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ffmpeg/libavcodec/cook.c

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/*
* COOK compatible decoder
* Copyright (c) 2003 Sascha Sommer
* Copyright (c) 2005 Benjamin Larsson
*
* 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
* Cook compatible decoder. Bastardization of the G.722.1 standard.
* This decoder handles RealNetworks, RealAudio G2 data.
* Cook is identified by the codec name cook in RM files.
*
* To use this decoder, a calling application must supply the extradata
* bytes provided from the RM container; 8+ bytes for mono streams and
* 16+ for stereo streams (maybe more).
*
* Codec technicalities (all this assume a buffer length of 1024):
* Cook works with several different techniques to achieve its compression.
* In the timedomain the buffer is divided into 8 pieces and quantized. If
* two neighboring pieces have different quantization index a smooth
* quantization curve is used to get a smooth overlap between the different
* pieces.
* To get to the transformdomain Cook uses a modulated lapped transform.
* The transform domain has 50 subbands with 20 elements each. This
* means only a maximum of 50*20=1000 coefficients are used out of the 1024
* available.
*/
#include "libavutil/channel_layout.h"
#include "libavutil/lfg.h"
#include "audiodsp.h"
#include "avcodec.h"
#include "get_bits.h"
#include "bytestream.h"
#include "fft.h"
#include "internal.h"
#include "sinewin.h"
#include "unary.h"
#include "cookdata.h"
/* the different Cook versions */
#define MONO 0x1000001
#define STEREO 0x1000002
#define JOINT_STEREO 0x1000003
#define MC_COOK 0x2000000
#define SUBBAND_SIZE 20
#define MAX_SUBPACKETS 5
typedef struct cook_gains {
int *now;
int *previous;
} cook_gains;
typedef struct COOKSubpacket {
int ch_idx;
int size;
int num_channels;
int cookversion;
int subbands;
int js_subband_start;
int js_vlc_bits;
int samples_per_channel;
int log2_numvector_size;
unsigned int channel_mask;
VLC channel_coupling;
int joint_stereo;
int bits_per_subpacket;
int bits_per_subpdiv;
int total_subbands;
int numvector_size; // 1 << log2_numvector_size;
float mono_previous_buffer1[1024];
float mono_previous_buffer2[1024];
cook_gains gains1;
cook_gains gains2;
int gain_1[9];
int gain_2[9];
int gain_3[9];
int gain_4[9];
} COOKSubpacket;
typedef struct cook {
/*
* The following 5 functions provide the lowlevel arithmetic on
* the internal audio buffers.
*/
void (*scalar_dequant)(struct cook *q, int index, int quant_index,
int *subband_coef_index, int *subband_coef_sign,
float *mlt_p);
void (*decouple)(struct cook *q,
COOKSubpacket *p,
int subband,
float f1, float f2,
float *decode_buffer,
float *mlt_buffer1, float *mlt_buffer2);
void (*imlt_window)(struct cook *q, float *buffer1,
cook_gains *gains_ptr, float *previous_buffer);
void (*interpolate)(struct cook *q, float *buffer,
int gain_index, int gain_index_next);
void (*saturate_output)(struct cook *q, float *out);
AVCodecContext* avctx;
AudioDSPContext adsp;
GetBitContext gb;
/* stream data */
int num_vectors;
int samples_per_channel;
/* states */
AVLFG random_state;
int discarded_packets;
/* transform data */
FFTContext mdct_ctx;
float* mlt_window;
/* VLC data */
VLC envelope_quant_index[13];
VLC sqvh[7]; // scalar quantization
/* generate tables and related variables */
int gain_size_factor;
float gain_table[23];
/* data buffers */
uint8_t* decoded_bytes_buffer;
DECLARE_ALIGNED(32, float, mono_mdct_output)[2048];
float decode_buffer_1[1024];
float decode_buffer_2[1024];
float decode_buffer_0[1060]; /* static allocation for joint decode */
const float *cplscales[5];
int num_subpackets;
