Direct Stream Digital (DSD) decoder

Signed-off-by: Peter Ross <pross@xvid.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

8 files changed, 345 insertions, 1 deletions

diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 11e0ebd5f9..d803ec3628 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -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
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index b948848a05..456d702950 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -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);
diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index 17bde9f517..a7bf52a1b7 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -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.
diff --git a/libavcodec/codec_desc.c b/libavcodec/codec_desc.c
index 7ce4be0421..e742b5d030 100644
--- a/libavcodec/codec_desc.c
+++ b/libavcodec/codec_desc.c
@@ -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 */
     {
diff --git a/libavcodec/dsd_tablegen.c b/libavcodec/dsd_tablegen.c
new file mode 100644
index 0000000000..dbeb9fe205
--- /dev/null
+++ b/libavcodec/dsd_tablegen.c
@@ -0,0 +1,38 @@
+/*
+ * 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;
+}
diff --git a/libavcodec/dsd_tablegen.h b/libavcodec/dsd_tablegen.h
new file mode 100644
index 0000000000..6afb4167da
--- /dev/null
+++ b/libavcodec/dsd_tablegen.h
@@ -0,0 +1,95 @@
+/*
+ * 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 */
diff --git a/libavcodec/dsddec.c b/libavcodec/dsddec.c
new file mode 100644
index 0000000000..1782d078e5
--- /dev/null
+++ b/libavcodec/dsddec.c
@@ -0,0 +1,167 @@
+/*
+ * 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")
diff --git a/libavcodec/utils.c b/libavcodec/utils.c
index d14ab4c60e..24319cc2e9 100644
--- a/libavcodec/utils.c
+++ b/libavcodec/utils.c
@@ -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: