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FFMpeg에서 DOP (DSD-over-PCM) 구현 패치

by 엉뚱도마뱀 posted Dec 20, 2018
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Signed-off-by: Peter Ross <pross at xvid.org>
---
 Changelog               |   1 +
 doc/general.texi        |   1 +
 libavcodec/Makefile     |   2 +
 libavcodec/allcodecs.c  |   1 +
 libavcodec/avcodec.h    |   1 +
 libavcodec/codec_desc.c |   7 ++
 libavcodec/dop.c        | 172 ++++++++++++++++++++++++++++++++++++++++++++++++
 libavcodec/utils.c      |   1 +
 8 files changed, 186 insertions(+)
 create mode 100644 libavcodec/dop.c

diff --git a/Changelog b/Changelog
index 8d61c9d..cbc30d0 100644
--- a/Changelog
+++ b/Changelog
@@ -22,6 +22,7 @@ version <next>:
 - On2 AVC (Audio for Video) decoder
 - support for decoding through DXVA2 in ffmpeg
 - libbs2b-based stereo-to-binaural audio filter
+- DoP (DSD-over-PCM)
 
 
 version 2.2:
diff --git a/doc/general.texi b/doc/general.texi
index 8a73cbf..99dc01c 100644
--- a/doc/general.texi
+++ b/doc/general.texi
@@ -891,6 +891,7 @@ following image formats are supported:
 @item COOK                   @tab     @tab  X
     @tab All versions except 5.1 are supported.
 @item DCA (DTS Coherent Acoustics)  @tab  X  @tab  X
+ at item DoP (DSD-over-PCM); signed 24-bit little-endian  @tab  X  @tab  X
 @item DPCM id RoQ            @tab  X  @tab  X
     @tab Used in Quake III, Jedi Knight 2 and other computer games.
 @item DPCM Interplay         @tab     @tab  X
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 7c475fb..433e438 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -177,6 +177,8 @@ OBJS-$(CONFIG_DIRAC_DECODER)           += diracdec.o dirac.o diracdsp.o \
 OBJS-$(CONFIG_DFA_DECODER)             += dfa.o
 OBJS-$(CONFIG_DNXHD_DECODER)           += dnxhddec.o dnxhddata.o
 OBJS-$(CONFIG_DNXHD_ENCODER)           += dnxhdenc.o dnxhddata.o
+OBJS-$(CONFIG_DOP_S24LE_DECODER)       += dop.o
+OBJS-$(CONFIG_DOP_S24LE_ENCODER)       += dop.o
 OBJS-$(CONFIG_DPX_DECODER)             += dpx.o
 OBJS-$(CONFIG_DPX_ENCODER)             += dpxenc.o
 OBJS-$(CONFIG_DSD_LSBF_DECODER)        += dsddec.o
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index eb6b0c3..c1bb3db 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -337,6 +337,7 @@ void avcodec_register_all(void)
     REGISTER_DECODER(BMV_AUDIO,         bmv_audio);
     REGISTER_DECODER(COOK,              cook);
     REGISTER_ENCDEC (DCA,               dca);
+    REGISTER_ENCDEC (DOP_S24LE,         dop_s24le);
     REGISTER_ENCDEC (DSD_LSBF,          dsd_lsbf);
     REGISTER_ENCDEC (DSD_MSBF,          dsd_msbf);
     REGISTER_ENCDEC (DSD_LSBF_PLANAR,   dsd_lsbf_planar);
diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index 94d4cf0..488a521 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -497,6 +497,7 @@ enum AVCodecID {
     AV_CODEC_ID_DSD_MSBF    = MKBETAG('D','S','D','M'),
     AV_CODEC_ID_DSD_LSBF_PLANAR = MKBETAG('D','S','D','1'),
     AV_CODEC_ID_DSD_MSBF_PLANAR = MKBETAG('D','S','D','8'),
+    AV_CODEC_ID_DOP_S24LE   = MKBETAG('D','O','P','L'),
 
     /* subtitle codecs */
     AV_CODEC_ID_FIRST_SUBTITLE = 0x17000,          ///< A dummy ID pointing at the start of subtitle codecs.
diff --git a/libavcodec/codec_desc.c b/libavcodec/codec_desc.c
index 057c6a3..3b474f8 100644
--- a/libavcodec/codec_desc.c
+++ b/libavcodec/codec_desc.c
@@ -2508,6 +2508,13 @@ static const AVCodecDescriptor codec_descriptors[] = {
         .long_name = NULL_IF_CONFIG_SMALL("DSD (Direct Stream Digital), most significant bit first, planar"),
         .props     = AV_CODEC_PROP_LOSSLESS,
     },
+    {
+        .id        = AV_CODEC_ID_DOP_S24LE,
+        .type      = AVMEDIA_TYPE_AUDIO,
+        .name      = "dop_s24le",
+        .long_name = NULL_IF_CONFIG_SMALL("DoP (DSD-over-PCM); signed 24-bit little-endian"),
+        .props     = AV_CODEC_PROP_LOSSLESS,
+    },
 
