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ffmpeg/libavcodec/ralf.c

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/*
* RealAudio Lossless decoder
*
* Copyright (c) 2012 Konstantin Shishkov
*
* 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
*/
/**
* @file
* This is a decoder for Real Audio Lossless format.
* Dedicated to the mastermind behind it, Ralph Wiggum.
*/
#include "libavutil/attributes.h"
#include "libavutil/channel_layout.h"
#include "avcodec.h"
#include "get_bits.h"
#include "golomb.h"
#include "internal.h"
#include "unary.h"
#include "ralfdata.h"
#define FILTER_NONE 0
#define FILTER_RAW 642
typedef struct VLCSet {
VLC filter_params;
VLC bias;
VLC coding_mode;
VLC filter_coeffs[10][11];
VLC short_codes[15];
VLC long_codes[125];
} VLCSet;
#define RALF_MAX_PKT_SIZE 8192
typedef struct RALFContext {
int version;
int max_frame_size;
VLCSet sets[3];
int32_t channel_data[2][4096];
int filter_params; ///< combined filter parameters for the current channel data
int filter_length; ///< length of the filter for the current channel data
int filter_bits; ///< filter precision for the current channel data
int32_t filter[64];
int bias[2]; ///< a constant value added to channel data after filtering
int num_blocks; ///< number of blocks inside the frame
int sample_offset;
int block_size[1 << 12]; ///< size of the blocks
int block_pts[1 << 12]; ///< block start time (in milliseconds)
uint8_t pkt[16384];
int has_pkt;
} RALFContext;
#define MAX_ELEMS 644 // no RALF table uses more than that
static av_cold int init_ralf_vlc(VLC *vlc, const uint8_t *data, int elems)
{
uint8_t lens[MAX_ELEMS];
uint16_t codes[MAX_ELEMS];
int counts[17], prefixes[18];
int i, cur_len;
int max_bits = 0;
int nb = 0;
for (i = 0; i <= 16; i++)
counts[i] = 0;
for (i = 0; i < elems; i++) {
cur_len = (nb ? *data & 0xF : *data >> 4) + 1;
counts[cur_len]++;
max_bits = FFMAX(max_bits, cur_len);
lens[i] = cur_len;
data += nb;
nb ^= 1;
}
prefixes[1] = 0;
for (i = 1; i <= 16; i++)
prefixes[i + 1] = (prefixes[i] + counts[i]) << 1;
for (i = 0; i < elems; i++)
codes[i] = prefixes[lens[i]]++;
return ff_init_vlc_sparse(vlc, FFMIN(max_bits, 9), elems,
lens, 1, 1, codes, 2, 2, NULL, 0, 0, 0);
}
static av_cold int decode_close(AVCodecContext *avctx)
{
RALFContext *ctx = avctx->priv_data;
int i, j, k;
for (i = 0; i < 3; i++) {
ff_free_vlc(&ctx->sets[i].filter_params);
ff_free_vlc(&ctx->sets[i].bias);
ff_free_vlc(&ctx->sets[i].coding_mode);
for (j = 0; j < 10; j++)
for (k = 0; k < 11; k++)
ff_free_vlc(&ctx->sets[i].filter_coeffs[j][k]);
for (j = 0; j < 15; j++)
ff_free_vlc(&ctx->sets[i].short_codes[j]);
for (j = 0; j < 125; j++)
ff_free_vlc(&ctx->sets[i].long_codes[j]);
}
return 0;
}
static av_cold int decode_init(AVCodecContext *avctx)
{
RALFContext *ctx = avctx->priv_data;
int i, j, k;
int ret;
if (avctx->extradata_size < 24 || memcmp(avctx->extradata, "LSD:", 4)) {
av_log(avctx, AV_LOG_ERROR, "Extradata is not groovy, dude\n");
return AVERROR_INVALIDDATA;
}
ctx->version = AV_RB16(avctx->extradata + 4);
if (ctx->version != 0x103) {
avpriv_request_sample(avctx, "Unknown version %X", ctx->version);
return AVERROR_PATCHWELCOME;
}
