mirror of
https://git.ffmpeg.org/ffmpeg.git
synced 2025-01-30 11:24:17 +00:00
avcodec: add WavArc decoder
This commit is contained in:
Paul B Mahol
parent
9a820ec8b1
commit
651da91915
@ -37,6 +37,7 @@ version <next>:
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- XMD ADPCM decoder and demuxer
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- media100 to mjpegb bsf
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- ffmpeg CLI new option: -fix_sub_duration_heartbeat
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- WavArc decoder
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version 5.1:
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@ -1356,6 +1356,7 @@ following image formats are supported:
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@item Vorbis @tab E @tab X
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@tab A native but very primitive encoder exists.
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@item Voxware MetaSound @tab @tab X
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@item Waveform Archiver @tab @tab X
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@item WavPack @tab X @tab X
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@item Westwood Audio (SND1) @tab @tab X
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@item Windows Media Audio 1 @tab X @tab X
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@ -781,6 +781,7 @@ OBJS-$(CONFIG_VP9_V4L2M2M_DECODER) += v4l2_m2m_dec.o
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OBJS-$(CONFIG_VQA_DECODER) += vqavideo.o
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OBJS-$(CONFIG_VQC_DECODER) += vqcdec.o
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OBJS-$(CONFIG_WADY_DPCM_DECODER) += dpcm.o
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OBJS-$(CONFIG_WAVARC_DECODER) += wavarc.o
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OBJS-$(CONFIG_WAVPACK_DECODER) += wavpack.o wavpackdata.o dsd.o
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OBJS-$(CONFIG_WAVPACK_ENCODER) += wavpackdata.o wavpackenc.o
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OBJS-$(CONFIG_WBMP_DECODER) += wbmpdec.o
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@ -538,6 +538,7 @@ extern const FFCodec ff_twinvq_decoder;
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extern const FFCodec ff_vmdaudio_decoder;
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extern const FFCodec ff_vorbis_encoder;
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extern const FFCodec ff_vorbis_decoder;
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extern const FFCodec ff_wavarc_decoder;
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extern const FFCodec ff_wavpack_encoder;
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extern const FFCodec ff_wavpack_decoder;
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extern const FFCodec ff_wmalossless_decoder;
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@ -3353,6 +3353,13 @@ static const AVCodecDescriptor codec_descriptors[] = {
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.long_name = NULL_IF_CONFIG_SMALL("FTR Voice"),
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.props = AV_CODEC_PROP_INTRA_ONLY | AV_CODEC_PROP_LOSSY,
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},
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{
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.id = AV_CODEC_ID_WAVARC,
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.type = AVMEDIA_TYPE_AUDIO,
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.name = "wavarc",
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.long_name = NULL_IF_CONFIG_SMALL("Waveform Archiver"),
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.props = AV_CODEC_PROP_INTRA_ONLY | AV_CODEC_PROP_LOSSLESS,
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},
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/* subtitle codecs */
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{
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@ -536,6 +536,7 @@ enum AVCodecID {
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AV_CODEC_ID_MISC4,
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AV_CODEC_ID_APAC,
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AV_CODEC_ID_FTR,
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AV_CODEC_ID_WAVARC,
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/* subtitle codecs */
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AV_CODEC_ID_FIRST_SUBTITLE = 0x17000, ///< A dummy ID pointing at the start of subtitle codecs.
