ffmpeg/libavcodec/apac.c

280 lines
8.1 KiB
C

/*
* 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/mem.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 < 2; 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 ||
avctx->bits_per_coded_sample < 8 ||
avctx->bits_per_coded_sample > 16
)
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;
s->bitstream_index = 0;
s->bitstream_size = 0;
return AVERROR_INVALIDDATA;
}
if (get_bits_left(gb) < c->block_length * c->bit_length) {
if (pkt->size) {
c->have_code = 1;
s->cur_ch = ch;
goto end;
} else {
break;
}
}
for (int i = 0; i < c->block_length; i++) {
int val = get_bits_long(gb, c->bit_length);
unsigned 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 |
#if FF_API_SUBFRAMES
AV_CODEC_CAP_SUBFRAMES |
#endif
AV_CODEC_CAP_DR1,
.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 },
};