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111 lines
3.6 KiB
C
111 lines
3.6 KiB
C
/*
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* Direct Stream Digital (DSD) decoder
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* based on BSD licensed dsd2pcm by Sebastian Gesemann
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* Copyright (c) 2009, 2011 Sebastian Gesemann. All rights reserved.
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* Copyright (c) 2014 Peter Ross
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* Direct Stream Digital (DSD) decoder
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*/
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#include "libavcodec/internal.h"
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#include "libavcodec/mathops.h"
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#include "avcodec.h"
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#include "dsd.h"
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static av_cold int decode_init(AVCodecContext *avctx)
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{
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DSDContext * s;
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int i;
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ff_init_dsd_data();
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s = av_malloc_array(sizeof(DSDContext), avctx->channels);
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if (!s)
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return AVERROR(ENOMEM);
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for (i = 0; i < avctx->channels; i++) {
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s[i].pos = 0;
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memset(s[i].buf, 0x69, sizeof(s[i].buf));
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/* 0x69 = 01101001
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* This pattern "on repeat" makes a low energy 352.8 kHz tone
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* and a high energy 1.0584 MHz tone which should be filtered
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* out completely by any playback system --> silence
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*/
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}
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avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
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avctx->priv_data = s;
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return 0;
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}
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static int decode_frame(AVCodecContext *avctx, void *data,
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int *got_frame_ptr, AVPacket *avpkt)
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{
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DSDContext * s = avctx->priv_data;
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AVFrame *frame = data;
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int ret, i;
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int lsbf = avctx->codec_id == AV_CODEC_ID_DSD_LSBF || avctx->codec_id == AV_CODEC_ID_DSD_LSBF_PLANAR;
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int src_next;
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int src_stride;
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frame->nb_samples = avpkt->size / avctx->channels;
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if (avctx->codec_id == AV_CODEC_ID_DSD_LSBF_PLANAR || avctx->codec_id == AV_CODEC_ID_DSD_MSBF_PLANAR) {
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src_next = frame->nb_samples;
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src_stride = 1;
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} else {
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src_next = 1;
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src_stride = avctx->channels;
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}
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if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
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return ret;
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for (i = 0; i < avctx->channels; i++) {
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float * dst = ((float **)frame->extended_data)[i];
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ff_dsd2pcm_translate(&s[i], frame->nb_samples, lsbf,
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avpkt->data + i * src_next, src_stride,
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dst, 1);
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}
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*got_frame_ptr = 1;
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return frame->nb_samples * avctx->channels;
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}
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#define DSD_DECODER(id_, name_, long_name_) \
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AVCodec ff_##name_##_decoder = { \
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.name = #name_, \
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.long_name = NULL_IF_CONFIG_SMALL(long_name_), \
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.type = AVMEDIA_TYPE_AUDIO, \
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.id = AV_CODEC_ID_##id_, \
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.init = decode_init, \
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.decode = decode_frame, \
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.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP, \
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AV_SAMPLE_FMT_NONE }, \
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};
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DSD_DECODER(DSD_LSBF, dsd_lsbf, "DSD (Direct Stream Digital), least significant bit first")
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DSD_DECODER(DSD_MSBF, dsd_msbf, "DSD (Direct Stream Digital), most significant bit first")
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DSD_DECODER(DSD_MSBF_PLANAR, dsd_msbf_planar, "DSD (Direct Stream Digital), most significant bit first, planar")
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DSD_DECODER(DSD_LSBF_PLANAR, dsd_lsbf_planar, "DSD (Direct Stream Digital), least significant bit first, planar")
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