avctx->channels = AV_RB16(avctx->extradata + 8);
avctx->sample_rate = AV_RB32(avctx->extradata + 12);
if (avctx->channels < 1 || avctx->channels > 2
|| avctx->sample_rate < 8000 || avctx->sample_rate > 96000) {
av_log(avctx, AV_LOG_ERROR, "Invalid coding parameters %d Hz %d ch\n",
avctx->sample_rate, avctx->channels);
return AVERROR_INVALIDDATA;
}
avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
avctx->channel_layout = (avctx->channels == 2) ? AV_CH_LAYOUT_STEREO
: AV_CH_LAYOUT_MONO;
ctx->max_frame_size = AV_RB32(avctx->extradata + 16);
if (ctx->max_frame_size > (1 << 20) || !ctx->max_frame_size) {
av_log(avctx, AV_LOG_ERROR, "invalid frame size %d\n",
ctx->max_frame_size);
}
ctx->max_frame_size = FFMAX(ctx->max_frame_size, avctx->sample_rate);
for (i = 0; i < 3; i++) {
ret = init_ralf_vlc(&ctx->sets[i].filter_params, filter_param_def[i],
FILTERPARAM_ELEMENTS);
if (ret < 0) {
decode_close(avctx);
return ret;
}
ret = init_ralf_vlc(&ctx->sets[i].bias, bias_def[i], BIAS_ELEMENTS);
if (ret < 0) {
decode_close(avctx);
return ret;
}
ret = init_ralf_vlc(&ctx->sets[i].coding_mode, coding_mode_def[i],
CODING_MODE_ELEMENTS);
if (ret < 0) {
decode_close(avctx);
return ret;
}
for (j = 0; j < 10; j++) {
for (k = 0; k < 11; k++) {
ret = init_ralf_vlc(&ctx->sets[i].filter_coeffs[j][k],
filter_coeffs_def[i][j][k],
FILTER_COEFFS_ELEMENTS);
if (ret < 0) {
decode_close(avctx);
return ret;
}
}
}
for (j = 0; j < 15; j++) {
ret = init_ralf_vlc(&ctx->sets[i].short_codes[j],
short_codes_def[i][j], SHORT_CODES_ELEMENTS);
if (ret < 0) {
decode_close(avctx);
return ret;
}
}
for (j = 0; j < 125; j++) {
ret = init_ralf_vlc(&ctx->sets[i].long_codes[j],
long_codes_def[i][j], LONG_CODES_ELEMENTS);
if (ret < 0) {
decode_close(avctx);
return ret;
}
}
}
return 0;
}
static inline int extend_code(GetBitContext *gb, int val, int range, int bits)
{
if (val == 0) {
val = -range - get_ue_golomb(gb);
} else if (val == range * 2) {
val = range + get_ue_golomb(gb);
} else {
val -= range;
}
if (bits)
val = ((unsigned)val << bits) | get_bits(gb, bits);
return val;
}
static int decode_channel(RALFContext *ctx, GetBitContext *gb, int ch,
int length, int mode, int bits)
{
int i, t;
int code_params;
VLCSet *set = ctx->sets + mode;
VLC *code_vlc; int range, range2, add_bits;
int *dst = ctx->channel_data[ch];
ctx->filter_params = get_vlc2(gb, set->filter_params.table, 9, 2);
ctx->filter_bits = (ctx->filter_params - 2) >> 6;
ctx->filter_length = ctx->filter_params - (ctx->filter_bits << 6) - 1;
if (ctx->filter_params == FILTER_RAW) {
for (i = 0; i < length; i++)
dst[i] = get_bits(gb, bits);
ctx->bias[ch] = 0;
return 0;
}
ctx->bias[ch] = get_vlc2(gb, set->bias.table, 9, 2);
ctx->bias[ch] = extend_code(gb, ctx->bias[ch], 127, 4);
if (ctx->filter_params == FILTER_NONE) {
memset(dst, 0, sizeof(*dst) * length);
return 0;
}
if (ctx->filter_params > 1) {
int cmode = 0, coeff = 0;
VLC *vlc = set->filter_coeffs[ctx->filter_bits] + 5;
add_bits = ctx->filter_bits;
for (i = 0; i < ctx->filter_length; i++) {
t = get_vlc2(gb, vlc[cmode].table, vlc[cmode].bits, 2);
t = extend_code(gb, t, 21, add_bits);
if (!cmode)
coeff -= 12 << add_bits;
coeff = t - coeff;
ctx->filter[i] = coeff;
