diff options
Diffstat (limited to 'libavcodec/dsddec.c')
| -rw-r--r-- | libavcodec/dsddec.c | 167 |
1 files changed, 167 insertions, 0 deletions
diff --git a/libavcodec/dsddec.c b/libavcodec/dsddec.c new file mode 100644 index 0000000..f1dfd4b --- /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_array(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") |
