Commit 5f4f9ee9 authored by Peter Ross's avatar Peter Ross Committed by Michael Niedermayer

Direct Stream Digital (DSD) decoder

Signed-off-by: 's avatarPeter Ross <pross@xvid.org>
Signed-off-by: 's avatarMichael Niedermayer <michaelni@gmx.at>
parent add5d7a8
......@@ -17,6 +17,7 @@ version <next>:
- GDI screen grabbing for Windows
- alternative rendition support for HTTP Live Streaming
- AVFoundation input device
- Direct Stream Digital (DSD) decoder
version 2.2:
......
......@@ -898,6 +898,10 @@ following image formats are supported:
@item DPCM Sol @tab @tab X
@item DPCM Xan @tab @tab X
@tab Used in Origin's Wing Commander IV AVI files.
@item DSD (Direct Stream Digitial), least significant bit first @tab @tab X
@item DSD (Direct Stream Digitial), most significant bit first @tab @tab X
@item DSD (Direct Stream Digitial), least significant bit first, planar @tab @tab X
@item DSD (Direct Stream Digitial), most significant bit first, planar @tab @tab X
@item DSP Group TrueSpeech @tab @tab X
@item DV audio @tab @tab X
@item Enhanced AC-3 @tab X @tab X
......
......@@ -179,6 +179,8 @@ OBJS-$(CONFIG_DNXHD_DECODER) += dnxhddec.o dnxhddata.o
OBJS-$(CONFIG_DNXHD_ENCODER) += dnxhdenc.o dnxhddata.o
OBJS-$(CONFIG_DPX_DECODER) += dpx.o
OBJS-$(CONFIG_DPX_ENCODER) += dpxenc.o
OBJS-$(CONFIG_DSD_LSBF_DECODER) += dsddec.o
OBJS-$(CONFIG_DSD_MSBF_DECODER) += dsddec.o
OBJS-$(CONFIG_DSICINAUDIO_DECODER) += dsicinav.o
OBJS-$(CONFIG_DSICINVIDEO_DECODER) += dsicinav.o
OBJS-$(CONFIG_DVBSUB_DECODER) += dvbsubdec.o
......@@ -847,6 +849,7 @@ HOSTPROGS = aac_tablegen \
aacps_tablegen \
cbrt_tablegen \
cos_tablegen \
dsd_tablegen \
dv_tablegen \
motionpixels_tablegen \
mpegaudio_tablegen \
......@@ -871,7 +874,7 @@ else
$(SUBDIR)%_tablegen$(HOSTEXESUF): HOSTCFLAGS += -DCONFIG_SMALL=0
endif
GEN_HEADERS = cbrt_tables.h aacps_tables.h aac_tables.h dv_tables.h \
GEN_HEADERS = cbrt_tables.h aacps_tables.h aac_tables.h dsd_tables.h dv_tables.h \
sinewin_tables.h mpegaudio_tables.h motionpixels_tables.h \
pcm_tables.h qdm2_tables.h
GEN_HEADERS := $(addprefix $(SUBDIR), $(GEN_HEADERS))
......@@ -883,6 +886,7 @@ ifdef CONFIG_HARDCODED_TABLES
$(SUBDIR)aacdec.o: $(SUBDIR)cbrt_tables.h
$(SUBDIR)aacps.o: $(SUBDIR)aacps_tables.h
$(SUBDIR)aactab.o: $(SUBDIR)aac_tables.h
$(SUBDIR)dsddec.o: $(SUBDIR)dsd_tables.h
$(SUBDIR)dvenc.o: $(SUBDIR)dv_tables.h
$(SUBDIR)sinewin.o: $(SUBDIR)sinewin_tables.h
$(SUBDIR)mpegaudiodec_fixed.o: $(SUBDIR)mpegaudio_tables.h
......
......@@ -337,6 +337,10 @@ void avcodec_register_all(void)
REGISTER_DECODER(BMV_AUDIO, bmv_audio);
REGISTER_DECODER(COOK, cook);
REGISTER_ENCDEC (DCA, dca);
REGISTER_DECODER(DSD_LSBF, dsd_lsbf);
REGISTER_DECODER(DSD_MSBF, dsd_msbf);
REGISTER_DECODER(DSD_LSBF_PLANAR, dsd_lsbf_planar);
REGISTER_DECODER(DSD_MSBF_PLANAR, dsd_msbf_planar);
REGISTER_DECODER(DSICINAUDIO, dsicinaudio);
REGISTER_ENCDEC (EAC3, eac3);
REGISTER_DECODER(EVRC, evrc);
......