cmode = coeff >> add_bits;
if (cmode < 0) {
cmode = -1 - av_log2(-cmode);
if (cmode < -5)
cmode = -5;
} else if (cmode > 0) {
cmode = 1 + av_log2(cmode);
if (cmode > 5)
cmode = 5;
}
}
}
code_params = get_vlc2(gb, set->coding_mode.table, set->coding_mode.bits, 2);
if (code_params >= 15) {
add_bits = av_clip((code_params / 5 - 3) / 2, 0, 10);
if (add_bits > 9 && (code_params % 5) != 2)
add_bits--;
range = 10;
range2 = 21;
code_vlc = set->long_codes + (code_params - 15);
} else {
add_bits = 0;
range = 6;
range2 = 13;
code_vlc = set->short_codes + code_params;
}
for (i = 0; i < length; i += 2) {
int code1, code2;
t = get_vlc2(gb, code_vlc->table, code_vlc->bits, 2);
code1 = t / range2;
code2 = t % range2;
dst[i] = extend_code(gb, code1, range, 0) * (1U << add_bits);
dst[i + 1] = extend_code(gb, code2, range, 0) * (1U << add_bits);
if (add_bits) {
dst[i] |= get_bits(gb, add_bits);
dst[i + 1] |= get_bits(gb, add_bits);
}
}
return 0;
}
static void apply_lpc(RALFContext *ctx, int ch, int length, int bits)
{
int i, j, acc;
int *audio = ctx->channel_data[ch];
int bias = 1 << (ctx->filter_bits - 1);
int max_clip = (1 << bits) - 1, min_clip = -max_clip - 1;
for (i = 1; i < length; i++) {
int flen = FFMIN(ctx->filter_length, i);
acc = 0;
for (j = 0; j < flen; j++)
acc += (unsigned)ctx->filter[j] * audio[i - j - 1];
if (acc < 0) {
acc = (acc + bias - 1) >> ctx->filter_bits;
acc = FFMAX(acc, min_clip);
} else {
acc = (acc + bias) >> ctx->filter_bits;
acc = FFMIN(acc, max_clip);
}
audio[i] += acc;
}
}
static int decode_block(AVCodecContext *avctx, GetBitContext *gb,
int16_t *dst0, int16_t *dst1)
{
RALFContext *ctx = avctx->priv_data;
int len, ch, ret;
int dmode, mode[2], bits[2];
int *ch0, *ch1;
int i, t, t2;
len = 12 - get_unary(gb, 0, 6);
if (len <= 7) len ^= 1; // codes for length = 6 and 7 are swapped
len = 1 << len;
if (ctx->sample_offset + len > ctx->max_frame_size) {
av_log(avctx, AV_LOG_ERROR,
"Decoder's stomach is crying, it ate too many samples\n");
return AVERROR_INVALIDDATA;
}
if (avctx->channels > 1)
dmode = get_bits(gb, 2) + 1;
else
dmode = 0;
mode[0] = (dmode == 4) ? 1 : 0;
mode[1] = (dmode >= 2) ? 2 : 0;
bits[0] = 16;
bits[1] = (mode[1] == 2) ? 17 : 16;
for (ch = 0; ch < avctx->channels; ch++) {
if ((ret = decode_channel(ctx, gb, ch, len, mode[ch], bits[ch])) < 0)
return ret;
if (ctx->filter_params > 1 && ctx->filter_params != FILTER_RAW) {
ctx->filter_bits += 3;
apply_lpc(ctx, ch, len, bits[ch]);
}
if (get_bits_left(gb) < 0)
return AVERROR_INVALIDDATA;
}
ch0 = ctx->channel_data[0];
ch1 = ctx->channel_data[1];
switch (dmode) {
case 0:
for (i = 0; i < len; i++)
dst0[i] = ch0[i] + ctx->bias[0];
break;
case 1:
for (i = 0; i < len; i++) {
dst0[i] = ch0[i] + ctx->bias[0];
dst1[i] = ch1[i] + ctx->bias[1];
}
break;
case 2:
for (i = 0; i < len; i++) {
ch0[i] += ctx->bias[0];
dst0[i] = ch0[i];
dst1[i] = ch0[i] - (ch1[i] + ctx->bias[1]);
}
break;
case 3:
for (i = 0; i < len; i++) {
t = ch0[i] + ctx->bias[0];
t2 = ch1[i] + ctx->bias[1];
dst0[i] = t + t2;
dst1[i] = t;
}
break;
case 4:
for (i = 0; i < len; i++) {
t = ch1[i] + ctx->bias[1];
t2 = ((ch0[i] + ctx->bias[0]) << 1) | (t & 1);
dst0[i] = (t2 + t) / 2;
