mirror of https://git.ffmpeg.org/ffmpeg.git
Direct Stream Digital (DSD) decoder
Signed-off-by: Peter Ross <pross@xvid.org> Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Peter Ross
Michael Niedermayer
parent
add5d7a8ce
commit
5f4f9ee99f
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@ -17,6 +17,7 @@ version <next>:
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- GDI screen grabbing for Windows
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- alternative rendition support for HTTP Live Streaming
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- AVFoundation input device
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- Direct Stream Digital (DSD) decoder
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version 2.2:
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@ -898,6 +898,10 @@ following image formats are supported:
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@item DPCM Sol @tab @tab X
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@item DPCM Xan @tab @tab X
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@tab Used in Origin's Wing Commander IV AVI files.
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@item DSD (Direct Stream Digitial), least significant bit first @tab @tab X
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@item DSD (Direct Stream Digitial), most significant bit first @tab @tab X
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@item DSD (Direct Stream Digitial), least significant bit first, planar @tab @tab X
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@item DSD (Direct Stream Digitial), most significant bit first, planar @tab @tab X
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@item DSP Group TrueSpeech @tab @tab X
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@item DV audio @tab @tab X
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@item Enhanced AC-3 @tab X @tab X
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@ -179,6 +179,8 @@ OBJS-$(CONFIG_DNXHD_DECODER) += dnxhddec.o dnxhddata.o
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OBJS-$(CONFIG_DNXHD_ENCODER) += dnxhdenc.o dnxhddata.o
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OBJS-$(CONFIG_DPX_DECODER) += dpx.o
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OBJS-$(CONFIG_DPX_ENCODER) += dpxenc.o
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OBJS-$(CONFIG_DSD_LSBF_DECODER) += dsddec.o
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OBJS-$(CONFIG_DSD_MSBF_DECODER) += dsddec.o
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OBJS-$(CONFIG_DSICINAUDIO_DECODER) += dsicinav.o
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OBJS-$(CONFIG_DSICINVIDEO_DECODER) += dsicinav.o
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OBJS-$(CONFIG_DVBSUB_DECODER) += dvbsubdec.o
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@ -847,6 +849,7 @@ HOSTPROGS = aac_tablegen \
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aacps_tablegen \
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cbrt_tablegen \
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cos_tablegen \
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dsd_tablegen \
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dv_tablegen \
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motionpixels_tablegen \
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mpegaudio_tablegen \
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@ -871,7 +874,7 @@ else
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$(SUBDIR)%_tablegen$(HOSTEXESUF): HOSTCFLAGS += -DCONFIG_SMALL=0
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endif
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GEN_HEADERS = cbrt_tables.h aacps_tables.h aac_tables.h dv_tables.h \
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GEN_HEADERS = cbrt_tables.h aacps_tables.h aac_tables.h dsd_tables.h dv_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), $(GEN_HEADERS))
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@ -883,6 +886,7 @@ ifdef CONFIG_HARDCODED_TABLES
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$(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)dsddec.o: $(SUBDIR)dsd_tables.h
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$(SUBDIR)dvenc.o: $(SUBDIR)dv_tables.h
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$(SUBDIR)sinewin.o: $(SUBDIR)sinewin_tables.h
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$(SUBDIR)mpegaudiodec_fixed.o: $(SUBDIR)mpegaudio_tables.h
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@ -337,6 +337,10 @@ void avcodec_register_all(void)
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REGISTER_DECODER(BMV_AUDIO, bmv_audio);
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REGISTER_DECODER(COOK, cook);
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REGISTER_ENCDEC (DCA, dca);
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REGISTER_DECODER(DSD_LSBF, dsd_lsbf);
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REGISTER_DECODER(DSD_MSBF, dsd_msbf);
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REGISTER_DECODER(DSD_LSBF_PLANAR, dsd_lsbf_planar);
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REGISTER_DECODER(DSD_MSBF_PLANAR, dsd_msbf_planar);
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REGISTER_DECODER(DSICINAUDIO, dsicinaudio);
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REGISTER_ENCDEC (EAC3, eac3);
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REGISTER_DECODER(EVRC, evrc);
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@ -489,6 +489,10 @@ enum AVCodecID {
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AV_CODEC_ID_TAK = MKBETAG('t','B','a','K'),
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AV_CODEC_ID_EVRC = MKBETAG('s','e','v','c'),
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AV_CODEC_ID_SMV = MKBETAG('s','s','m','v'),
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AV_CODEC_ID_DSD_LSBF = MKBETAG('D','S','D','L'),
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AV_CODEC_ID_DSD_MSBF = MKBETAG('D','S','D','M'),
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AV_CODEC_ID_DSD_LSBF_PLANAR = MKBETAG('D','S','D','1'),
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AV_CODEC_ID_DSD_MSBF_PLANAR = MKBETAG('D','S','D','8'),
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/* subtitle codecs */
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AV_CODEC_ID_FIRST_SUBTITLE = 0x17000, ///< A dummy ID pointing at the start of subtitle codecs.
