On 16/04/14 16:52, Kostya Shishkov wrote: > On Wed, Apr 16, 2014 at 04:13:19PM +0200, Luca Barbato wrote: >> From: Peter Ross <[email protected]> >> >> Signed-off-by: Peter Ross <[email protected]> >> Signed-off-by: Michael Niedermayer <[email protected]> >> Signed-off-by: Jean-Baptiste Kempf <[email protected]> >> Signed-off-by: Luca Barbato <[email protected]> >> --- >> >> The tests are missing, Vittorio willing to bake some? >> >> Changelog | 1 + >> doc/general.texi | 4 + >> libavcodec/Makefile | 8 +- >> libavcodec/allcodecs.c | 4 + >> libavcodec/avcodec.h | 4 + >> libavcodec/codec_desc.c | 28 +++++++ >> libavcodec/dsd_tablegen.c | 38 ++++++++++ >> libavcodec/dsd_tablegen.h | 95 ++++++++++++++++++++++++ >> libavcodec/dsddec.c | 185 >> ++++++++++++++++++++++++++++++++++++++++++++++ >> libavcodec/utils.c | 4 + >> 10 files changed, 370 insertions(+), 1 deletion(-) >> create mode 100644 libavcodec/dsd_tablegen.c >> create mode 100644 libavcodec/dsd_tablegen.h >> create mode 100644 libavcodec/dsddec.c >> >> diff --git a/Changelog b/Changelog >> index 55216e8..d895b94 100644 >> --- a/Changelog >> +++ b/Changelog >> @@ -19,6 +19,7 @@ version <next>: >> - LucasArts SMUSH demuxer >> - MP2 encoding via TwoLAME >> - asettb filter >> +- Direct Stream Digital (DSD) decoder >> >> >> version 10: >> diff --git a/doc/general.texi b/doc/general.texi >> index 2612eb1..4b16075 100644 >> --- a/doc/general.texi >> +++ b/doc/general.texi >> @@ -822,6 +822,10 @@ 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 > > one item should be enough > >> @item DSP Group TrueSpeech @tab @tab X >> @item DV audio @tab @tab X >> @item Enhanced AC-3 @tab X @tab X > [...] >> diff --git a/libavcodec/dsd_tablegen.h b/libavcodec/dsd_tablegen.h >> new file mode 100644 >> index 0000000..0bebaea >> --- /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 Libav. >> + * >> + * Libav 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. >> + * >> + * Libav 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 Libav; 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 > > barf > >> + * 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); > > unless one decides to switch to non-multiple-of-8 tap filter k=8 in all cases > (otherwise one can pad filter anyway and see above) > >> + 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 0000000..6876f16 >> --- /dev/null >> +++ b/libavcodec/dsddec.c >> @@ -0,0 +1,185 @@ >> +/* >> + * 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 Libav. >> + * >> + * Libav 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. >> + * >> + * Libav 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 Libav; 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]; > > uint8_t? > >> + unsigned pos; >> +} DSDContext; >> + >> +static void dsd2pcm_translate(DSDContext* s, size_t samples, int lsbf, >> + const uint8_t *src, ptrdiff_t src_stride, >> + float *dst, ptrdiff_t dst_stride) >> +{ >> + unsigned int pos, i; >> + uint8_t *p; >> + double sum; >> + >> + pos = s->pos; >> + >> + while (samples-- > 0) { > > while (samples--) > > it's more honest here > >> + 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++) { >> + const uint8_t a = s->buf[(pos - i) & >> FIFOMASK]; >> + const uint8_t 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 dsd_init_static_data(AVCodec *unused) >> +{ >> + dsd_ctables_tableinit(); >> +} >> + >> +static av_cold int dsd_decode_init(AVCodecContext *avctx) >> +{ >> + DSDContext * s; >> + int i; >> + >> + 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 av_always_inline >> +int dsd_decode_frame_internal(AVCodecContext *avctx, void *data, >> + int *got_frame_ptr, AVPacket *avpkt, >> + int lsbf, int planar) >> +{ >> + DSDContext * s = avctx->priv_data; >> + AVFrame *frame = data; >> + int ret, i; >> + int src_next; >> + int src_stride; >> + >> + frame->nb_samples = avpkt->size / avctx->channels; >> + >> + if (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; >> +} >> + >> +static int dsd_lsbf_decode_frame(AVCodecContext *avctx, void *data, >> + int *got_frame, AVPacket *avpkt) >> +{ >> + return dsd_decode_frame_internal(avctx, data, got_frame, avpkt, 1, 0); >> +} > > empty line missing > >> +static int dsd_msbf_decode_frame(AVCodecContext *avctx, void *data, >> + int *got_frame, AVPacket *avpkt) >> +{ >> + return dsd_decode_frame_internal(avctx, data, got_frame, avpkt, 0, 0); >> +} >> + >> +static int dsd_lsbf_planar_decode_frame(AVCodecContext *avctx, void *data, >> + int *got_frame, AVPacket *avpkt) >> +{ >> + return dsd_decode_frame_internal(avctx, data, got_frame, avpkt, 1, 1); >> +} >> +static int dsd_msbf_planar_decode_frame(AVCodecContext *avctx, void *data, >> + int *got_frame, AVPacket *avpkt) >> +{ >> + return dsd_decode_frame_internal(avctx, data, got_frame, avpkt, 0, 1); >> +} >> + >> +#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 = dsd_decode_init, \ >> + .decode = name_ ## _decode_frame, \ >> + .init_static_data = dsd_init_static_data, \ >> + .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") > > I wonder if it makes sense to contract it to "LSB/MSB first" > >> diff --git a/libavcodec/utils.c b/libavcodec/utils.c >> index d9832e2..38a138b 100644 >> --- a/libavcodec/utils.c >> +++ b/libavcodec/utils.c >> @@ -1929,6 +1929,10 @@ 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: >> -- > > in general LGTM
Everything mentioned fixed, shall I push it or wait for jb to source the demuxer bits? lu _______________________________________________ libav-devel mailing list [email protected] https://lists.libav.org/mailman/listinfo/libav-devel
