From: Paul B Mahol <onemda@gmail.com>
To: FFmpeg development discussions and patches <ffmpeg-devel@ffmpeg.org>
Subject: [FFmpeg-devel] [PATCH] A-pac demuxer and decoder
Date: Wed, 21 Sep 2022 09:23:33 +0200
Message-ID: <CAPYw7P6arVSmzk61pJ9U0NNQjCvbz_O64ZCfs_W82BHvDRdDUA@mail.gmail.com> (raw)
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Patches attached.
[-- Attachment #2: 0001-avcodec-add-APAC-decoder.patch --]
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From 751526c456d09bd22a6316196219d79876ae74f1 Mon Sep 17 00:00:00 2001
From: Paul B Mahol <onemda@gmail.com>
Date: Mon, 19 Sep 2022 22:14:05 +0200
Subject: [PATCH 1/2] avcodec: add APAC decoder
Signed-off-by: Paul B Mahol <onemda@gmail.com>
---
libavcodec/Makefile | 1 +
libavcodec/allcodecs.c | 1 +
libavcodec/apac.c | 269 ++++++++++++++++++++++++++++++++++++++++
libavcodec/codec_desc.c | 7 ++
libavcodec/codec_id.h | 1 +
5 files changed, 279 insertions(+)
create mode 100644 libavcodec/apac.c
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index c836252664..b9aa6efaac 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -214,6 +214,7 @@ OBJS-$(CONFIG_AMRWB_DECODER) += amrwbdec.o celp_filters.o \
OBJS-$(CONFIG_AMV_ENCODER) += mjpegenc.o mjpegenc_common.o
OBJS-$(CONFIG_ANM_DECODER) += anm.o
OBJS-$(CONFIG_ANSI_DECODER) += ansi.o cga_data.o
+OBJS-$(CONFIG_APAC_DECODER) += apac.o
OBJS-$(CONFIG_APE_DECODER) += apedec.o
OBJS-$(CONFIG_APTX_DECODER) += aptxdec.o aptx.o
OBJS-$(CONFIG_APTX_ENCODER) += aptxenc.o aptx.o
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index 447225e26b..fc88e25fda 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -432,6 +432,7 @@ extern const FFCodec ff_alac_decoder;
extern const FFCodec ff_als_decoder;
extern const FFCodec ff_amrnb_decoder;
extern const FFCodec ff_amrwb_decoder;
+extern const FFCodec ff_apac_decoder;
extern const FFCodec ff_ape_decoder;
extern const FFCodec ff_aptx_encoder;
extern const FFCodec ff_aptx_decoder;
diff --git a/libavcodec/apac.c b/libavcodec/apac.c
new file mode 100644
index 0000000000..2dc9aacd44
--- /dev/null
+++ b/libavcodec/apac.c
@@ -0,0 +1,269 @@
+/*
+ * APAC audio decoder
+ *
+ * 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
+ */
+
+#include "libavutil/audio_fifo.h"
+#include "libavutil/internal.h"
+#include "libavutil/intreadwrite.h"
+#include "avcodec.h"
+#include "codec_internal.h"
+#include "decode.h"
+#include "get_bits.h"
+
+typedef struct ChContext {
+ int have_code;
+ int last_sample;
+ int last_delta;
+ int bit_length;
+ int block_length;
+ uint8_t block[32 * 2];
+ AVAudioFifo *samples;
+} ChContext;
+
+typedef struct APACContext {
+ GetBitContext gb;
+ int skip;
+
+ int cur_ch;
+ ChContext ch[2];
+
+ uint8_t *bitstream;
+ int64_t max_framesize;
+ int bitstream_size;
+ int bitstream_index;
+} APACContext;
+
+static av_cold int apac_close(AVCodecContext *avctx)
+{
+ APACContext *s = avctx->priv_data;
+
+ av_freep(&s->bitstream);
+ s->bitstream_size = 0;
+
+ for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
