You should have received a copy of the GNU Lesser General Public
|
||||
* License along with FFmpeg; if not, write to the Free Software
|
||||
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
|
||||
*/
|
||||
|
||||
#ifndef AVCODEC_RDFT_H
|
||||
#define AVCODEC_RDFT_H
|
||||
|
||||
#include "config.h"
|
||||
#include "fft.h"
|
||||
|
||||
#if CONFIG_HARDCODED_TABLES
|
||||
# define SINTABLE_CONST const
|
||||
#else
|
||||
# define SINTABLE_CONST
|
||||
#endif
|
||||
|
||||
#define SINTABLE(size) \
|
||||
SINTABLE_CONST DECLARE_ALIGNED(16, FFTSample, ff_sin_##size)[size/2]
|
||||
|
||||
extern SINTABLE(16);
|
||||
extern SINTABLE(32);
|
||||
extern SINTABLE(64);
|
||||
extern SINTABLE(128);
|
||||
extern SINTABLE(256);
|
||||
extern SINTABLE(512);
|
||||
extern SINTABLE(1024);
|
||||
extern SINTABLE(2048);
|
||||
extern SINTABLE(4096);
|
||||
extern SINTABLE(8192);
|
||||
extern SINTABLE(16384);
|
||||
extern SINTABLE(32768);
|
||||
extern SINTABLE(65536);
|
||||
|
||||
struct RDFTContext {
|
||||
int nbits;
|
||||
int inverse;
|
||||
int sign_convention;
|
||||
|
||||
/* pre/post rotation tables */
|
||||
const FFTSample *tcos;
|
||||
SINTABLE_CONST FFTSample *tsin;
|
||||
FFTContext fft;
|
||||
void (*rdft_calc)(struct RDFTContext *s, FFTSample *z);
|
||||
};
|
||||
|
||||
/**
|
||||
* Set up a real FFT.
|
||||
* @param nbits log2 of the length of the input array
|
||||
* @param trans the type of transform
|
||||
*/
|
||||
int ff_rdft_init(RDFTContext *s, int nbits, enum RDFTransformType trans);
|
||||
void ff_rdft_end(RDFTContext *s);
|
||||
|
||||
void ff_rdft_init_arm(RDFTContext *s);
|
||||
|
||||
|
||||
#endif
|
20
libavcodec/sinewin.c
Normal file
20
libavcodec/sinewin.c
Normal file
@ -0,0 +1,20 @@
|
||||
/*
|
||||
* This file is part of FFmpeg.
|
||||
*
|
||||
* FFmpeg is free software; you can redistribute it and/or
|
||||
* modify it under the terms of the GNU Lesser General Public
|
||||
* License as published by the Free Software Foundation; either
|
||||
* version 2.1 of the License, or (at your option) any later version.
|
||||
*
|
||||
* FFmpeg 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
|
||||
* Lesser General Public License for more details.
|
||||
*
|
||||
* You should have received a copy of the GNU Lesser General Public
|
||||
* License along with FFmpeg; if not, write to the Free Software
|
||||
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
|
||||
*/
|
||||
|
||||
#include "sinewin.h"
|
||||
#include "sinewin_tablegen.h"
|
59
libavcodec/sinewin.h
Normal file
59
libavcodec/sinewin.h
Normal file
@ -0,0 +1,59 @@
|
||||
/*
|
||||
* Copyright (c) 2008 Robert Swain
|
||||
*
|
||||
* This file is part of FFmpeg.
|
||||
*
|
||||
* FFmpeg is free software; you can redistribute it and/or
|
||||
* modify it under the terms of the GNU Lesser General Public
|
||||
* License as published by the Free Software Foundation; either
|
||||
* version 2.1 of the License, or (at your option) any later version.
|
||||
*
|
||||
* FFmpeg 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
|
||||
* Lesser General Public License for more details.