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@ -29,7 +29,7 @@
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#include "version_major.h"
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#define LIBAVCODEC_VERSION_MINOR 61
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#define LIBAVCODEC_VERSION_MINOR 62
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#define LIBAVCODEC_VERSION_MICRO 100
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#define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
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460
libavcodec/wavarc.c
Normal file
460
libavcodec/wavarc.c
Normal file
@ -0,0 +1,460 @@
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/*
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* WavArc audio decoder
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* Copyright (c) 2023 Paul B Mahol
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/internal.h"
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#include "libavutil/intreadwrite.h"
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#include "avcodec.h"
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#include "codec_internal.h"
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#include "decode.h"
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#include "get_bits.h"
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#include "bytestream.h"
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#include "mathops.h"
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#include "unary.h"
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typedef struct WavArcContext {
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GetBitContext gb;
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int shift;
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int nb_samples;
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int offset;
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int eof;
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int skip;
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uint8_t *bitstream;
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int64_t max_framesize;
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int bitstream_size;
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int bitstream_index;
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int pred[2][70];
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int filter[2][70];
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int samples[2][640];
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} WavArcContext;
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static av_cold int wavarc_init(AVCodecContext *avctx)
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{
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WavArcContext *s = avctx->priv_data;
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if (avctx->extradata_size < 44)
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return AVERROR_INVALIDDATA;
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if (AV_RL32(avctx->extradata + 16) != MKTAG('R','I','F','F'))
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return AVERROR_INVALIDDATA;
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if (AV_RL32(avctx->extradata + 24) != MKTAG('W','A','V','E'))
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return AVERROR_INVALIDDATA;
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if (AV_RL32(avctx->extradata + 28) != MKTAG('f','m','t',' '))
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return AVERROR_INVALIDDATA;
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if (AV_RL16(avctx->extradata + 38) != 1 &&
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AV_RL16(avctx->extradata + 38) != 2)
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return AVERROR_INVALIDDATA;
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av_channel_layout_uninit(&avctx->ch_layout);
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av_channel_layout_default(&avctx->ch_layout, AV_RL16(avctx->extradata + 38));
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avctx->sample_rate = AV_RL32(avctx->extradata + 40);
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switch (avctx->extradata[36]) {
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case 0: avctx->sample_fmt = AV_SAMPLE_FMT_U8P; break;
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case 1: avctx->sample_fmt = AV_SAMPLE_FMT_S16P; break;
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}
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s->shift = 0;
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switch (avctx->codec_tag) {
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case MKTAG('1','D','I','F'):
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s->nb_samples = 256;
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s->offset = 4;
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break;
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case MKTAG('2','S','L','P'):
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case MKTAG('3','N','L','P'):
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case MKTAG('4','A','L','P'):
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s->nb_samples = 570;
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s->offset = 70;
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break;
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default:
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return AVERROR_INVALIDDATA;
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}
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s->max_framesize = s->nb_samples * 16;
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s->bitstream = av_calloc(s->max_framesize, sizeof(*s->bitstream));
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if (!s->bitstream)
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return AVERROR(ENOMEM);
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return 0;
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}
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static unsigned get_urice(GetBitContext *gb, int k)
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{
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unsigned x = get_unary(gb, 1, get_bits_left(gb));
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unsigned y = get_bits_long(gb, k);
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unsigned z = (x << k) | y;
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return z;
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}
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static int get_srice(GetBitContext *gb, int k)
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{
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unsigned z = get_urice(gb, k);
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return (z & 1) ? ~((int)(z >> 1)) : z >> 1;
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}
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static void do_stereo(WavArcContext *s, int ch, int correlated, int len)
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{
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const int nb_samples = s->nb_samples;
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const int shift = s->shift;
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if (ch == 0) {
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if (correlated) {
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for (int n = 0; n < len; n++) {
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s->samples[0][n] = s->samples[0][nb_samples + n] >> shift;
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s->samples[1][n] = s->pred[1][n] >> shift;
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}
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} else {
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for (int n = 0; n < len; n++) {
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s->samples[0][n] = s->samples[0][nb_samples + n] >> shift;
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s->samples[1][n] = s->pred[0][n] >> shift;
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}
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}
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} else {
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if (correlated) {
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for (int n = 0; n < nb_samples; n++)
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s->samples[1][n + len] += s->samples[0][n + len];
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}
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for (int n = 0; n < len; n++) {
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s->pred[0][n] = s->samples[1][nb_samples + n];
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s->pred[1][n] = s->pred[0][n] - s->samples[0][nb_samples + n];
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}
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}
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}
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static int decode_1dif(AVCodecContext *avctx,