+ ChContext *c = &s->ch[ch];
+
+ av_audio_fifo_free(c->samples);
+ }
+
+ return 0;
+}
+
+static av_cold int apac_init(AVCodecContext *avctx)
+{
+ APACContext *s = avctx->priv_data;
+
+ if (avctx->bits_per_coded_sample > 8)
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
+ else
+ avctx->sample_fmt = AV_SAMPLE_FMT_U8P;
+
+ if (avctx->ch_layout.nb_channels < 1 ||
+ avctx->ch_layout.nb_channels > 2)
+ return AVERROR_INVALIDDATA;
+
+ for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
+ ChContext *c = &s->ch[ch];
+
+ c->bit_length = avctx->bits_per_coded_sample;
+ c->block_length = 8;
+ c->have_code = 0;
+ c->samples = av_audio_fifo_alloc(avctx->sample_fmt, 1, 1024);
+ if (!c->samples)
+ return AVERROR(ENOMEM);
+ }
+
+ s->max_framesize = 1024;
+ s->bitstream = av_realloc_f(s->bitstream, s->max_framesize + AV_INPUT_BUFFER_PADDING_SIZE, sizeof(*s->bitstream));
+ if (!s->bitstream)
+ return AVERROR(ENOMEM);
+
+ return 0;
+}
+
+static int get_code(ChContext *c, GetBitContext *gb)
+{
+ if (get_bits1(gb)) {
+ int code = get_bits(gb, 2);
+
+ switch (code) {
+ case 0:
+ c->bit_length--;
+ break;
+ case 1:
+ c->bit_length++;
+ break;
+ case 2:
+ c->bit_length = get_bits(gb, 5);
+ break;
+ case 3:
+ c->block_length = get_bits(gb, 4);
+ return 1;
+ }
+ }
+
+ return 0;
+}
+
+static int apac_decode(AVCodecContext *avctx, AVFrame *frame,
+ int *got_frame_ptr, AVPacket *pkt)
+{
+ APACContext *s = avctx->priv_data;
+ GetBitContext *gb = &s->gb;
+ int ret, n, buf_size, input_buf_size;
+ const uint8_t *buf;
+ int nb_samples;
+
+ if (!pkt->size && s->bitstream_size <= 0) {
+ *got_frame_ptr = 0;
+ return 0;
+ }
+
+ buf_size = pkt->size;
+ input_buf_size = buf_size;
+
+ if (s->bitstream_index > 0 && s->bitstream_size > 0) {
+ memmove(s->bitstream, &s->bitstream[s->bitstream_index], s->bitstream_size);
+ s->bitstream_index = 0;
+ }
+
+ if (s->bitstream_index + s->bitstream_size + buf_size > s->max_framesize) {
+ s->bitstream = av_realloc_f(s->bitstream, s->bitstream_index +
+ s->bitstream_size +
+ buf_size + AV_INPUT_BUFFER_PADDING_SIZE,
+ sizeof(*s->bitstream));
+ if (!s->bitstream)
+ return AVERROR(ENOMEM);
+ s->max_framesize = s->bitstream_index + s->bitstream_size + buf_size;
+ }
+ if (pkt->data)
+ memcpy(&s->bitstream[s->bitstream_index + s->bitstream_size], pkt->data, buf_size);
+ buf = &s->bitstream[s->bitstream_index];
+ buf_size += s->bitstream_size;
+ s->bitstream_size = buf_size;
+
+ frame->nb_samples = s->bitstream_size * 16 * 8;
+ if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
+ return ret;
+
+ if ((ret = init_get_bits8(gb, buf, buf_size)) < 0)
+ return ret;
+
+ skip_bits(gb, s->skip);
+ s->skip = 0;
+
+ while (get_bits_left(gb) > 0) {
+ for (int ch = s->cur_ch; ch < avctx->ch_layout.nb_channels; ch++) {
+ ChContext *c = &s->ch[ch];
+ int16_t *dst16 = (int16_t *)c->block;
+ uint8_t *dst8 = (uint8_t *)c->block;
+ void *samples[4];
+
+ samples[0] = &c->block[0];
+ if (get_bits_left(gb) < 16 && pkt->size) {
+ s->cur_ch = ch;
+ goto end;
+ }
+
+ if (!c->have_code && get_code(c, gb))
