videolan/ffmpeg
1
Fork 0
mirror of https://git.videolan.org/git/ffmpeg.git synced 2024-08-04 18:29:59 +02:00
Code Releases Activity
494bce6224
ffmpeg/libavcodec/ralf.c
Kostya Shishkov 494bce6224 ralf: read Huffman code lengths without GetBitContext 
Those descriptions are stored in nibbles, so they are easy to extract.
And this way we don't need to pad tables for possible bit reader overreads.
2012-03-24 11:53:26 +01:00

537 lines
16 KiB
C
Raw Blame History

/*
* RealAudio Lossless decoder
*
* Copyright (c) 2012 Konstantin Shishkov
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; 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 "avcodec.h"
#include "get_bits.h"
#include "golomb.h"
#include "unary.h"
#include "libavutil/audioconvert.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 {
AVFrame frame;
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 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) {
av_log_ask_for_sample(avctx, "unknown version %X\n", 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_S16;
avctx->channel_layout = (avctx->channels == 2) ? AV_CH_LAYOUT_STEREO
: AV_CH_LAYOUT_MONO;
avcodec_get_frame_defaults(&ctx->frame);
avctx->coded_frame = &ctx->frame;
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 = (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) << add_bits;
dst[i + 1] = extend_code(gb, code2, range, 0) << 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 += 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 *dst)
{
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++)
*dst++ = ch0[i] + ctx->bias[0];
break;
case 1:
for (i = 0; i < len; i++) {
*dst++ = ch0[i] + ctx->bias[0];
*dst++ = ch1[i] + ctx->bias[1];
}
break;
case 2:
for (i = 0; i < len; i++) {
ch0[i] += ctx->bias[0];
*dst++ = ch0[i];
*dst++ = 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];
*dst++ = t + t2;
*dst++ = 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);
*dst++ = (t2 + t) / 2;
*dst++ = (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;
int16_t *samples;
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;
}
ctx->frame.nb_samples = ctx->max_frame_size;
if ((ret = avctx->get_buffer(avctx, &ctx->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "Me fail get_buffer()? That's unpossible!\n");
return ret;
}
samples = (int16_t*)ctx->frame.data[0];
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, 15);
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, samples + ctx->sample_offset
* avctx->channels) < 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];
}
ctx->frame.nb_samples = ctx->sample_offset;
*got_frame_ptr = ctx->sample_offset > 0;
*(AVFrame*)data = ctx->frame;
return avpkt->size;
}
static void decode_flush(AVCodecContext *avctx)
{
RALFContext *ctx = avctx->priv_data;
ctx->has_pkt = 0;
}
AVCodec ff_ralf_decoder = {
.name = "ralf",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_RALF,
.priv_data_size = sizeof(RALFContext),
.init = decode_init,
.close = decode_close,
.decode = decode_frame,
.flush = decode_flush,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("RealAudio Lossless")
};
View Git Blame Copy Permalink