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WavArcContext *s, GetBitContext *gb)
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{
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int ch, finished, fill, correlated;
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ch = 0;
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finished = 0;
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while (!finished) {
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int *samples = s->samples[ch];
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int k, block_type;
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if (get_bits_left(gb) <= 0)
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return AVERROR_INVALIDDATA;
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block_type = get_urice(gb, 1);
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if (block_type < 4 && block_type >= 0) {
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k = 1 + (avctx->sample_fmt == AV_SAMPLE_FMT_S16P);
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k = get_urice(gb, k) + 1;
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}
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switch (block_type) {
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case 8:
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s->eof = 1;
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return AVERROR_EOF;
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case 7:
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s->nb_samples = get_bits(gb, 8);
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continue;
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case 6:
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s->shift = get_urice(gb, 2);
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continue;
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case 5:
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if (avctx->sample_fmt == AV_SAMPLE_FMT_U8P) {
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fill = (int8_t)get_bits(gb, 8);
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fill -= 0x80;
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} else {
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fill = (int16_t)get_bits(gb, 16);
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fill -= 0x8000;
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}
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for (int n = 0; n < s->nb_samples; n++)
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samples[n + 4] = fill;
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finished = 1;
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break;
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case 4:
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for (int n = 0; n < s->nb_samples; n++)
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samples[n + 4] = 0;
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finished = 1;
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break;
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case 3:
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for (int n = 0; n < s->nb_samples; n++)
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samples[n + 4] = get_srice(gb, k) + (samples[n + 3] - samples[n + 2]) * 3 +
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samples[n + 1];
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finished = 1;
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break;
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case 2:
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for (int n = 0; n < s->nb_samples; n++)
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samples[n + 4] = get_srice(gb, k) + (samples[n + 3] * 2 - samples[n + 2]);
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finished = 1;
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break;
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case 1:
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for (int n = 0; n < s->nb_samples; n++)
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samples[n + 4] = get_srice(gb, k) + samples[n + 3];
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finished = 1;
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break;
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case 0:
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for (int n = 0; n < s->nb_samples; n++)
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samples[n + 4] = get_srice(gb, k);
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finished = 1;
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break;
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default:
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return AVERROR_INVALIDDATA;
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}
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if (finished == 1 && avctx->ch_layout.nb_channels == 2) {
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if (ch == 0)
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correlated = get_bits1(gb);
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finished = ch != 0;
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do_stereo(s, ch, correlated, 4);
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ch = 1;
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}
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}
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if (avctx->ch_layout.nb_channels == 1) {
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for (int n = 0; n < 4; n++)
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s->samples[0][n] = s->samples[0][s->nb_samples + n] >> s->shift;
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}
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return 0;
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}
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static int decode_2slp(AVCodecContext *avctx,
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WavArcContext *s, GetBitContext *gb)
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{
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int ch, finished, fill, correlated, order;
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ch = 0;
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finished = 0;
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while (!finished) {
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int *samples = s->samples[ch];
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int k, block_type;
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if (get_bits_left(gb) <= 0)
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return AVERROR_INVALIDDATA;
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block_type = get_urice(gb, 1);
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if (block_type < 5 && block_type >= 0) {
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k = 1 + (avctx->sample_fmt == AV_SAMPLE_FMT_S16P);
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k = get_urice(gb, k) + 1;
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}
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switch (block_type) {
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case 9:
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s->eof = 1;
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return AVERROR_EOF;
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case 8:
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s->nb_samples = get_urice(gb, 8);
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continue;
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case 7:
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s->shift = get_urice(gb, 2);
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continue;
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case 6:
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if (avctx->sample_fmt == AV_SAMPLE_FMT_U8P) {
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fill = (int8_t)get_bits(gb, 8);
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fill -= 0x80;
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} else {
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fill = (int16_t)get_bits(gb, 16);
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fill -= 0x8000;
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}
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for (int n = 0; n < s->nb_samples; n++)
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samples[n + 70] = fill;
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finished = 1;
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break;
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case 5:
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for (int n = 0; n < s->nb_samples; n++)
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samples[n + 70] = 0;
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finished = 1;
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break;
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case 4:
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for (int n = 0; n < s->nb_samples; n++)