+ get_code(c, gb);
+ c->have_code = 0;
+
+ if (c->block_length <= 0)
+ continue;
+
+ if (c->bit_length < 0 ||
+ c->bit_length > 17) {
+ c->bit_length = avctx->bits_per_coded_sample;
+ return AVERROR_INVALIDDATA;
+ }
+
+ if (get_bits_left(gb) < c->block_length * c->bit_length && pkt->size) {
+ c->have_code = 1;
+ s->cur_ch = ch;
+ goto end;
+ }
+
+ for (int i = 0; i < c->block_length; i++) {
+ int val = get_bits_long(gb, c->bit_length);
+ int delta = (val & 1) ? ~(val >> 1) : (val >> 1);
+ int sample;
+
+ delta += c->last_delta;
+ sample = c->last_sample + delta;
+ c->last_delta = delta;
+ c->last_sample = sample;
+
+ switch (avctx->sample_fmt) {
+ case AV_SAMPLE_FMT_S16P:
+ dst16[i] = sample;
+ break;
+ case AV_SAMPLE_FMT_U8P:
+ dst8[i] = sample;
+ break;
+ }
+ }
+
+ av_audio_fifo_write(c->samples, samples, c->block_length);
+ }
+
+ s->cur_ch = 0;
+ }
+end:
+ nb_samples = frame->nb_samples;
+ for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++)
+ nb_samples = FFMIN(av_audio_fifo_size(s->ch[ch].samples), nb_samples);
+
+ frame->nb_samples = nb_samples;
+ for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
+ void *samples[1] = { frame->extended_data[ch] };
+ av_audio_fifo_read(s->ch[ch].samples, samples, nb_samples);
+ }
+
+ s->skip = get_bits_count(gb) - 8 * (get_bits_count(gb) / 8);
+ n = get_bits_count(gb) / 8;
+
+ if (nb_samples > 0 || pkt->size)
+ *got_frame_ptr = 1;
+
+ if (s->bitstream_size > 0) {
+ s->bitstream_index += n;
+ s->bitstream_size -= n;
+ return input_buf_size;
+ }
+ return n;
+}
+
+const FFCodec ff_apac_decoder = {
+ .p.name = "apac",
+ CODEC_LONG_NAME("Marian's A-pac audio"),
+ .p.type = AVMEDIA_TYPE_AUDIO,
+ .p.id = AV_CODEC_ID_APAC,
+ .priv_data_size = sizeof(APACContext),
+ .init = apac_init,
+ FF_CODEC_DECODE_CB(apac_decode),
+ .close = apac_close,
+ .p.capabilities = AV_CODEC_CAP_DELAY |
+ AV_CODEC_CAP_DR1 |
+ AV_CODEC_CAP_SUBFRAMES,
+ .caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
+ .p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,
+ AV_SAMPLE_FMT_S16P,
+ AV_SAMPLE_FMT_NONE },
+};
diff --git a/libavcodec/codec_desc.c b/libavcodec/codec_desc.c
index 648c518b3c..e8e1529401 100644
--- a/libavcodec/codec_desc.c
+++ b/libavcodec/codec_desc.c
@@ -3304,6 +3304,13 @@ static const AVCodecDescriptor codec_descriptors[] = {
.long_name = NULL_IF_CONFIG_SMALL("Micronas SC-4 Audio"),
.props = AV_CODEC_PROP_LOSSY | AV_CODEC_PROP_INTRA_ONLY,
},
+ {
+ .id = AV_CODEC_ID_APAC,
+ .type = AVMEDIA_TYPE_AUDIO,
+ .name = "apac",
+ .long_name = NULL_IF_CONFIG_SMALL("Marian's A-pac audio"),
+ .props = AV_CODEC_PROP_INTRA_ONLY | AV_CODEC_PROP_LOSSLESS,
+ },
/* subtitle codecs */
{
diff --git a/libavcodec/codec_id.h b/libavcodec/codec_id.h
index bc8226ff07..9c01ea9750 100644
--- a/libavcodec/codec_id.h
+++ b/libavcodec/codec_id.h
@@ -529,6 +529,7 @@ enum AVCodecID {
AV_CODEC_ID_DFPWM,
AV_CODEC_ID_BONK,
AV_CODEC_ID_MISC4,
+ AV_CODEC_ID_APAC,
/* subtitle codecs */
AV_CODEC_ID_FIRST_SUBTITLE = 0x17000, ///< A dummy ID pointing at the start of subtitle codecs.