|
||||
*
|
||||
* You should have received a copy of the GNU Lesser General Public
|
||||
* License along with FFmpeg; if not, write to the Free Software
|
||||
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
|
||||
*/
|
||||
|
||||
#ifndef AVCODEC_SINEWIN_H
|
||||
#define AVCODEC_SINEWIN_H
|
||||
|
||||
#include "config.h"
|
||||
#include "libavutil/mem.h"
|
||||
|
||||
#if CONFIG_HARDCODED_TABLES
|
||||
# define SINETABLE_CONST const
|
||||
#else
|
||||
# define SINETABLE_CONST
|
||||
#endif
|
||||
|
||||
#define SINETABLE(size) \
|
||||
SINETABLE_CONST DECLARE_ALIGNED(16, float, ff_sine_##size)[size]
|
||||
|
||||
/**
|
||||
* Generate a sine window.
|
||||
* @param window pointer to half window
|
||||
* @param n size of half window
|
||||
*/
|
||||
void ff_sine_window_init(float *window, int n);
|
||||
|
||||
/**
|
||||
* initialize the specified entry of ff_sine_windows
|
||||
*/
|
||||
void ff_init_ff_sine_windows(int index);
|
||||
|
||||
extern SINETABLE( 32);
|
||||
extern SINETABLE( 64);
|
||||
extern SINETABLE( 128);
|
||||
extern SINETABLE( 256);
|
||||
extern SINETABLE( 512);
|
||||
extern SINETABLE(1024);
|
||||
extern SINETABLE(2048);
|
||||
extern SINETABLE(4096);
|
||||
|
||||
extern SINETABLE_CONST float * const ff_sine_windows[13];
|
||||
|
||||
#endif
|
@ -1,5 +1,5 @@
|
||||
/*
|
||||
* Generate a header file for hardcoded MDCT tables
|
||||
* Generate a header file for hardcoded sine windows
|
||||
*
|
||||
* Copyright (c) 2009 Reimar Döffinger <Reimar.Doeffinger@gmx.de>
|
||||
*
|
||||
@ -29,7 +29,7 @@
|
||||
#ifndef M_PI
|
||||
#define M_PI 3.14159265358979323846
|
||||
#endif
|
||||
#include "mdct_tablegen.h"
|
||||
#include "sinewin_tablegen.h"
|
||||
#include "tableprint.h"
|
||||
|
||||
int main(void)
|
@ -1,5 +1,5 @@
|
||||
/*
|
||||
* Header file for hardcoded MDCT tables
|
||||
* Header file for hardcoded sine windows
|
||||
*
|
||||
* Copyright (c) 2009 Reimar Döffinger <Reimar.Doeffinger@gmx.de>
|
||||
*
|
||||
@ -36,7 +36,7 @@ SINETABLE(1024);
|
||||
SINETABLE(2048);
|
||||
SINETABLE(4096);
|
||||
#else
|
||||
#include "libavcodec/mdct_tables.h"
|
||||
#include "libavcodec/sinewin_tables.h"
|
||||
#endif
|
||||
|
||||
SINETABLE_CONST float * const ff_sine_windows[] = {
|
@ -29,7 +29,7 @@ static void synth_filter_float(FFTContext *imdct,
|
||||
float *synth_buf= synth_buf_ptr + *synth_buf_offset;
|
||||
int i, j;
|
||||
|
||||
ff_imdct_half(imdct, synth_buf, in);
|
||||
imdct->imdct_half(imdct, synth_buf, in);
|
||||
|
||||
for (i = 0; i < 16; i++){
|
||||
float a= synth_buf2[i ];
|
||||
|
@ -24,6 +24,7 @@
|
||||
#include "dsputil.h"
|
||||
#include "fft.h"
|
||||
#include "lsp.h"
|
||||
#include "sinewin.h"
|
||||
|
||||
#include <math.h>
|
||||
#include <stdint.h>
|
||||
@ -608,6 +609,7 @@ static void dec_lpc_spectrum_inv(TwinContext *tctx, float *lsp,
|
||||
static void imdct_and_window(TwinContext *tctx, enum FrameType ftype, int wtype,
|
||||
float *in, float *prev, int ch)
