mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-07 11:13:41 +02:00
4243da4ff4
This is possible, because every given FFCodec has to implement exactly one of these. Doing so decreases sizeof(FFCodec) and therefore decreases the size of the binary. Notice that in case of position-independent code the decrease is in .data.rel.ro, so that this translates to decreased memory consumption. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
135 lines
4.5 KiB
C
135 lines
4.5 KiB
C
/*
|
|
* 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 "avcodec.h"
|
|
#include "codec_internal.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_, \
|
|
.p.long_name = NULL_IF_CONFIG_SMALL(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 }, \
|
|
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE, \
|
|
};
|
|
|
|
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")
|