/*
  * 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_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 unsigned char *src, ptrdiff_t src_stride,
                               float *dst, ptrdiff_t dst_stride)
 {
     unsigned pos, i;
     unsigned char* 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++) {
             unsigned char a = s->buf[(pos                   - i) & FIFOMASK];
             unsigned char 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 init_static_data(void)
 {
     static int done = 0;
     if (done)
         return;
     dsd_ctables_tableinit();
     done = 1;
 }
 
 static av_cold int decode_init(AVCodecContext *avctx)
 {
     DSDContext * s;
     int i;
 
     init_static_data();
 

     s = av_malloc_array(sizeof(DSDContext), avctx->channels);

     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 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];
         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")