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@ -2460,6 +2460,34 @@ static const AVCodecDescriptor codec_descriptors[] = {
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.long_name = NULL_IF_CONFIG_SMALL("SMV (Selectable Mode Vocoder)"),
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.props = AV_CODEC_PROP_LOSSY,
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},
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{
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.id = AV_CODEC_ID_DSD_LSBF,
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.type = AVMEDIA_TYPE_AUDIO,
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.name = "dsd_lsbf",
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.long_name = NULL_IF_CONFIG_SMALL("DSD (Direct Stream Digital), least significant bit first"),
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.props = AV_CODEC_PROP_LOSSY,
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},
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{
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.id = AV_CODEC_ID_DSD_MSBF,
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.type = AVMEDIA_TYPE_AUDIO,
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.name = "dsd_msbf",
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.long_name = NULL_IF_CONFIG_SMALL("DSD (Direct Stream Digital), most significant bit first"),
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.props = AV_CODEC_PROP_LOSSY,
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},
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{
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.id = AV_CODEC_ID_DSD_LSBF_PLANAR,
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.type = AVMEDIA_TYPE_AUDIO,
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.name = "dsd_lsbf_planar",
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.long_name = NULL_IF_CONFIG_SMALL("DSD (Direct Stream Digital), least significant bit first, planar"),
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.props = AV_CODEC_PROP_LOSSY,
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},
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{
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.id = AV_CODEC_ID_DSD_MSBF_PLANAR,
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.type = AVMEDIA_TYPE_AUDIO,
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.name = "dsd_msbf_planar",
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.long_name = NULL_IF_CONFIG_SMALL("DSD (Direct Stream Digital), most significant bit first, planar"),
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.props = AV_CODEC_PROP_LOSSY,
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},
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/* subtitle codecs */
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{
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@ -0,0 +1,38 @@
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/*
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* Generate a header file for hardcoded DSD tables
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include <stdlib.h>
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#define CONFIG_HARDCODED_TABLES 0
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#include "dsd_tablegen.h"
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#include "tableprint.h"
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#include <inttypes.h>
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int main(void)
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{
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dsd_ctables_tableinit();
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write_fileheader();
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printf("static const double ctables[CTABLES][256] = {\n");
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write_float_2d_array(ctables, CTABLES, 256);
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printf("};\n");
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return 0;
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}
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@ -0,0 +1,95 @@
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/*
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* Header file for hardcoded DSD tables
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* based on BSD licensed dsd2pcm by Sebastian Gesemann
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* Copyright (c) 2009, 2011 Sebastian Gesemann. All rights reserved.
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#ifndef AVCODEC_DSD_TABLEGEN_H
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#define AVCODEC_DSD_TABLEGEN_H
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#include <stdint.h>
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#include "libavutil/attributes.h"
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#define HTAPS 48 /** number of FIR constants */
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#define CTABLES ((HTAPS + 7) / 8) /** number of "8 MACs" lookup tables */
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#if CONFIG_HARDCODED_TABLES
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#define dsd_ctables_tableinit()
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#include "libavcodec/dsd_tables.h"
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#else
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#include "libavutil/common.h"
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/*
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* Properties of this 96-tap lowpass filter when applied on a signal
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* with sampling rate of 44100*64 Hz:
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*
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* () has a delay of 17 microseconds.
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*
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* () flat response up to 48 kHz
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*
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* () if you downsample afterwards by a factor of 8, the
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* spectrum below 70 kHz is practically alias-free.
