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submodules/ffmpeg/Sources/FFMpeg/ffmpeg-4.1/libavcodec/dsddec.c

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+/*
+ * 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.h"
+
+#define DSD_SILENCE 0x69
+/* 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;
+
+    ff_init_dsd_data();
+
+    s = av_malloc_array(sizeof(DSDContext), avctx->channels);
+    if (!s)
+        return AVERROR(ENOMEM);
+
+    silence = avctx->codec_id == AV_CODEC_ID_DSD_LSBF || avctx->codec_id == AV_CODEC_ID_DSD_LSBF_PLANAR ? ff_reverse[DSD_SILENCE] : DSD_SILENCE;
+    for (i = 0; i < avctx->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;
+}
+
+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];
+        ff_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")