|
||||
{
|
||||
FFTContext *mdct = &tctx->mdct_ctx[ftype];
|
||||
const ModeTab *mtab = tctx->mtab;
|
||||
int bsize = mtab->size / mtab->fmode[ftype].sub;
|
||||
int size = mtab->size;
|
||||
@ -640,7 +642,7 @@ static void imdct_and_window(TwinContext *tctx, enum FrameType ftype, int wtype,
|
||||
|
||||
wsize = types_sizes[wtype_to_wsize[sub_wtype]];
|
||||
|
||||
ff_imdct_half(&tctx->mdct_ctx[ftype], buf1 + bsize*j, in + bsize*j);
|
||||
mdct->imdct_half(mdct, buf1 + bsize*j, in + bsize*j);
|
||||
|
||||
tctx->dsp.vector_fmul_window(out2,
|
||||
prev_buf + (bsize-wsize)/2,
|
||||
|
@ -1448,7 +1448,7 @@ void vorbis_inverse_coupling(float *mag, float *ang, int blocksize)
|
||||
static int vorbis_parse_audio_packet(vorbis_context *vc)
|
||||
{
|
||||
GetBitContext *gb = &vc->gb;
|
||||
|
||||
FFTContext *mdct;
|
||||
uint_fast8_t previous_window = vc->previous_window;
|
||||
uint_fast8_t mode_number;
|
||||
uint_fast8_t blockflag;
|
||||
@ -1552,11 +1552,13 @@ static int vorbis_parse_audio_packet(vorbis_context *vc)
|
||||
|
||||
// Dotproduct, MDCT
|
||||
|
||||
mdct = &vc->mdct[blockflag];
|
||||
|
||||
for (j = vc->audio_channels-1;j >= 0; j--) {
|
||||
ch_floor_ptr = vc->channel_floors + j * blocksize / 2;
|
||||
ch_res_ptr = vc->channel_residues + res_chan[j] * blocksize / 2;
|
||||
vc->dsp.vector_fmul(ch_floor_ptr, ch_floor_ptr, ch_res_ptr, blocksize / 2);
|
||||
ff_imdct_half(&vc->mdct[blockflag], ch_res_ptr, ch_floor_ptr);
|
||||
mdct->imdct_half(mdct, ch_res_ptr, ch_floor_ptr);
|
||||
}
|
||||
|
||||
// Overlap/add, save data for next overlapping FPMATH
|
||||
|
@ -935,7 +935,7 @@ static int apply_window_and_mdct(vorbis_enc_context *venc, const signed short *a
|
||||
}
|
||||
|
||||
for (channel = 0; channel < venc->channels; channel++)
|
||||
ff_mdct_calc(&venc->mdct[0], venc->coeffs + channel * window_len,
|
||||
venc->mdct[0].mdct_calc(&venc->mdct[0], venc->coeffs + channel * window_len,
|
||||
venc->samples + channel * window_len * 2);
|
||||
|
||||
if (samples) {
|
||||
|
@ -20,6 +20,7 @@
|
||||
*/
|
||||
|
||||
#include "avcodec.h"
|
||||
#include "sinewin.h"
|
||||
#include "wma.h"
|
||||
#include "wmadata.h"
|
||||
|
||||
|
@ -447,6 +447,7 @@ static int wma_decode_block(WMACodecContext *s)
|
||||
int coef_nb_bits, total_gain;
|
||||
int nb_coefs[MAX_CHANNELS];
|
||||
float mdct_norm;
|
||||
FFTContext *mdct;
|
||||
|
||||
#ifdef TRACE
|
||||
tprintf(s->avctx, "***decode_block: %d:%d\n", s->frame_count - 1, s->block_num);
|
||||
@ -742,12 +743,14 @@ static int wma_decode_block(WMACodecContext *s)
|
||||
}
|
||||
|
||||
next:
|
||||
mdct = &s->mdct_ctx[bsize];
|
||||
|
||||
for(ch = 0; ch < s->nb_channels; ch++) {
|
||||
int n4, index;
|
||||
|
||||
n4 = s->block_len / 2;
|
||||
if(s->channel_coded[ch]){
|
||||
ff_imdct_calc(&s->mdct_ctx[bsize], s->output, s->coefs[ch]);
|
||||
mdct->imdct_calc(mdct, s->output, s->coefs[ch]);
|
||||
}else if(!(s->ms_stereo && ch==1))
|
||||