COOKSubpacket subpacket[MAX_SUBPACKETS];
} COOKContext;
static float pow2tab[127];
static float rootpow2tab[127];
/*************** init functions ***************/
/* table generator */
static av_cold void init_pow2table(void)
{
/* fast way of computing 2^i and 2^(0.5*i) for -63 <= i < 64 */
int i;
static const float exp2_tab[2] = {1, M_SQRT2};
float exp2_val = powf(2, -63);
float root_val = powf(2, -32);
for (i = -63; i < 64; i++) {
if (!(i & 1))
root_val *= 2;
pow2tab[63 + i] = exp2_val;
rootpow2tab[63 + i] = root_val * exp2_tab[i & 1];
exp2_val *= 2;
}
}
/* table generator */
static av_cold void init_gain_table(COOKContext *q)
{
int i;
q->gain_size_factor = q->samples_per_channel / 8;
for (i = 0; i < 23; i++)
q->gain_table[i] = pow(pow2tab[i + 52],
(1.0 / (double) q->gain_size_factor));
}
static av_cold int init_cook_vlc_tables(COOKContext *q)
{
int i, result;
result = 0;
for (i = 0; i < 13; i++) {
result |= init_vlc(&q->envelope_quant_index[i], 9, 24,
envelope_quant_index_huffbits[i], 1, 1,
envelope_quant_index_huffcodes[i], 2, 2, 0);
}
av_log(q->avctx, AV_LOG_DEBUG, "sqvh VLC init\n");
for (i = 0; i < 7; i++) {
result |= init_vlc(&q->sqvh[i], vhvlcsize_tab[i], vhsize_tab[i],
cvh_huffbits[i], 1, 1,
cvh_huffcodes[i], 2, 2, 0);
}
for (i = 0; i < q->num_subpackets; i++) {
if (q->subpacket[i].joint_stereo == 1) {
result |= init_vlc(&q->subpacket[i].channel_coupling, 6,
(1 << q->subpacket[i].js_vlc_bits) - 1,
ccpl_huffbits[q->subpacket[i].js_vlc_bits - 2], 1, 1,
ccpl_huffcodes[q->subpacket[i].js_vlc_bits - 2], 2, 2, 0);
av_log(q->avctx, AV_LOG_DEBUG, "subpacket %i Joint-stereo VLC used.\n", i);
}
}
av_log(q->avctx, AV_LOG_DEBUG, "VLC tables initialized.\n");
return result;
}
static av_cold int init_cook_mlt(COOKContext *q)
{
int j, ret;
int mlt_size = q->samples_per_channel;
if ((q->mlt_window = av_malloc_array(mlt_size, sizeof(*q->mlt_window))) == 0)
return AVERROR(ENOMEM);
/* Initialize the MLT window: simple sine window. */
ff_sine_window_init(q->mlt_window, mlt_size);
for (j = 0; j < mlt_size; j++)
q->mlt_window[j] *= sqrt(2.0 / q->samples_per_channel);
/* Initialize the MDCT. */
if ((ret = ff_mdct_init(&q->mdct_ctx, av_log2(mlt_size) + 1, 1, 1.0 / 32768.0))) {
av_freep(&q->mlt_window);
return ret;
}
av_log(q->avctx, AV_LOG_DEBUG, "MDCT initialized, order = %d.\n",
av_log2(mlt_size) + 1);
return 0;
}
static av_cold void init_cplscales_table(COOKContext *q)
{
int i;
for (i = 0; i < 5; i++)
q->cplscales[i] = cplscales[i];
}
/*************** init functions end ***********/
#define DECODE_BYTES_PAD1(bytes) (3 - ((bytes) + 3) % 4)
#define DECODE_BYTES_PAD2(bytes) ((bytes) % 4 + DECODE_BYTES_PAD1(2 * (bytes)))
/**
* Cook indata decoding, every 32 bits are XORed with 0x37c511f2.
* Why? No idea, some checksum/error detection method maybe.
*
* Out buffer size: extra bytes are needed to cope with
* padding/misalignment.
* Subpackets passed to the decoder can contain two, consecutive
* half-subpackets, of identical but arbitrary size.
* 1234 1234 1234 1234 extraA extraB
* Case 1: AAAA BBBB 0 0
* Case 2: AAAA ABBB BB-- 3 3
* Case 3: AAAA AABB BBBB 2 2
* Case 4: AAAA AAAB BBBB BB-- 1 5
*
* Nice way to waste CPU cycles.
*
* @param inbuffer pointer to byte array of indata
* @param out pointer to byte array of outdata
* @param bytes number of bytes
*/
static inline int decode_bytes(const uint8_t *inbuffer, uint8_t *out, int bytes)
{
static const uint32_t tab[4] = {
AV_BE2NE32C(0x37c511f2u), AV_BE2NE32C(0xf237c511u),
AV_BE2NE32C(0x11f237c5u), AV_BE2NE32C(0xc511f237u),
};
int i, off;
uint32_t c;
const uint32_t *buf;
uint32_t *obuf = (uint32_t *) out;
/* FIXME: 64 bit platforms would be able to do 64 bits at a time.