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samples[n + 70] = get_srice(gb, k) + (samples[n + 69] - samples[n + 68]) * 3 +
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samples[n + 67];
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finished = 1;
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break;
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case 3:
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for (int n = 0; n < s->nb_samples; n++)
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samples[n + 70] = get_srice(gb, k) + (samples[n + 69] * 2 - samples[n + 68]);
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finished = 1;
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break;
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case 2:
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for (int n = 0; n < s->nb_samples; n++)
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samples[n + 70] = get_srice(gb, k);
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finished = 1;
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break;
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case 1:
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for (int n = 0; n < s->nb_samples; n++)
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samples[n + 70] = get_srice(gb, k) + samples[n + 69];
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finished = 1;
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break;
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case 0:
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order = get_urice(gb, 2);
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for (int o = 0; o < order; o++)
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s->filter[ch][o] = get_srice(gb, 2);
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for (int n = 0; n < s->nb_samples; n++) {
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int sum = 15;
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for (int o = 0; o < order; o++)
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sum += s->filter[ch][o] * samples[n + 70 - o - 1];
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samples[n + 70] = get_srice(gb, k) + (sum >> 4);
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}
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finished = 1;
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break;
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default:
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return AVERROR_INVALIDDATA;
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}
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if (finished == 1 && avctx->ch_layout.nb_channels == 2) {
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if (ch == 0)
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correlated = get_bits1(gb);
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finished = ch != 0;
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do_stereo(s, ch, correlated, 70);
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ch = 1;
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}
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}
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if (avctx->ch_layout.nb_channels == 1) {
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for (int n = 0; n < 70; n++)
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s->samples[0][n] = s->samples[0][s->nb_samples + n] >> s->shift;
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}
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return 0;
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}
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static int wavarc_decode(AVCodecContext *avctx, AVFrame *frame,
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int *got_frame_ptr, AVPacket *pkt)
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{
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WavArcContext *s = avctx->priv_data;
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GetBitContext *gb = &s->gb;
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int buf_size, input_buf_size;
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const uint8_t *buf;
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int ret, n;
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if ((!pkt->size && !s->bitstream_size) || s->nb_samples == 0 || s->eof) {
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*got_frame_ptr = 0;
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return pkt->size;
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}
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buf_size = FFMIN(pkt->size, s->max_framesize - s->bitstream_size);
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input_buf_size = buf_size;
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if (s->bitstream_index + s->bitstream_size + buf_size + AV_INPUT_BUFFER_PADDING_SIZE > s->max_framesize) {
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memmove(s->bitstream, &s->bitstream[s->bitstream_index], s->bitstream_size);
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s->bitstream_index = 0;
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}
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if (pkt->data)
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memcpy(&s->bitstream[s->bitstream_index + s->bitstream_size], pkt->data, buf_size);
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buf = &s->bitstream[s->bitstream_index];
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buf_size += s->bitstream_size;
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s->bitstream_size = buf_size;
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if (buf_size < s->max_framesize && pkt->data) {
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*got_frame_ptr = 0;
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return input_buf_size;
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}
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if ((ret = init_get_bits8(gb, buf, buf_size)) < 0)
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return ret;
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skip_bits(gb, s->skip);
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switch (avctx->codec_tag) {
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case MKTAG('1','D','I','F'):
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ret = decode_1dif(avctx, s, gb);
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break;
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case MKTAG('2','S','L','P'):
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case MKTAG('3','N','L','P'):
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case MKTAG('4','A','L','P'):
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ret = decode_2slp(avctx, s, gb);
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||||
break;
|
||||
default:
|
||||
ret = AVERROR_INVALIDDATA;
|
||||
}
|
||||
|
||||
if (ret < 0)
|
||||
goto fail;
|
||||
|
||||
s->skip = get_bits_count(gb) - 8 * (get_bits_count(gb) / 8);
|
||||
n = get_bits_count(gb) / 8;
|
||||
|
||||
if (n > buf_size) {
|
||||
fail:
|
||||
s->bitstream_size = 0;
|
||||
s->bitstream_index = 0;
|
||||
return ret;
|
||||
}
|
||||
|
||||
frame->nb_samples = s->nb_samples;
|
||||
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
|
||||
return ret;
|
||||
|
||||
switch (avctx->sample_fmt) {
|
||||
case AV_SAMPLE_FMT_U8P:
|
||||
for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
|
||||
uint8_t *dst = (uint8_t *)frame->extended_data[ch];
|
||||
const int *src = s->samples[ch] + s->offset;
|
||||
|
||||
for (int n = 0; n < frame->nb_samples; n++)
|
||||
dst[n] = src[n] * (1 << s->shift);
|
||||
}
|
||||
break;
|
||||
case AV_SAMPLE_FMT_S16P:
|
||||
for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
|
||||
int16_t *dst = (int16_t *)frame->extended_data[ch];
|
||||
const int *src = s->samples[ch] + s->offset;
|
||||
|
||||
for (int n = 0; n < frame->nb_samples; n++)
|
||||
dst[n] = src[n] * (1 << s->shift);
|
||||
}
|
||||
break;
|
||||
}
|
||||
|
||||
*got_frame_ptr = 1;
|
||||
|
||||
if (s->bitstream_size) {
|
||||
s->bitstream_index += n;
|
||||
s->bitstream_size -= n;
|
||||
return input_buf_size;
|
||||
}
|
||||
|
||||
return n;
|
||||
}
|
||||
|
||||
static av_cold int wavarc_close(AVCodecContext *avctx)
|
||||
{
|
||||
WavArcContext *s = avctx->priv_data;
|
||||
|
||||
av_freep(&s->bitstream);
|
||||
s->bitstream_size = 0;
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
const FFCodec ff_wavarc_decoder = {
|
||||
.p.name = "wavarc",
|
||||
CODEC_LONG_NAME("Waveform Archiver"),
|
||||
.p.type = AVMEDIA_TYPE_AUDIO,
|
||||
.p.id = AV_CODEC_ID_WAVARC,
|
||||
.priv_data_size = sizeof(WavArcContext),
|
||||
.init = wavarc_init,
|
||||
FF_CODEC_DECODE_CB(wavarc_decode),
|
||||
.close = wavarc_close,
|
||||
.p.capabilities = AV_CODEC_CAP_DR1 |
|
||||
AV_CODEC_CAP_SUBFRAMES |
|
||||
AV_CODEC_CAP_DELAY,
|
||||
.p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,
|
||||
AV_SAMPLE_FMT_S16P,
|
||||
AV_SAMPLE_FMT_NONE },
|
||||
};
|
||||
Loading…
Reference in New Issue
Block a user