ffmpeg-2.8.5

git-svn-id: svn://kolibrios.org@6147 a494cfbc-eb01-0410-851d-a64ba20cac60

contrib/sdk/sources/ffmpeg/ffmpeg-2.8/libavcodec/dsddec.c

@@ -0,0 +1,167 @@
+/*
+ * 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")