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*
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* () stopband rejection is about 160 dB
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*
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* The coefficient tables ("ctables") take only 6 Kibi Bytes and
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* should fit into a modern processor's fast cache.
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*/
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/**
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* The 2nd half (48 coeffs) of a 96-tap symmetric lowpass filter
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*/
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static const double htaps[HTAPS] = {
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0.09950731974056658, 0.09562845727714668, 0.08819647126516944,
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0.07782552527068175, 0.06534876523171299, 0.05172629311427257,
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0.0379429484910187, 0.02490921351762261, 0.0133774746265897,
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0.003883043418804416, -0.003284703416210726, -0.008080250212687497,
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-0.01067241812471033, -0.01139427235000863, -0.0106813877974587,
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-0.009007905078766049, -0.006828859761015335, -0.004535184322001496,
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-0.002425035959059578, -0.0006922187080790708, 0.0005700762133516592,
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0.001353838005269448, 0.001713709169690937, 0.001742046839472948,
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0.001545601648013235, 0.001226696225277855, 0.0008704322683580222,
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0.0005381636200535649, 0.000266446345425276, 7.002968738383528e-05,
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-5.279407053811266e-05, -0.0001140625650874684, -0.0001304796361231895,
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-0.0001189970287491285, -9.396247155265073e-05, -6.577634378272832e-05,
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-4.07492895872535e-05, -2.17407957554587e-05, -9.163058931391722e-06,
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-2.017460145032201e-06, 1.249721855219005e-06, 2.166655190537392e-06,
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1.930520892991082e-06, 1.319400334374195e-06, 7.410039764949091e-07,
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3.423230509967409e-07, 1.244182214744588e-07, 3.130441005359396e-08
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};
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static float ctables[CTABLES][256];
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static av_cold void dsd_ctables_tableinit(void)
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{
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int t, e, m, k;
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double acc;
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for (t = 0; t < CTABLES; ++t) {
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k = FFMIN(HTAPS - t * 8, 8);
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for (e = 0; e < 256; ++e) {
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acc = 0.0;
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for (m = 0; m < k; ++m)
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acc += (((e >> (7 - m)) & 1) * 2 - 1) * htaps[t * 8 + m];
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ctables[CTABLES - 1 - t][e] = (float)acc;
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}
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}
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}
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#endif /* CONFIG_HARDCODED_TABLES */
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#endif /* AVCODEC_DSD_TABLEGEN_H */
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@ -0,0 +1,167 @@
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/*
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* Direct Stream Digital (DSD) decoder
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* based on BSD licensed dsd2pcm by Sebastian Gesemann
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* Copyright (c) 2009, 2011 Sebastian Gesemann. All rights reserved.
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* Copyright (c) 2014 Peter Ross
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
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* Lesser General Public License for more details.
|
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
|
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* Direct Stream Digital (DSD) decoder
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*/
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#include "libavcodec/internal.h"
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#include "libavcodec/mathops.h"
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#include "avcodec.h"
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#include "dsd_tablegen.h"
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#define FIFOSIZE 16 /** must be a power of two */
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#define FIFOMASK (FIFOSIZE - 1) /** bit mask for FIFO offsets */
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#if FIFOSIZE * 8 < HTAPS * 2
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#error "FIFOSIZE too small"
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#endif
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/**
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* Per-channel buffer
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*/
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typedef struct {
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unsigned char buf[FIFOSIZE];
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unsigned pos;
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} DSDContext;
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static void dsd2pcm_translate(DSDContext* s, size_t samples, int lsbf,
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const unsigned char *src, ptrdiff_t src_stride,
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float *dst, ptrdiff_t dst_stride)
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{
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unsigned pos, i;
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unsigned char* p;
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double sum;
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pos = s->pos;
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while (samples-- > 0) {
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s->buf[pos] = lsbf ? ff_reverse[*src] : *src;
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src += src_stride;