     /* subtitle codecs */
     {
diff --git a/libavcodec/dop.c b/libavcodec/dop.c
new file mode 100644
index 0000000..0c04add
--- /dev/null
+++ b/libavcodec/dop.c
@@ -0,0 +1,172 @@
+/*
+ * DSD-over-PCM (DOP) encoder and decoder
+ * 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
+ * DSD-over-PCM (DOP) encoder and decoder
+ */
+
+#include "libavcodec/internal.h"
+#include "bytestream.h"
+#include "avcodec.h"
+
+typedef struct {
+    int state;
+    int sync;
+} DOPContext;
+
+static const uint8_t dop_marker[2] = {0x05, 0xFA};
+
+static av_cold int decode_init(AVCodecContext *avctx)
+{
+    DOPContext * s = avctx->priv_data;
+    avctx->sample_fmt = AV_SAMPLE_FMT_DSD;
+
+    switch(avctx->sample_rate) {
+    case 176400:
+    case 352800:
+       avctx->sample_rate *= 16;
+       break;
+    case 2822400:
+    case 5644800:
+       break;
+    default:
+       av_log(avctx, AV_LOG_WARNING, "unsupported sample rate\n");
+    }
+    s->sync = 32 * avctx->channels;
+    return 0;
+}
+
+static int decode_frame(AVCodecContext *avctx, void *data,
+                        int *got_frame_ptr, AVPacket *avpkt)
+{
+    DOPContext * s = avctx->priv_data;
+    const uint8_t *src = avpkt->data;
+    uint8_t *dst;
+    AVFrame *frame = data;
+    int ret, i, ch;
+    int pcm_samples = avpkt->size / (3 * avctx->channels);
+    frame->nb_samples = 2 * pcm_samples;
+    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
+        return ret;
+
+    dst = frame->data[0];
+
+    for (i = 0; i < pcm_samples; i++) {
+        for (ch = 0; ch < avctx->channels; ch++) {
+            uint8_t d1     = src[0];
+            uint8_t d0     = src[1];
+
+            uint8_t marker = src[2];
+            if (!s->sync) {
+                if (marker == 0x05 || marker == 0xFA) {
+                    s->state = marker == 0xFA;
+                    s->sync++;
+                }
+                d0 = d1 = 0x69;
+            } else {
+                if (marker != dop_marker[s->state]) { /* lost sync */
+                    s->sync = 0;
+                    d0 = d1 = 0x69;
+                } else if (s->sync < 32 * avctx->channels) { /* resynching */
+                    s->sync++;
+                    d0 = d1 = 0x69;
+                }
+            }
+            src += 3;
+
+            dst[0]               = d0;
+            dst[avctx->channels] = d1;
+            dst++;
+        }
+        dst += avctx->channels;
+        s->state ^= 1;
+    }
+
+    *got_frame_ptr = 1;
+    return pcm_samples * 3 * avctx->channels;
+}
+
+AVCodec ff_dop_s24le_decoder = {
+    .name           = "dop_s24le",
+    .priv_data_size = sizeof(DOPContext),
+    .long_name      = NULL_IF_CONFIG_SMALL("DoP (DSD-over-PCM); signed 24-bit little-endian"),
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = AV_CODEC_ID_DOP_S24LE,
+    .init           = decode_init,
+    .decode         = decode_frame,
+    .sample_fmts    = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_DSD,
+                                                    AV_SAMPLE_FMT_NONE },
+};
+
+static av_cold int encode_init(AVCodecContext *avctx)
+{
+    avctx->bits_per_coded_sample = av_get_bits_per_sample(avctx->codec->id);
+    avctx->block_align           = avctx->channels * avctx->bits_per_coded_sample / 8;
+    avctx->bit_rate              = avctx->block_align * avctx->sample_rate * 8;
+    avctx->sample_rate          /= 16;
+    return 0;
+}
+
+static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
+                            const AVFrame *frame, int *got_packet_ptr)
+{
+    DOPContext * s = avctx->priv_data;
+    int ret, i, ch;
+    const uint8_t *src;
+    uint8_t *dst;
+
+    if (frame->nb_samples % 2)
+        av_log(avctx, AV_LOG_WARNING, "expected mod 16 samples\n");
+
+    if ((ret = ff_alloc_packet2(avctx, avpkt, ((frame->nb_samples) / 2) * 3 * avctx->channels)) < 0)
+        return ret;
+
+    src = frame->data[0];
+    dst = avpkt->data;
+    for (i = 0; i < frame->nb_samples; i += 2) {
+        for (ch = 0; ch < avctx->channels; ch++) {
+            dst[0] = src[avctx->channels];
+            dst[1] = src[0];
+            dst[2] = dop_marker[s->state];
+            src++;
+            dst += 3;
+        }
+        src += avctx->channels;
+        s->state ^= 1;
+    }
+
+    *got_packet_ptr = 1;
+    return 0;
+}
+
+AVCodec ff_dop_s24le_encoder = {
+    .name           = "dop_s24le",
+    .priv_data_size = sizeof(DOPContext),
+    .long_name      = NULL_IF_CONFIG_SMALL("DoP (DSD-over-PCM); signed 24-bit little-endian"),
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = AV_CODEC_ID_DOP_S24LE,
+    .init           = encode_init,
+    .encode2        = encode_frame,
+    .capabilities   = CODEC_CAP_VARIABLE_FRAME_SIZE,
+    .sample_fmts    = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_DSD,
+                                                    AV_SAMPLE_FMT_NONE},
+};
diff --git a/libavcodec/utils.c b/libavcodec/utils.c
index 8bb2579..593db69 100644
--- a/libavcodec/utils.c
+++ b/libavcodec/utils.c
@@ -3007,6 +3007,7 @@ int av_get_exact_bits_per_sample(enum AVCodecID codec_id)
     case AV_CODEC_ID_PCM_U16BE:
     case AV_CODEC_ID_PCM_U16LE:
         return 16;
+    case AV_CODEC_ID_DOP_S24LE:
     case AV_CODEC_ID_PCM_S24DAUD:
     case AV_CODEC_ID_PCM_S24BE:
     case AV_CODEC_ID_PCM_S24LE:
-- 
1.8.3.2

-- Peter
(A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B)
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