On Wed, 16 Apr 2014 16:13:19 +0200, Luca Barbato <[email protected]> wrote:
> 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];
> + 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) {
> + 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);
Eeeew.
This looks really horrible, please change it to something sane.
> + 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);
> +}
> +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 },
> \
Missing DR cap.
--
Anton Khirnov
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