memset(s->output, 0, sizeof(s->output));
|
||||
|
||||
|
@ -77,6 +77,7 @@ static int encode_init(AVCodecContext * avctx){
|
||||
static void apply_window_and_mdct(AVCodecContext * avctx, const signed short * audio, int len) {
|
||||
WMACodecContext *s = avctx->priv_data;
|
||||
int window_index= s->frame_len_bits - s->block_len_bits;
|
||||
FFTContext *mdct = &s->mdct_ctx[window_index];
|
||||
int i, j, channel;
|
||||
const float * win = s->windows[window_index];
|
||||
int window_len = 1 << s->block_len_bits;
|
||||
@ -89,7 +90,7 @@ static void apply_window_and_mdct(AVCodecContext * avctx, const signed short * a
|
||||
s->output[i+window_len] = audio[j] / n * win[window_len - i - 1];
|
||||
s->frame_out[channel][i] = audio[j] / n * win[i];
|
||||
}
|
||||
ff_mdct_calc(&s->mdct_ctx[window_index], s->coefs[channel], s->output);
|
||||
mdct->mdct_calc(mdct, s->coefs[channel], s->output);
|
||||
}
|
||||
}
|
||||
|
||||
|
@ -92,6 +92,7 @@
|
||||
#include "put_bits.h"
|
||||
#include "wmaprodata.h"
|
||||
#include "dsputil.h"
|
||||
#include "sinewin.h"
|
||||
#include "wma.h"
|
||||
|
||||
/** current decoder limitations */
|
||||
@ -1222,6 +1223,7 @@ static int decode_subframe(WMAProDecodeCtx *s)
|
||||
get_bits_count(&s->gb) - s->subframe_offset);
|
||||
|
||||
if (transmit_coeffs) {
|
||||
FFTContext *mdct = &s->mdct_ctx[av_log2(subframe_len) - WMAPRO_BLOCK_MIN_BITS];
|
||||
/** reconstruct the per channel data */
|
||||
inverse_channel_transform(s);
|
||||
for (i = 0; i < s->channels_for_cur_subframe; i++) {
|
||||
@ -1246,9 +1248,8 @@ static int decode_subframe(WMAProDecodeCtx *s)
|
||||
quant, end - start);
|
||||
}
|
||||
|
||||
/** apply imdct (ff_imdct_half == DCTIV with reverse) */
|
||||
ff_imdct_half(&s->mdct_ctx[av_log2(subframe_len) - WMAPRO_BLOCK_MIN_BITS],
|
||||
s->channel[c].coeffs, s->tmp);
|
||||
/** apply imdct (imdct_half == DCTIV with reverse) */
|
||||
mdct->imdct_half(mdct, s->channel[c].coeffs, s->tmp);
|
||||
}
|
||||
}
|
||||
|
||||
|
@ -36,8 +36,9 @@
|
||||
#include "acelp_filters.h"
|
||||
#include "lsp.h"
|
||||
#include "libavutil/lzo.h"
|
||||
#include "avfft.h"
|
||||
#include "fft.h"
|
||||
#include "dct.h"
|
||||
#include "rdft.h"
|
||||
#include "sinewin.h"
|
||||
|
||||
#define MAX_BLOCKS 8 ///< maximum number of blocks per frame
|
||||
#define MAX_LSPS 16 ///< maximum filter order
|
||||
@ -558,7 +559,7 @@ static void calc_input_response(WMAVoiceContext *s, float *lpcs,
|
||||
int n, idx;
|
||||
|
||||
/* Create frequency power spectrum of speech input (i.e. RDFT of LPCs) */
|
||||
ff_rdft_calc(&s->rdft, lpcs);
|
||||
s->rdft.rdft_calc(&s->rdft, lpcs);
|
||||
#define log_range(var, assign) do { \
|
||||
float tmp = log10f(assign); var = tmp; \
|
||||
max = FFMAX(max, tmp); min = FFMIN(min, tmp); \
|
||||
@ -601,8 +602,8 @@ static void calc_input_response(WMAVoiceContext *s, float *lpcs,
|
||||
* is a sinus input) by doing a phase shift (in theory, H(sin())=cos()).
|
||||
* Hilbert_Transform(RDFT(x)) = Laplace_Transform(x), which calculates the
|
||||
* "moment" of the LPCs in this filter. */
|
||||
ff_dct_calc(&s->dct, lpcs);
|
||||
ff_dct_calc(&s->dst, lpcs);
|
||||
s->dct.dct_calc(&s->dct, lpcs);
|
||||
s->dst.dct_calc(&s->dst, lpcs);
|
||||
|
||||
/* Split out the coefficient indexes into phase/magnitude pairs */
|
||||
idx = 255 + av_clip(lpcs[64], -255, 255);
|
||||
@ -623,7 +624,7 @@ static void calc_input_response(WMAVoiceContext *s, float *lpcs,
|
||||
coeffs[1] = last_coeff;
|
||||
|
||||
/* move into real domain */
|
||||
ff_rdft_calc(&s->irdft, coeffs);
|
||||
s->irdft.rdft_calc(&s->irdft, coeffs);
|
||||
|
||||
/* tilt correction and normalize scale */
|
||||
memset(&coeffs[remainder], 0, sizeof(coeffs[0]) * (128 - remainder));
|
||||
@ -693,8 +694,8 @@ static void wiener_denoise(WMAVoiceContext *s, int fcb_type,
|
||||
/* apply coefficients (in frequency spectrum domain), i.e. complex
|
||||
* number multiplication */
|
||||
memset(&synth_pf[size], 0, sizeof(synth_pf[0]) * (128 - size));
|
||||
ff_rdft_calc(&s->rdft, synth_pf);
|
||||
ff_rdft_calc(&s->rdft, coeffs);
|
||||
s->rdft.rdft_calc(&s->rdft, synth_pf);
|
||||
s->rdft.rdft_calc(&s->rdft, coeffs);
|
||||
synth_pf[0] *= coeffs[0];
|
||||
synth_pf[1] *= coeffs[1];
|
||||
for (n = 1; n < 64; n++) {
|
||||
@ -702,7 +703,7 @@ static void wiener_denoise(WMAVoiceContext *s, int fcb_type,
|
||||
synth_pf[n * 2] = v1 * coeffs[n * 2] - v2 * coeffs[n * 2 + 1];
|
||||
synth_pf[n * 2 + 1] = v2 * coeffs[n * 2] + v1 * coeffs[n * 2 + 1];
|
||||
}
|
||||
ff_rdft_calc(&s->irdft, synth_pf);
|
||||
s->irdft.rdft_calc(&s->irdft, synth_pf);
|
||||
}
|
||||
|
||||
/* merge filter output with the history of previous runs */
|
||||
|
@ -18,6 +18,7 @@
|
||||
|
||||
#include "libavutil/cpu.h"
|
||||
#include "libavcodec/dsputil.h"
|
||||
#include "libavcodec/dct.h"
|
||||
#include "fft.h"
|
||||
|
||||
av_cold void ff_fft_init_mmx(FFTContext *s)
|
||||
|
Loading…
Reference in New Issue
Block a user