......@@ -489,6 +489,10 @@ enum AVCodecID {
AV_CODEC_ID_TAK = MKBETAG('t','B','a','K'),
AV_CODEC_ID_EVRC = MKBETAG('s','e','v','c'),
AV_CODEC_ID_SMV = MKBETAG('s','s','m','v'),
AV_CODEC_ID_DSD_LSBF = MKBETAG('D','S','D','L'),
AV_CODEC_ID_DSD_MSBF = MKBETAG('D','S','D','M'),
AV_CODEC_ID_DSD_LSBF_PLANAR = MKBETAG('D','S','D','1'),
AV_CODEC_ID_DSD_MSBF_PLANAR = MKBETAG('D','S','D','8'),
/* subtitle codecs */
AV_CODEC_ID_FIRST_SUBTITLE = 0x17000, ///< A dummy ID pointing at the start of subtitle codecs.
......
......@@ -2460,6 +2460,34 @@ static const AVCodecDescriptor codec_descriptors[] = {
.long_name = NULL_IF_CONFIG_SMALL("SMV (Selectable Mode Vocoder)"),
.props = AV_CODEC_PROP_LOSSY,
},
{
.id = AV_CODEC_ID_DSD_LSBF,
.type = AVMEDIA_TYPE_AUDIO,
.name = "dsd_lsbf",
.long_name = NULL_IF_CONFIG_SMALL("DSD (Direct Stream Digital), least significant bit first"),
.props = AV_CODEC_PROP_LOSSY,
},
{
.id = AV_CODEC_ID_DSD_MSBF,
.type = AVMEDIA_TYPE_AUDIO,
.name = "dsd_msbf",
.long_name = NULL_IF_CONFIG_SMALL("DSD (Direct Stream Digital), most significant bit first"),
.props = AV_CODEC_PROP_LOSSY,
},
{
.id = AV_CODEC_ID_DSD_LSBF_PLANAR,
.type = AVMEDIA_TYPE_AUDIO,
.name = "dsd_lsbf_planar",
.long_name = NULL_IF_CONFIG_SMALL("DSD (Direct Stream Digital), least significant bit first, planar"),
.props = AV_CODEC_PROP_LOSSY,
},
{
.id = AV_CODEC_ID_DSD_MSBF_PLANAR,
.type = AVMEDIA_TYPE_AUDIO,
.name = "dsd_msbf_planar",
.long_name = NULL_IF_CONFIG_SMALL("DSD (Direct Stream Digital), most significant bit first, planar"),
.props = AV_CODEC_PROP_LOSSY,
},
/* subtitle codecs */
{
......
/*
* Generate a header file for hardcoded DSD tables
*
* 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 <stdlib.h>
#define CONFIG_HARDCODED_TABLES 0
#include "dsd_tablegen.h"
#include "tableprint.h"
#include <inttypes.h>
int main(void)
{
dsd_ctables_tableinit();
write_fileheader();
printf("static const double ctables[CTABLES][256] = {\n");
write_float_2d_array(ctables, CTABLES, 256);
printf("};\n");
return 0;
}
/*
* Header file for hardcoded DSD tables
* based on BSD licensed dsd2pcm by Sebastian Gesemann
* Copyright (c) 2009, 2011 Sebastian Gesemann. All rights reserved.
*
* 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_DSD_TABLEGEN_H
#define AVCODEC_DSD_TABLEGEN_H
#include <stdint.h>
#include "libavutil/attributes.h"
#define HTAPS 48 /** number of FIR constants */
#define CTABLES ((HTAPS + 7) / 8) /** number of "8 MACs" lookup tables */
#if CONFIG_HARDCODED_TABLES
#define dsd_ctables_tableinit()
#include "libavcodec/dsd_tables.h"
#else
#include "libavutil/common.h"
/*
* Properties of this 96-tap lowpass filter when applied on a signal
* with sampling rate of 44100*64 Hz:
*
* () has a delay of 17 microseconds.
*
* () flat response up to 48 kHz
*
* () if you downsample afterwards by a factor of 8, the
* spectrum below 70 kHz is practically alias-free.
*
* () stopband rejection is about 160 dB
*
* The coefficient tables ("ctables") take only 6 Kibi Bytes and
* should fit into a modern processor's fast cache.
