ffmpeg/libavcodec/osq.c

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2023-06-27 17:54:25 +00:00
/*
* OSQ audio decoder
* Copyright (c) 2023 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/internal.h"
#include "libavutil/intreadwrite.h"
#include "avcodec.h"
#include "codec_internal.h"
#include "decode.h"
#include "internal.h"
#define BITSTREAM_READER_LE
#include "get_bits.h"
#include "unary.h"
#define OFFSET 5
typedef struct OSQChannel {
unsigned prediction;
unsigned coding_mode;
unsigned residue_parameter;
unsigned residue_bits;
unsigned history[3];
unsigned pos, count;
double sum;
int32_t prev;
} OSQChannel;
typedef struct OSQContext {
GetBitContext gb;
OSQChannel ch[2];
uint8_t *bitstream;
size_t max_framesize;
size_t bitstream_size;
int decorrelate;
int frame_samples;
int64_t nb_samples;
int32_t *decode_buffer[2];
AVPacket *pkt;
int pkt_offset;
} OSQContext;
static av_cold int osq_close(AVCodecContext *avctx)
{
OSQContext *s = avctx->priv_data;
av_freep(&s->bitstream);
s->bitstream_size = 0;
for (int ch = 0; ch < FF_ARRAY_ELEMS(s->decode_buffer); ch++)
av_freep(&s->decode_buffer[ch]);
return 0;
}
static av_cold int osq_init(AVCodecContext *avctx)
{
OSQContext *s = avctx->priv_data;
if (avctx->extradata_size < 48)
return AVERROR(EINVAL);
if (avctx->extradata[0] != 1) {
av_log(avctx, AV_LOG_ERROR, "Unsupported version.\n");
return AVERROR_INVALIDDATA;
}
avctx->sample_rate = AV_RL32(avctx->extradata + 4);
if (avctx->sample_rate < 1)
return AVERROR_INVALIDDATA;
av_channel_layout_uninit(&avctx->ch_layout);
avctx->ch_layout.order = AV_CHANNEL_ORDER_UNSPEC;
avctx->ch_layout.nb_channels = avctx->extradata[3];
if (avctx->ch_layout.nb_channels < 1)
return AVERROR_INVALIDDATA;
if (avctx->ch_layout.nb_channels > FF_ARRAY_ELEMS(s->decode_buffer))
return AVERROR_INVALIDDATA;
switch (avctx->extradata[2]) {
case 8: avctx->sample_fmt = AV_SAMPLE_FMT_U8P; break;
case 16: avctx->sample_fmt = AV_SAMPLE_FMT_S16P; break;
case 20:
case 24:
case 28:
case 32: avctx->sample_fmt = AV_SAMPLE_FMT_S32P; break;
default: return AVERROR_INVALIDDATA;
}
s->nb_samples = AV_RL64(avctx->extradata + 16);
s->frame_samples = AV_RL16(avctx->extradata + 8);
s->max_framesize = (s->frame_samples * 16 + 1024) * avctx->ch_layout.nb_channels;
s->bitstream = av_calloc(s->max_framesize + AV_INPUT_BUFFER_PADDING_SIZE, sizeof(*s->bitstream));
if (!s->bitstream)
return AVERROR(ENOMEM);
for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
s->decode_buffer[ch] = av_calloc(s->frame_samples + OFFSET,
sizeof(*s->decode_buffer[ch]));
if (!s->decode_buffer[ch])
return AVERROR(ENOMEM);
}
s->pkt = avctx->internal->in_pkt;
return 0;
}
static void reset_stats(OSQChannel *cb)
{
memset(cb->history, 0, sizeof(cb->history));
cb->pos = cb->count = cb->sum = 0;
}
static void update_stats(OSQChannel *cb, int val)
{
cb->sum += FFABS(val) - cb->history[cb->pos];
cb->history[cb->pos] = FFABS(val);
cb->pos++;
cb->count++;
if (cb->pos >= FF_ARRAY_ELEMS(cb->history))
cb->pos = 0;
}
static int update_residue_parameter(OSQChannel *cb)
{
double sum, x;
int rice_k;
sum = cb->sum;
x = sum / cb->count;
rice_k = av_ceil_log2(x);
if (rice_k >= 30) {
rice_k = floor(sum / 1.4426952 + 0.5);
if (rice_k < 1)
rice_k = 1;
}
return rice_k;
}
static uint32_t get_urice(GetBitContext *gb, int k)
{