* I'm too lazy though, should be something like
* for (i = 0; i < bitamount / 64; i++)
* (int64_t) out[i] = 0x37c511f237c511f2 ^ av_be2ne64(int64_t) in[i]);
* Buffer alignment needs to be checked. */
off = (intptr_t) inbuffer & 3;
buf = (const uint32_t *) (inbuffer - off);
c = tab[off];
bytes += 3 + off;
for (i = 0; i < bytes / 4; i++)
obuf[i] = c ^ buf[i];
return off;
}
static av_cold int cook_decode_close(AVCodecContext *avctx)
{
int i;
COOKContext *q = avctx->priv_data;
av_log(avctx, AV_LOG_DEBUG, "Deallocating memory.\n");
/* Free allocated memory buffers. */
av_freep(&q->mlt_window);
av_freep(&q->decoded_bytes_buffer);
/* Free the transform. */
ff_mdct_end(&q->mdct_ctx);
/* Free the VLC tables. */
for (i = 0; i < 13; i++)
ff_free_vlc(&q->envelope_quant_index[i]);
for (i = 0; i < 7; i++)
ff_free_vlc(&q->sqvh[i]);
for (i = 0; i < q->num_subpackets; i++)
ff_free_vlc(&q->subpacket[i].channel_coupling);
av_log(avctx, AV_LOG_DEBUG, "Memory deallocated.\n");
return 0;
}
/**
* Fill the gain array for the timedomain quantization.
*
* @param gb pointer to the GetBitContext
* @param gaininfo array[9] of gain indexes
*/
static void decode_gain_info(GetBitContext *gb, int *gaininfo)
{
int i, n;
n = get_unary(gb, 0, get_bits_left(gb)); // amount of elements*2 to update
i = 0;
while (n--) {
int index = get_bits(gb, 3);
int gain = get_bits1(gb) ? get_bits(gb, 4) - 7 : -1;
while (i <= index)
gaininfo[i++] = gain;
}
while (i <= 8)
gaininfo[i++] = 0;
}
/**
* Create the quant index table needed for the envelope.
*
* @param q pointer to the COOKContext
* @param quant_index_table pointer to the array
*/
static int decode_envelope(COOKContext *q, COOKSubpacket *p,
int *quant_index_table)
{
int i, j, vlc_index;
quant_index_table[0] = get_bits(&q->gb, 6) - 6; // This is used later in categorize
for (i = 1; i < p->total_subbands; i++) {
vlc_index = i;
if (i >= p->js_subband_start * 2) {
vlc_index -= p->js_subband_start;
} else {
vlc_index /= 2;
if (vlc_index < 1)
vlc_index = 1;
}
if (vlc_index > 13)
vlc_index = 13; // the VLC tables >13 are identical to No. 13
j = get_vlc2(&q->gb, q->envelope_quant_index[vlc_index - 1].table,
q->envelope_quant_index[vlc_index - 1].bits, 2);
quant_index_table[i] = quant_index_table[i - 1] + j - 12; // differential encoding
if (quant_index_table[i] > 63 || quant_index_table[i] < -63) {
av_log(q->avctx, AV_LOG_ERROR,
"Invalid quantizer %d at position %d, outside [-63, 63] range\n",
quant_index_table[i], i);
return AVERROR_INVALIDDATA;
}
}
return 0;
}
/**
* Calculate the category and category_index vector.
*
* @param q pointer to the COOKContext
* @param quant_index_table pointer to the array
* @param category pointer to the category array
* @param category_index pointer to the category_index array
*/
static void categorize(COOKContext *q, COOKSubpacket *p, const int *quant_index_table,
int *category, int *category_index)
{
int exp_idx, bias, tmpbias1, tmpbias2, bits_left, num_bits, index, v, i, j;
int exp_index2[102] = { 0 };
int exp_index1[102] = { 0 };
int tmp_categorize_array[128 * 2] = { 0 };
int tmp_categorize_array1_idx = p->numvector_size;
int tmp_categorize_array2_idx = p->numvector_size;
bits_left = p->bits_per_subpacket - get_bits_count(&q->gb);
if (bits_left > q->samples_per_channel)
bits_left = q->samples_per_channel +
((bits_left - q->samples_per_channel) * 5) / 8;
bias = -32;
/* Estimate bias. */
for (i = 32; i > 0; i = i / 2) {
num_bits = 0;
index = 0;
for (j = p->total_subbands; j > 0; j--) {
exp_idx = av_clip_uintp2((i - quant_index_table[index] + bias) / 2, 3);
index++;
num_bits += expbits_tab[exp_idx];
}
if (num_bits >= bits_left - 32)
bias += i;
}
/* Calculate total number of bits. */
num_bits = 0;
for (i = 0; i < p->total_subbands; i++) {
exp_idx = av_clip_uintp2((bias - quant_index_table[i]) / 2, 3);
num_bits += expbits_tab[exp_idx];
exp_index1[i] = exp_idx;
exp_index2[i] = exp_idx;
}
tmpbias1 = tmpbias2 = num_bits;
for (j = 1; j < p->numvector_size; j++) {
if (tmpbias1 + tmpbias2 > 2 * bits_left) { /* ---> */
int max = -999999;
index = -1;
for (i = 0; i < p->total_subbands; i++) {
if (exp_index1[i] < 7) {
v = (-2 * exp_index1[i]) - quant_index_table[i] + bias;
if (v >= max) {
max = v;
index = i;
}
}
}
if (index == -1)
break;
tmp_categorize_array[tmp_categorize_array1_idx++] = index;
tmpbias1 -= expbits_tab[exp_index1[index]] -
expbits_tab[exp_index1[index] + 1];
++exp_index1[index];
} else { /* <--- */
int min = 999999;
index = -1;
for (i = 0; i < p->total_subbands; i++) {
if (exp_index2[i] > 0) {
v = (-2 * exp_index2[i]) - quant_index_table[i] + bias;
if (v < min) {
min = v;
index = i;
}
}
}
if (index == -1)
break;
tmp_categorize_array[--tmp_categorize_array2_idx] = index;
tmpbias2 -= expbits_tab[exp_index2[index]] -
expbits_tab[exp_index2[index] - 1];
--exp_index2[index];
}
}
for (i = 0; i < p->total_subbands; i++)
category[i] = exp_index2[i];
for (i = 0; i < p->numvector_size - 1; i++)
category_index[i] = tmp_categorize_array[tmp_categorize_array2_idx++];
}
/**
* Expand the category vector.
