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/*
* 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.h"
#define DSD_SILENCE 0x69
/* 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
*/
static av_cold int decode_init(AVCodecContext *avctx)
{
DSDContext * s;
int i;
uint8_t silence;
if (!avctx->channels)
return AVERROR_INVALIDDATA;
ff_init_dsd_data();
s = av_malloc_array(sizeof(DSDContext), avctx->channels);
if (!s)
return AVERROR(ENOMEM);
silence = avctx->codec_id == AV_CODEC_ID_DSD_LSBF || avctx->codec_id == AV_CODEC_ID_DSD_LSBF_PLANAR ? ff_reverse[DSD_SILENCE] : DSD_SILENCE;
for (i = 0; i < avctx->channels; i++) {
s[i].pos = 0;
memset(s[i].buf, silence, sizeof(s[i].buf));
}
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];
ff_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")
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