uint32_t z, x, b;
x = get_unary(gb, 1, 512);
b = get_bits_long(gb, k);
z = b | x << k;
return z;
}
static int32_t get_srice(GetBitContext *gb, int x)
{
int32_t y = get_urice(gb, x);
return get_bits1(gb) ? -y : y;
}
static int osq_channel_parameters(AVCodecContext *avctx, int ch)
{
OSQContext *s = avctx->priv_data;
OSQChannel *cb = &s->ch[ch];
GetBitContext *gb = &s->gb;
cb->prev = 0;
cb->prediction = get_urice(gb, 5);
cb->coding_mode = get_urice(gb, 3);
if (cb->prediction >= 15)
return AVERROR_INVALIDDATA;
if (cb->coding_mode > 0 && cb->coding_mode < 3) {
cb->residue_parameter = get_urice(gb, 4);
if (!cb->residue_parameter || cb->residue_parameter >= 31)
return AVERROR_INVALIDDATA;
} else if (cb->coding_mode == 3) {
cb->residue_bits = get_urice(gb, 4);
if (!cb->residue_bits || cb->residue_bits >= 31)
return AVERROR_INVALIDDATA;
} else if (cb->coding_mode) {
return AVERROR_INVALIDDATA;
}
if (cb->coding_mode == 2)
reset_stats(cb);
return 0;
}
#define A (-1)
#define B (-2)
#define C (-3)
#define D (-4)
#define E (-5)
#define P2 ((dst[A] + dst[A]) - dst[B])
#define P3 ((dst[A] - dst[B]) * 3 + dst[C])
static int do_decode(AVCodecContext *avctx, AVFrame *frame, int decorrelate, int downsample)
{
OSQContext *s = avctx->priv_data;
const int nb_channels = avctx->ch_layout.nb_channels;
const int nb_samples = frame->nb_samples;
GetBitContext *gb = &s->gb;
for (int n = 0; n < nb_samples; n++) {
for (int ch = 0; ch < nb_channels; ch++) {
OSQChannel *cb = &s->ch[ch];
int32_t *dst = s->decode_buffer[ch] + OFFSET;
int32_t p, prev = cb->prev;
if (nb_channels == 2 && ch == 1 && decorrelate != s->decorrelate) {
if (!decorrelate) {
s->decode_buffer[1][OFFSET+A] += s->decode_buffer[0][OFFSET+B];
s->decode_buffer[1][OFFSET+B] += s->decode_buffer[0][OFFSET+C];
s->decode_buffer[1][OFFSET+C] += s->decode_buffer[0][OFFSET+D];
s->decode_buffer[1][OFFSET+D] += s->decode_buffer[0][OFFSET+E];
} else {
s->decode_buffer[1][OFFSET+A] -= s->decode_buffer[0][OFFSET+B];
s->decode_buffer[1][OFFSET+B] -= s->decode_buffer[0][OFFSET+C];
s->decode_buffer[1][OFFSET+C] -= s->decode_buffer[0][OFFSET+D];
s->decode_buffer[1][OFFSET+D] -= s->decode_buffer[0][OFFSET+E];
}
s->decorrelate = decorrelate;
}
if (!cb->coding_mode) {
dst[n] = 0;
} else if (cb->coding_mode == 3) {
dst[n] = get_sbits_long(gb, cb->residue_bits);
} else {
dst[n] = get_srice(gb, cb->residue_parameter);
}
if (get_bits_left(gb) < 0) {
av_log(avctx, AV_LOG_ERROR, "overread!\n");
return AVERROR_INVALIDDATA;
}
p = prev / 2;
prev = dst[n];
switch (cb->prediction) {
case 0:
break;
case 1:
dst[n] += dst[A];
break;
case 2:
dst[n] += dst[A] + p;
break;
case 3:
dst[n] += P2;
break;
case 4:
dst[n] += P2 + p;
break;
case 5:
dst[n] += P3;
break;
case 6:
dst[n] += P3 + p;
break;
case 7:
dst[n] += (P2 + P3) / 2 + p;
break;
case 8:
dst[n] += (P2 + P3) / 2;
break;
case 9:
dst[n] += (P2 * 2 + P3) / 3 + p;
break;
case 10:
dst[n] += (P2 + P3 * 2) / 3 + p;
break;
case 11:
dst[n] += (dst[A] + dst[B]) / 2;
break;
case 12:
dst[n] += dst[B];
break;
case 13:
dst[n] += (dst[D] + dst[B]) / 2;
break;
case 14:
dst[n] += (P2 + dst[A]) / 2 + p;
break;
default:
return AVERROR_INVALIDDATA;
}
cb->prev = prev;
if (downsample)
dst[n] *= 256;
dst[E] = dst[D];
dst[D] = dst[C];
dst[C] = dst[B];
dst[B] = dst[A];
dst[A] = dst[n];
if (cb->coding_mode == 2) {
update_stats(cb, dst[n]);
cb->residue_parameter = update_residue_parameter(cb);
}
if (nb_channels == 2 && ch == 1) {
if (decorrelate)
dst[n] += s->decode_buffer[0][OFFSET+n];
}
if (downsample)
dst[A] /= 256;
}
}
return 0;
}
static int osq_decode_block(AVCodecContext *avctx, AVFrame *frame)
{
const int nb_channels = avctx->ch_layout.nb_channels;
OSQContext *s = avctx->priv_data;
int ret, decorrelate, downsample;
GetBitContext *gb = &s->gb;
skip_bits1(gb);
decorrelate = get_bits1(gb);
downsample = get_bits1(gb);
for (int ch = 0; ch < nb_channels; ch++) {
if ((ret = osq_channel_parameters(avctx, ch)) < 0) {
av_log(avctx, AV_LOG_ERROR, "invalid channel parameters\n");
return ret;
}
}
if ((ret = do_decode(avctx, frame, decorrelate, downsample)) < 0)
return ret;
align_get_bits(gb);
switch (avctx->sample_fmt) {
case AV_SAMPLE_FMT_U8P:
for (int ch = 0; ch < nb_channels; ch++) {
uint8_t *dst = (uint8_t *)frame->extended_data[ch];
int32_t *src = s->decode_buffer[ch] + OFFSET;
for (int n = 0; n < frame->nb_samples; n++)
dst[n] = av_clip_uint8(src[n] + 0x80);
}
break;
case AV_SAMPLE_FMT_S16P:
for (int ch = 0; ch < nb_channels; ch++) {
int16_t *dst = (int16_t *)frame->extended_data[ch];
int32_t *src = s->decode_buffer[ch] + OFFSET;
for (int n = 0; n < frame->nb_samples; n++)
dst[n] = (int16_t)src[n];
}
break;
case AV_SAMPLE_FMT_S32P:
for (int ch = 0; ch < nb_channels; ch++) {
int32_t *dst = (int32_t *)frame->extended_data[ch];
int32_t *src = s->decode_buffer[ch] + OFFSET;
for (int n = 0; n < frame->nb_samples; n++)
dst[n] = src[n];
}
break;
default:
return AVERROR_BUG;
}
return 0;
}
static int osq_receive_frame(AVCodecContext *avctx, AVFrame *frame)
{
OSQContext *s = avctx->priv_data;
GetBitContext *gb = &s->gb;
int ret, n;
while (s->bitstream_size < s->max_framesize) {
int size;
if (!s->pkt->data) {
ret = ff_decode_get_packet(avctx, s->pkt);
if (ret == AVERROR_EOF && s->bitstream_size > 0)
break;
if (ret < 0)
return ret;
}
size = FFMIN(s->pkt->size - s->pkt_offset, s->max_framesize - s->bitstream_size);
memcpy(s->bitstream + s->bitstream_size, s->pkt->data + s->pkt_offset, size);
s->bitstream_size += size;
s->pkt_offset += size;
if (s->pkt_offset == s->pkt->size) {
av_packet_unref(s->pkt);
s->pkt_offset = 0;
}
}
frame->nb_samples = FFMIN(s->frame_samples, s->nb_samples);
if (frame->nb_samples <= 0)
return AVERROR_EOF;
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
goto fail;
if ((ret = init_get_bits8(gb, s->bitstream, s->bitstream_size)) < 0)
goto fail;
if ((ret = osq_decode_block(avctx, frame)) < 0)
goto fail;
s->nb_samples -= frame->nb_samples;
n = get_bits_count(gb) / 8;
if (n > s->bitstream_size) {
ret = AVERROR_INVALIDDATA;
goto fail;
}
memmove(s->bitstream, &s->bitstream[n], s->bitstream_size - n);
s->bitstream_size -= n;
return 0;
fail:
s->bitstream_size = 0;
s->pkt_offset = 0;
av_packet_unref(s->pkt);
return ret;
}
const FFCodec ff_osq_decoder = {
.p.name = "osq",
CODEC_LONG_NAME("OSQ (Original Sound Quality)"),
.p.type = AVMEDIA_TYPE_AUDIO,
.p.id = AV_CODEC_ID_OSQ,
.priv_data_size = sizeof(OSQContext),
.init = osq_init,
FF_CODEC_RECEIVE_FRAME_CB(osq_receive_frame),
.close = osq_close,
.p.capabilities = AV_CODEC_CAP_CHANNEL_CONF |
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_S32P,
AV_SAMPLE_FMT_NONE },
};