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p = s->buf + ((pos - CTABLES) & FIFOMASK);
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*p = ff_reverse[*p];
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sum = 0.0;
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for (i = 0; i < CTABLES; i++) {
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unsigned char a = s->buf[(pos - i) & FIFOMASK];
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unsigned char b = s->buf[(pos - (CTABLES*2 - 1) + i) & FIFOMASK];
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sum += ctables[i][a] + ctables[i][b];
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}
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*dst = (float)sum;
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dst += dst_stride;
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pos = (pos + 1) & FIFOMASK;
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}
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s->pos = pos;
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}
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static av_cold void init_static_data(void)
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{
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static int done = 0;
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if (done)
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return;
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dsd_ctables_tableinit();
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done = 1;
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}
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static av_cold int decode_init(AVCodecContext *avctx)
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{
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DSDContext * s;
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int i;
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init_static_data();
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s = av_malloc(sizeof(DSDContext) * avctx->channels);
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if (!s)
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return AVERROR(ENOMEM);
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for (i = 0; i < avctx->channels; i++) {
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s[i].pos = 0;
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memset(s[i].buf, 0x69, sizeof(s[i].buf));
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/* 0x69 = 01101001
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* This pattern "on repeat" makes a low energy 352.8 kHz tone
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* and a high energy 1.0584 MHz tone which should be filtered
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* out completely by any playback system --> silence
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*/
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}
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avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
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avctx->priv_data = s;
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return 0;
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}
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static int decode_frame(AVCodecContext *avctx, void *data,
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int *got_frame_ptr, AVPacket *avpkt)
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{
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DSDContext * s = avctx->priv_data;
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AVFrame *frame = data;
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int ret, i;
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int lsbf = avctx->codec_id == AV_CODEC_ID_DSD_LSBF || avctx->codec_id == AV_CODEC_ID_DSD_LSBF_PLANAR;
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int src_next;
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int src_stride;
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frame->nb_samples = avpkt->size / avctx->channels;
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if (avctx->codec_id == AV_CODEC_ID_DSD_LSBF_PLANAR || avctx->codec_id == AV_CODEC_ID_DSD_MSBF_PLANAR) {
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src_next = frame->nb_samples;
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src_stride = 1;
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} else {
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src_next = 1;
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src_stride = avctx->channels;
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}
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if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
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return ret;
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for (i = 0; i < avctx->channels; i++) {
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float * dst = ((float **)frame->extended_data)[i];
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dsd2pcm_translate(&s[i], frame->nb_samples, lsbf,
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avpkt->data + i * src_next, src_stride,
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dst, 1);
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}
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*got_frame_ptr = 1;
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return frame->nb_samples * avctx->channels;
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}
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#define DSD_DECODER(id_, name_, long_name_) \
|
||||
AVCodec ff_##name_##_decoder = { \
|
||||
.name = #name_, \
|
||||
.long_name = NULL_IF_CONFIG_SMALL(long_name_), \
|
||||
.type = AVMEDIA_TYPE_AUDIO, \
|
||||
.id = AV_CODEC_ID_##id_, \
|
||||
.init = decode_init, \
|
||||
.decode = decode_frame, \
|
||||
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP, \
|
||||
AV_SAMPLE_FMT_NONE }, \
|
||||
};
|
||||
|
||||
DSD_DECODER(DSD_LSBF, dsd_lsbf, "DSD (Direct Stream Digital), least significant bit first")
|
||||
DSD_DECODER(DSD_MSBF, dsd_msbf, "DSD (Direct Stream Digital), most significant bit first")
|
||||
DSD_DECODER(DSD_MSBF_PLANAR, dsd_msbf_planar, "DSD (Direct Stream Digital), most significant bit first, planar")
|
||||
DSD_DECODER(DSD_LSBF_PLANAR, dsd_lsbf_planar, "DSD (Direct Stream Digital), least significant bit first, planar")
|
||||
|
|
@ -2994,6 +2994,10 @@ int av_get_exact_bits_per_sample(enum AVCodecID codec_id)
|
|||
case AV_CODEC_ID_ADPCM_G722:
|
||||
case AV_CODEC_ID_ADPCM_YAMAHA:
|
||||
return 4;
|
||||
case AV_CODEC_ID_DSD_LSBF:
|
||||
case AV_CODEC_ID_DSD_MSBF:
|
||||
case AV_CODEC_ID_DSD_LSBF_PLANAR:
|
||||
case AV_CODEC_ID_DSD_MSBF_PLANAR:
|
||||
case AV_CODEC_ID_PCM_ALAW:
|
||||
case AV_CODEC_ID_PCM_MULAW:
|
||||
case AV_CODEC_ID_PCM_S8:
|
||||
|
|
|
|||
Loading…
Reference in New Issue