/* * 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 "libavutil/mem.h" #include "avcodec.h" #include "codec_internal.h" #include "decode.h" #include "dsd.h" #define DSD_SILENCE 0x69 #define DSD_SILENCE_REVERSED 0x96 /* 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->ch_layout.nb_channels) return AVERROR_INVALIDDATA; ff_init_dsd_data(); s = av_malloc_array(sizeof(DSDContext), avctx->ch_layout.nb_channels); if (!s) return AVERROR(ENOMEM); silence = avctx->codec_id == AV_CODEC_ID_DSD_LSBF || avctx->codec_id == AV_CODEC_ID_DSD_LSBF_PLANAR ? DSD_SILENCE_REVERSED : DSD_SILENCE; for (i = 0; i < avctx->ch_layout.nb_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; } typedef struct ThreadData { AVFrame *frame; const AVPacket *avpkt; } ThreadData; static int dsd_channel(AVCodecContext *avctx, void *tdata, int j, int threadnr) { int lsbf = avctx->codec_id == AV_CODEC_ID_DSD_LSBF || avctx->codec_id == AV_CODEC_ID_DSD_LSBF_PLANAR; DSDContext *s = avctx->priv_data; ThreadData *td = tdata; AVFrame *frame = td->frame; const AVPacket *avpkt = td->avpkt; int src_next, src_stride; float *dst = ((float **)frame->extended_data)[j]; 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->ch_layout.nb_channels; } ff_dsd2pcm_translate(&s[j], frame->nb_samples, lsbf, avpkt->data + j * src_next, src_stride, dst, 1); return 0; } static int decode_frame(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, AVPacket *avpkt) { ThreadData td; int ret; frame->nb_samples = avpkt->size / avctx->ch_layout.nb_channels; if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) return ret; td.frame = frame; td.avpkt = avpkt; avctx->execute2(avctx, dsd_channel, &td, NULL, avctx->ch_layout.nb_channels); *got_frame_ptr = 1; return frame->nb_samples * avctx->ch_layout.nb_channels; } #define DSD_DECODER(id_, name_, long_name_) \ const FFCodec ff_ ## name_ ## _decoder = { \ .p.name = #name_, \ CODEC_LONG_NAME(long_name_), \ .p.type = AVMEDIA_TYPE_AUDIO, \ .p.id = AV_CODEC_ID_##id_, \ .init = decode_init, \ FF_CODEC_DECODE_CB(decode_frame), \ .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_SLICE_THREADS, \ .p.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")