*/
/**
* The 2nd half (48 coeffs) of a 96-tap symmetric lowpass filter
*/
static const double htaps[HTAPS] = {
0.09950731974056658, 0.09562845727714668, 0.08819647126516944,
0.07782552527068175, 0.06534876523171299, 0.05172629311427257,
0.0379429484910187, 0.02490921351762261, 0.0133774746265897,
0.003883043418804416, -0.003284703416210726, -0.008080250212687497,
-0.01067241812471033, -0.01139427235000863, -0.0106813877974587,
-0.009007905078766049, -0.006828859761015335, -0.004535184322001496,
-0.002425035959059578, -0.0006922187080790708, 0.0005700762133516592,
0.001353838005269448, 0.001713709169690937, 0.001742046839472948,
0.001545601648013235, 0.001226696225277855, 0.0008704322683580222,
0.0005381636200535649, 0.000266446345425276, 7.002968738383528e-05,
-5.279407053811266e-05, -0.0001140625650874684, -0.0001304796361231895,
-0.0001189970287491285, -9.396247155265073e-05, -6.577634378272832e-05,
-4.07492895872535e-05, -2.17407957554587e-05, -9.163058931391722e-06,
-2.017460145032201e-06, 1.249721855219005e-06, 2.166655190537392e-06,
1.930520892991082e-06, 1.319400334374195e-06, 7.410039764949091e-07,
3.423230509967409e-07, 1.244182214744588e-07, 3.130441005359396e-08
};
static float ctables[CTABLES][256];
static av_cold void dsd_ctables_tableinit(void)
{
int t, e, m, k;
double acc;
for (t = 0; t < CTABLES; ++t) {
k = FFMIN(HTAPS - t * 8, 8);
for (e = 0; e < 256; ++e) {
acc = 0.0;
for (m = 0; m < k; ++m)
acc += (((e >> (7 - m)) & 1) * 2 - 1) * htaps[t * 8 + m];
ctables[CTABLES - 1 - t][e] = (float)acc;
}
}
}
#endif /* CONFIG_HARDCODED_TABLES */
#endif /* AVCODEC_DSD_TABLEGEN_H */
/*
* Direct Stream Digital (DSD) decoder
* based on BSD licensed dsd2pcm by Sebastian Gesemann
* Copyright (c) 2009, 2011 Sebastian Gesemann. All rights reserved.
* Copyright (c) 2014 Peter Ross
*
* 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
*/
/**
* @file
* Direct Stream Digital (DSD) decoder
*/
#include "libavcodec/internal.h"
#include "libavcodec/mathops.h"
#include "avcodec.h"
#include "dsd_tablegen.h"
#define FIFOSIZE 16 /** must be a power of two */
#define FIFOMASK (FIFOSIZE - 1) /** bit mask for FIFO offsets */
#if FIFOSIZE * 8 < HTAPS * 2
#error "FIFOSIZE too small"
#endif
/**
* Per-channel buffer
*/
typedef struct {
unsigned char buf[FIFOSIZE];
unsigned pos;
} DSDContext;
static void dsd2pcm_translate(DSDContext* s, size_t samples, int lsbf,
const unsigned char *src, ptrdiff_t src_stride,
float *dst, ptrdiff_t dst_stride)
{
unsigned pos, i;
unsigned char* p;
double sum;
pos = s->pos;
while (samples-- > 0) {
s->buf[pos] = lsbf ? ff_reverse[*src] : *src;
src += src_stride;
p = s->buf + ((pos - CTABLES) & FIFOMASK);
*p = ff_reverse[*p];
sum = 0.0;
for (i = 0; i < CTABLES; i++) {
unsigned char a = s->buf[(pos - i) & FIFOMASK];
unsigned char b = s->buf[(pos - (CTABLES*2 - 1) + i) & FIFOMASK];
sum += ctables[i][a] + ctables[i][b];
}
*dst = (float)sum;
dst += dst_stride;
pos = (pos + 1) & FIFOMASK;
}
s->pos = pos;
}
static av_cold void init_static_data(void)
{
static int done = 0;
if (done)
return;
dsd_ctables_tableinit();
done = 1;
}
static av_cold int decode_init(AVCodecContext *avctx)
{
DSDContext * s;
int i;
init_static_data();
s = av_malloc(sizeof(DSDContext) * avctx->channels);
if (!s)
return AVERROR(ENOMEM);
for (i = 0; i < avctx->channels; i++) {
s[i].pos = 0;
memset(s[i].buf, 0x69, sizeof(s[i].buf));
/* 0x69 = 01101001
* This pattern "on repeat" makes a low energy 352.8 kHz tone
* and a high energy 1.0584 MHz tone which should be filtered
* out completely by any playback system --> silence
*/
}
avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
avctx->priv_data = s;
return 0;
}
static int decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
DSDContext * s = avctx->priv_data;
AVFrame *frame = data;
int ret, i;
int lsbf = avctx->codec_id == AV_CODEC_ID_DSD_LSBF || avctx->codec_id == AV_CODEC_ID_DSD_LSBF_PLANAR;
int src_next;
int src_stride;
frame->nb_samples = avpkt->size / avctx->channels;
if (avctx->codec_id == AV_CODEC_ID_DSD_LSBF_PLANAR || avctx->codec_id == AV_CODEC_ID_DSD_MSBF_PLANAR) {
src_next = frame->nb_samples;
src_stride = 1;
} else {
src_next = 1;
src_stride = avctx->channels;
}
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
for (i = 0; i < avctx->channels; i++) {
float * dst = ((float **)frame->extended_data)[i];
dsd2pcm_translate(&s[i], frame->nb_samples, lsbf,
avpkt->data + i * src_next, src_stride,
dst, 1);
}
*got_frame_ptr = 1;
return frame->nb_samples * avctx->channels;
}
#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:
......
Markdown is supported
0% or
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment