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avcodec: add APAC decoder
This commit is contained in:
Paul B Mahol
parent
750f378bec
commit
84f467454b
@ -14,6 +14,7 @@ version <next>:
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- bonk decoder and demuxer
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- Micronas SC-4 audio decoder
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- LAF demuxer
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- APAC decoder
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version 5.1:
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@ -214,6 +214,7 @@ OBJS-$(CONFIG_AMRWB_DECODER) += amrwbdec.o celp_filters.o \
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OBJS-$(CONFIG_AMV_ENCODER) += mjpegenc.o mjpegenc_common.o
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OBJS-$(CONFIG_ANM_DECODER) += anm.o
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OBJS-$(CONFIG_ANSI_DECODER) += ansi.o cga_data.o
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OBJS-$(CONFIG_APAC_DECODER) += apac.o
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OBJS-$(CONFIG_APE_DECODER) += apedec.o
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OBJS-$(CONFIG_APTX_DECODER) += aptxdec.o aptx.o
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OBJS-$(CONFIG_APTX_ENCODER) += aptxenc.o aptx.o
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@ -432,6 +432,7 @@ extern const FFCodec ff_alac_decoder;
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extern const FFCodec ff_als_decoder;
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extern const FFCodec ff_amrnb_decoder;
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extern const FFCodec ff_amrwb_decoder;
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extern const FFCodec ff_apac_decoder;
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extern const FFCodec ff_ape_decoder;
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extern const FFCodec ff_aptx_encoder;
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extern const FFCodec ff_aptx_decoder;
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269
libavcodec/apac.c
Normal file
269
libavcodec/apac.c
Normal file
@ -0,0 +1,269 @@
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/*
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* APAC audio decoder
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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/audio_fifo.h"
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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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typedef struct ChContext {
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int have_code;
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int last_sample;
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int last_delta;
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int bit_length;
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int block_length;
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uint8_t block[32 * 2];
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AVAudioFifo *samples;
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} ChContext;
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typedef struct APACContext {
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GetBitContext gb;
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int skip;
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int cur_ch;
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ChContext ch[2];
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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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} APACContext;
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static av_cold int apac_close(AVCodecContext *avctx)
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{
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APACContext *s = avctx->priv_data;
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av_freep(&s->bitstream);
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s->bitstream_size = 0;
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for (int ch = 0; ch < 2; ch++) {
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ChContext *c = &s->ch[ch];
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av_audio_fifo_free(c->samples);
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}
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return 0;
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}
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static av_cold int apac_init(AVCodecContext *avctx)
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{
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APACContext *s = avctx->priv_data;
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if (avctx->bits_per_coded_sample > 8)
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avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
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else
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avctx->sample_fmt = AV_SAMPLE_FMT_U8P;
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if (avctx->ch_layout.nb_channels < 1 ||
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avctx->ch_layout.nb_channels > 2)
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return AVERROR_INVALIDDATA;
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for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
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ChContext *c = &s->ch[ch];
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c->bit_length = avctx->bits_per_coded_sample;
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c->block_length = 8;
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c->have_code = 0;
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c->samples = av_audio_fifo_alloc(avctx->sample_fmt, 1, 1024);
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if (!c->samples)
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return AVERROR(ENOMEM);
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}
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s->max_framesize = 1024;
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s->bitstream = av_realloc_f(s->bitstream, s->max_framesize + AV_INPUT_BUFFER_PADDING_SIZE, 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 int get_code(ChContext *c, GetBitContext *gb)
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{
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if (get_bits1(gb)) {
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int code = get_bits(gb, 2);
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switch (code) {
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case 0:
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c->bit_length--;
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break;
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case 1:
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c->bit_length++;
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break;
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case 2:
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c->bit_length = get_bits(gb, 5);
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break;
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case 3:
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c->block_length = get_bits(gb, 4);
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return 1;
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}
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}
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return 0;
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}
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static int apac_decode(AVCodecContext *avctx, AVFrame *frame,
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int *got_frame_ptr, AVPacket *pkt)
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{
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APACContext *s = avctx->priv_data;
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GetBitContext *gb = &s->gb;
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int ret, n, buf_size, input_buf_size;
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const uint8_t *buf;
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int nb_samples;
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if (!pkt->size && s->bitstream_size <= 0) {
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*got_frame_ptr = 0;
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return 0;
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}
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buf_size = pkt->size;
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input_buf_size = buf_size;
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if (s->bitstream_index > 0 && s->bitstream_size > 0) {
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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 (s->bitstream_index + s->bitstream_size + buf_size > s->max_framesize) {
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s->bitstream = av_realloc_f(s->bitstream, s->bitstream_index +
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s->bitstream_size +
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buf_size + AV_INPUT_BUFFER_PADDING_SIZE,
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sizeof(*s->bitstream));
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if (!s->bitstream)
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return AVERROR(ENOMEM);
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s->max_framesize = s->bitstream_index + s->bitstream_size + buf_size;
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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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frame->nb_samples = s->bitstream_size * 16 * 8;
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if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
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return ret;
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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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s->skip = 0;
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while (get_bits_left(gb) > 0) {
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for (int ch = s->cur_ch; ch < avctx->ch_layout.nb_channels; ch++) {
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ChContext *c = &s->ch[ch];
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int16_t *dst16 = (int16_t *)c->block;
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uint8_t *dst8 = (uint8_t *)c->block;
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void *samples[4];
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samples[0] = &c->block[0];
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if (get_bits_left(gb) < 16 && pkt->size) {
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s->cur_ch = ch;
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goto end;
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}
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if (!c->have_code && get_code(c, gb))
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get_code(c, gb);
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c->have_code = 0;
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if (c->block_length <= 0)
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continue;
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if (c->bit_length < 0 ||
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c->bit_length > 17) {
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c->bit_length = avctx->bits_per_coded_sample;
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return AVERROR_INVALIDDATA;
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}
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if (get_bits_left(gb) < c->block_length * c->bit_length && pkt->size) {
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c->have_code = 1;
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s->cur_ch = ch;
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goto end;
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}
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for (int i = 0; i < c->block_length; i++) {
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int val = get_bits_long(gb, c->bit_length);
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int delta = (val & 1) ? ~(val >> 1) : (val >> 1);
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int sample;
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delta += c->last_delta;
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sample = c->last_sample + delta;
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c->last_delta = delta;
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c->last_sample = sample;
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switch (avctx->sample_fmt) {
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case AV_SAMPLE_FMT_S16P:
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dst16[i] = sample;
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break;
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case AV_SAMPLE_FMT_U8P:
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dst8[i] = sample;
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break;
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}
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}
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av_audio_fifo_write(c->samples, samples, c->block_length);
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}
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s->cur_ch = 0;
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}
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end:
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nb_samples = frame->nb_samples;
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for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++)
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nb_samples = FFMIN(av_audio_fifo_size(s->ch[ch].samples), nb_samples);
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frame->nb_samples = nb_samples;
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for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
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void *samples[1] = { frame->extended_data[ch] };
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av_audio_fifo_read(s->ch[ch].samples, samples, nb_samples);
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}
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s->skip = get_bits_count(gb) - 8 * (get_bits_count(gb) / 8);
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n = get_bits_count(gb) / 8;
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if (nb_samples > 0 || pkt->size)
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*got_frame_ptr = 1;
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if (s->bitstream_size > 0) {
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s->bitstream_index += n;
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s->bitstream_size -= n;
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return input_buf_size;
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}
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return n;
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}
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const FFCodec ff_apac_decoder = {
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.p.name = "apac",
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CODEC_LONG_NAME("Marian's A-pac audio"),
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.p.type = AVMEDIA_TYPE_AUDIO,
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.p.id = AV_CODEC_ID_APAC,
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.priv_data_size = sizeof(APACContext),
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.init = apac_init,
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FF_CODEC_DECODE_CB(apac_decode),
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.close = apac_close,
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.p.capabilities = AV_CODEC_CAP_DELAY |
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AV_CODEC_CAP_DR1 |
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AV_CODEC_CAP_SUBFRAMES,
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.caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
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.p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,
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AV_SAMPLE_FMT_S16P,
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AV_SAMPLE_FMT_NONE },
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};
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@ -3304,6 +3304,13 @@ static const AVCodecDescriptor codec_descriptors[] = {
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.long_name = NULL_IF_CONFIG_SMALL("Micronas SC-4 Audio"),
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.props = AV_CODEC_PROP_LOSSY | AV_CODEC_PROP_INTRA_ONLY,
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},
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{
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.id = AV_CODEC_ID_APAC,
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.type = AVMEDIA_TYPE_AUDIO,
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.name = "apac",
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.long_name = NULL_IF_CONFIG_SMALL("Marian's A-pac audio"),
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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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@ -529,6 +529,7 @@ enum AVCodecID {
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AV_CODEC_ID_DFPWM,
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AV_CODEC_ID_BONK,
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AV_CODEC_ID_MISC4,
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AV_CODEC_ID_APAC,
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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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@ -29,8 +29,8 @@
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#include "version_major.h"
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#define LIBAVCODEC_VERSION_MINOR 44
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#define LIBAVCODEC_VERSION_MICRO 101
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#define LIBAVCODEC_VERSION_MINOR 45
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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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LIBAVCODEC_VERSION_MINOR, \
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