2 * Direct Stream Digital (DSD) decoder
3 * based on BSD licensed dsd2pcm by Sebastian Gesemann
4 * Copyright (c) 2009, 2011 Sebastian Gesemann. All rights reserved.
5 * Copyright (c) 2014 Peter Ross
7 * This file is part of FFmpeg.
9 * FFmpeg is free software; you can redistribute it and/or
10 * modify it under the terms of the GNU Lesser General Public
11 * License as published by the Free Software Foundation; either
12 * version 2.1 of the License, or (at your option) any later version.
14 * FFmpeg is distributed in the hope that it will be useful,
15 * but WITHOUT ANY WARRANTY; without even the implied warranty of
16 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
17 * Lesser General Public License for more details.
19 * You should have received a copy of the GNU Lesser General Public
20 * License along with FFmpeg; if not, write to the Free Software
21 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
26 * Direct Stream Digital (DSD) decoder
29 #include "libavcodec/internal.h"
30 #include "libavcodec/mathops.h"
34 #define DSD_SILENCE 0x69
36 * This pattern "on repeat" makes a low energy 352.8 kHz tone
37 * and a high energy 1.0584 MHz tone which should be filtered
38 * out completely by any playback system --> silence
41 static av_cold int decode_init(AVCodecContext *avctx)
49 s = av_malloc_array(sizeof(DSDContext), avctx->channels);
51 return AVERROR(ENOMEM);
53 silence = avctx->codec_id == AV_CODEC_ID_DSD_LSBF || avctx->codec_id == AV_CODEC_ID_DSD_LSBF_PLANAR ? ff_reverse[DSD_SILENCE] : DSD_SILENCE;
54 for (i = 0; i < avctx->channels; i++) {
56 memset(s[i].buf, silence, sizeof(s[i].buf));
59 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
64 static int decode_frame(AVCodecContext *avctx, void *data,
65 int *got_frame_ptr, AVPacket *avpkt)
67 DSDContext * s = avctx->priv_data;
68 AVFrame *frame = data;
70 int lsbf = avctx->codec_id == AV_CODEC_ID_DSD_LSBF || avctx->codec_id == AV_CODEC_ID_DSD_LSBF_PLANAR;
74 frame->nb_samples = avpkt->size / avctx->channels;
76 if (avctx->codec_id == AV_CODEC_ID_DSD_LSBF_PLANAR || avctx->codec_id == AV_CODEC_ID_DSD_MSBF_PLANAR) {
77 src_next = frame->nb_samples;
81 src_stride = avctx->channels;
84 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
87 for (i = 0; i < avctx->channels; i++) {
88 float * dst = ((float **)frame->extended_data)[i];
89 ff_dsd2pcm_translate(&s[i], frame->nb_samples, lsbf,
90 avpkt->data + i * src_next, src_stride,
95 return frame->nb_samples * avctx->channels;
98 #define DSD_DECODER(id_, name_, long_name_) \
99 AVCodec ff_##name_##_decoder = { \
101 .long_name = NULL_IF_CONFIG_SMALL(long_name_), \
102 .type = AVMEDIA_TYPE_AUDIO, \
103 .id = AV_CODEC_ID_##id_, \
104 .init = decode_init, \
105 .decode = decode_frame, \
106 .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP, \
107 AV_SAMPLE_FMT_NONE }, \
110 DSD_DECODER(DSD_LSBF, dsd_lsbf, "DSD (Direct Stream Digital), least significant bit first")
111 DSD_DECODER(DSD_MSBF, dsd_msbf, "DSD (Direct Stream Digital), most significant bit first")
112 DSD_DECODER(DSD_MSBF_PLANAR, dsd_msbf_planar, "DSD (Direct Stream Digital), most significant bit first, planar")
113 DSD_DECODER(DSD_LSBF_PLANAR, dsd_lsbf_planar, "DSD (Direct Stream Digital), least significant bit first, planar")