--
2.37.2
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From d2d7ed7a0539424fe3dd72fdc68a95840d54123b Mon Sep 17 00:00:00 2001
From: Paul B Mahol <onemda@gmail.com>
Date: Mon, 19 Sep 2022 22:05:20 +0200
Subject: [PATCH 2/2] avformat: add APAC demuxer
Signed-off-by: Paul B Mahol <onemda@gmail.com>
---
libavformat/Makefile | 1 +
libavformat/allformats.c | 1 +
libavformat/apac.c | 85 ++++++++++++++++++++++++++++++++++++++++
3 files changed, 87 insertions(+)
create mode 100644 libavformat/apac.c
diff --git a/libavformat/Makefile b/libavformat/Makefile
index 19deb33c2e..d669dc8f44 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -101,6 +101,7 @@ OBJS-$(CONFIG_AMRNB_DEMUXER) += amr.o rawdec.o
OBJS-$(CONFIG_AMRWB_DEMUXER) += amr.o rawdec.o
OBJS-$(CONFIG_AMV_MUXER) += amvenc.o
OBJS-$(CONFIG_ANM_DEMUXER) += anm.o
+OBJS-$(CONFIG_APAC_DEMUXER) += apac.o rawdec.o
OBJS-$(CONFIG_APC_DEMUXER) += apc.o
OBJS-$(CONFIG_APE_DEMUXER) += ape.o apetag.o img2.o
OBJS-$(CONFIG_APM_DEMUXER) += apm.o
diff --git a/libavformat/allformats.c b/libavformat/allformats.c
index a545b5ff45..47c419a009 100644
--- a/libavformat/allformats.c
+++ b/libavformat/allformats.c
@@ -52,6 +52,7 @@ extern const AVInputFormat ff_amrnb_demuxer;
extern const AVInputFormat ff_amrwb_demuxer;
extern const AVOutputFormat ff_amv_muxer;
extern const AVInputFormat ff_anm_demuxer;
+extern const AVInputFormat ff_apac_demuxer;
extern const AVInputFormat ff_apc_demuxer;
extern const AVInputFormat ff_ape_demuxer;
extern const AVInputFormat ff_apm_demuxer;
diff --git a/libavformat/apac.c b/libavformat/apac.c
new file mode 100644
index 0000000000..4d484221fe
--- /dev/null
+++ b/libavformat/apac.c
@@ -0,0 +1,85 @@
+/*
+ * APAC demuxer
+ * Copyright (c) 2022 Paul B Mahol
+ *
+ * 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
+ */
+
+#include "libavutil/intreadwrite.h"
+#include "avformat.h"
+#include "demux.h"
+#include "internal.h"
+#include "rawdec.h"
+
+static int apac_probe(const AVProbeData *p)
+{
+ if (AV_RB32(p->buf) == MKBETAG('A','P','A','C') &&
+ AV_RB32(p->buf + 8) == MKBETAG('P','R','O','F') &&
+ AV_RB32(p->buf + 12) == MKBETAG('N','A','D',' '))
+ return AVPROBE_SCORE_MAX;
+
+ return 0;
+}
+
+static int apac_read_header(AVFormatContext *s)
+{
+ AVIOContext *pb = s->pb;
+ uint32_t chunk_size;
+ AVStream *st;
+ int64_t pos;
+
+ avio_skip(pb, 16);
+ chunk_size = avio_rl32(pb);
+ avio_skip(pb, chunk_size);
+ if (avio_rb32(pb) != MKBETAG('P','F','M','T'))
+ return AVERROR_INVALIDDATA;
+ chunk_size = avio_rl32(pb);
+ pos = avio_tell(pb);
+ avio_skip(pb, 2);
+ st = avformat_new_stream(s, NULL);
+ if (!st)
+ return AVERROR(ENOMEM);
+ st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
+ st->codecpar->codec_id = AV_CODEC_ID_APAC;
+ st->codecpar->ch_layout.nb_channels = avio_rl16(pb);
+ st->codecpar->sample_rate = avio_rl32(pb);
+ if (st->codecpar->ch_layout.nb_channels <= 0 ||
+ st->codecpar->sample_rate <= 0)
+ return AVERROR_INVALIDDATA;
+ avio_skip(pb, 2);
+ st->codecpar->bits_per_coded_sample = avio_rl16(pb);
+ avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);
+ avio_skip(pb, (chunk_size + pos) - avio_tell(pb) + (chunk_size & 1));
+ if (avio_rb32(pb) != MKBETAG('P','A','D',' '))
+ return AVERROR_INVALIDDATA;
+ avio_skip(pb, 4);
+
+ return 0;
+}
+
+const AVInputFormat ff_apac_demuxer = {
+ .name = "apac",
+ .long_name = NULL_IF_CONFIG_SMALL("raw APAC"),
+ .read_probe = apac_probe,
+ .read_header = apac_read_header,
+ .read_packet = ff_raw_read_partial_packet,
+ .extensions = "apc",
+ .flags = AVFMT_NOBINSEARCH | AVFMT_NOGENSEARCH | AVFMT_NO_BYTE_SEEK | AVFMT_NOTIMESTAMPS,
+ .raw_codec_id = AV_CODEC_ID_APAC,
+ .priv_data_size = sizeof(FFRawDemuxerContext),
+ .priv_class = &ff_raw_demuxer_class,
+};
--
2.37.2
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next reply other threads:[~2022-09-21 7:23 UTC|newest]
Thread overview: 3+ messages / expand[flat|nested] mbox.gz Atom feed top
2022-09-21 7:23 Paul B Mahol [this message]
2022-09-23 17:48 ` Paul B Mahol
2022-09-23 18:31 ` James Almer
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