dst1[i] = (t2 - t) / 2;
}
break;
}
ctx->sample_offset += len;
return 0;
}
static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,
AVPacket *avpkt)
{
RALFContext *ctx = avctx->priv_data;
AVFrame *frame = data;
int16_t *samples0;
int16_t *samples1;
int ret;
GetBitContext gb;
int table_size, table_bytes, i;
const uint8_t *src, *block_pointer;
int src_size;
int bytes_left;
if (ctx->has_pkt) {
ctx->has_pkt = 0;
table_bytes = (AV_RB16(avpkt->data) + 7) >> 3;
if (table_bytes + 3 > avpkt->size || avpkt->size > RALF_MAX_PKT_SIZE) {
av_log(avctx, AV_LOG_ERROR, "Wrong packet's breath smells of wrong data!\n");
return AVERROR_INVALIDDATA;
}
if (memcmp(ctx->pkt, avpkt->data, 2 + table_bytes)) {
av_log(avctx, AV_LOG_ERROR, "Wrong packet tails are wrong!\n");
return AVERROR_INVALIDDATA;
}
src = ctx->pkt;
src_size = RALF_MAX_PKT_SIZE + avpkt->size;
memcpy(ctx->pkt + RALF_MAX_PKT_SIZE, avpkt->data + 2 + table_bytes,
avpkt->size - 2 - table_bytes);
} else {
if (avpkt->size == RALF_MAX_PKT_SIZE) {
memcpy(ctx->pkt, avpkt->data, avpkt->size);
ctx->has_pkt = 1;
*got_frame_ptr = 0;
return avpkt->size;
}
src = avpkt->data;
src_size = avpkt->size;
}
frame->nb_samples = ctx->max_frame_size;
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
samples0 = (int16_t *)frame->data[0];
samples1 = (int16_t *)frame->data[1];
if (src_size < 5) {
av_log(avctx, AV_LOG_ERROR, "too short packets are too short!\n");
return AVERROR_INVALIDDATA;
}
table_size = AV_RB16(src);
table_bytes = (table_size + 7) >> 3;
if (src_size < table_bytes + 3) {
av_log(avctx, AV_LOG_ERROR, "short packets are short!\n");
return AVERROR_INVALIDDATA;
}
init_get_bits(&gb, src + 2, table_size);
ctx->num_blocks = 0;
while (get_bits_left(&gb) > 0) {
ctx->block_size[ctx->num_blocks] = get_bits(&gb, 13 + avctx->channels);
if (get_bits1(&gb)) {
ctx->block_pts[ctx->num_blocks] = get_bits(&gb, 9);
} else {
ctx->block_pts[ctx->num_blocks] = 0;
}
ctx->num_blocks++;
}
block_pointer = src + table_bytes + 2;
bytes_left = src_size - table_bytes - 2;
ctx->sample_offset = 0;
for (i = 0; i < ctx->num_blocks; i++) {
if (bytes_left < ctx->block_size[i]) {
av_log(avctx, AV_LOG_ERROR, "I'm pedaling backwards\n");
break;
}
init_get_bits(&gb, block_pointer, ctx->block_size[i] * 8);
if (decode_block(avctx, &gb, samples0 + ctx->sample_offset,
samples1 + ctx->sample_offset) < 0) {
av_log(avctx, AV_LOG_ERROR, "Sir, I got carsick in your office. Not decoding the rest of packet.\n");
break;
}
block_pointer += ctx->block_size[i];
bytes_left -= ctx->block_size[i];
}
frame->nb_samples = ctx->sample_offset;
*got_frame_ptr = ctx->sample_offset > 0;
return avpkt->size;
}
static void decode_flush(AVCodecContext *avctx)
{
RALFContext *ctx = avctx->priv_data;
ctx->has_pkt = 0;
}
AVCodec ff_ralf_decoder = {
.name = "ralf",
.long_name = NULL_IF_CONFIG_SMALL("RealAudio Lossless"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_RALF,
.priv_data_size = sizeof(RALFContext),
.init = decode_init,
.close = decode_close,
.decode = decode_frame,
.flush = decode_flush,
.capabilities = AV_CODEC_CAP_DR1,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_NONE },
};
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