*
* @param q pointer to the COOKContext
* @param category pointer to the category array
* @param category_index pointer to the category_index array
*/
static inline void expand_category(COOKContext *q, int *category,
int *category_index)
{
int i;
for (i = 0; i < q->num_vectors; i++)
{
int idx = category_index[i];
if (++category[idx] >= FF_ARRAY_ELEMS(dither_tab))
--category[idx];
}
}
/**
* The real requantization of the mltcoefs
*
* @param q pointer to the COOKContext
* @param index index
* @param quant_index quantisation index
* @param subband_coef_index array of indexes to quant_centroid_tab
* @param subband_coef_sign signs of coefficients
* @param mlt_p pointer into the mlt buffer
*/
static void scalar_dequant_float(COOKContext *q, int index, int quant_index,
int *subband_coef_index, int *subband_coef_sign,
float *mlt_p)
{
int i;
float f1;
for (i = 0; i < SUBBAND_SIZE; i++) {
if (subband_coef_index[i]) {
f1 = quant_centroid_tab[index][subband_coef_index[i]];
if (subband_coef_sign[i])
f1 = -f1;
} else {
/* noise coding if subband_coef_index[i] == 0 */
f1 = dither_tab[index];
if (av_lfg_get(&q->random_state) < 0x80000000)
f1 = -f1;
}
mlt_p[i] = f1 * rootpow2tab[quant_index + 63];
}
}
/**
* Unpack the subband_coef_index and subband_coef_sign vectors.
*
* @param q pointer to the COOKContext
* @param category pointer to the category array
* @param subband_coef_index array of indexes to quant_centroid_tab
* @param subband_coef_sign signs of coefficients
*/
static int unpack_SQVH(COOKContext *q, COOKSubpacket *p, int category,
int *subband_coef_index, int *subband_coef_sign)
{
int i, j;
int vlc, vd, tmp, result;
vd = vd_tab[category];
result = 0;
for (i = 0; i < vpr_tab[category]; i++) {
vlc = get_vlc2(&q->gb, q->sqvh[category].table, q->sqvh[category].bits, 3);
if (p->bits_per_subpacket < get_bits_count(&q->gb)) {
vlc = 0;
result = 1;
}
for (j = vd - 1; j >= 0; j--) {
tmp = (vlc * invradix_tab[category]) / 0x100000;
subband_coef_index[vd * i + j] = vlc - tmp * (kmax_tab[category] + 1);
vlc = tmp;
}
for (j = 0; j < vd; j++) {
if (subband_coef_index[i * vd + j]) {
if (get_bits_count(&q->gb) < p->bits_per_subpacket) {
subband_coef_sign[i * vd + j] = get_bits1(&q->gb);
} else {
result = 1;
subband_coef_sign[i * vd + j] = 0;
}
} else {
subband_coef_sign[i * vd + j] = 0;
}
}
}
return result;
}
/**
* Fill the mlt_buffer with mlt coefficients.
*
* @param q pointer to the COOKContext
* @param category pointer to the category array
* @param quant_index_table pointer to the array
* @param mlt_buffer pointer to mlt coefficients
*/
static void decode_vectors(COOKContext *q, COOKSubpacket *p, int *category,
int *quant_index_table, float *mlt_buffer)
{
/* A zero in this table means that the subband coefficient is
random noise coded. */
int subband_coef_index[SUBBAND_SIZE];
/* A zero in this table means that the subband coefficient is a
positive multiplicator. */
int subband_coef_sign[SUBBAND_SIZE];
int band, j;
int index = 0;
for (band = 0; band < p->total_subbands; band++) {
index = category[band];
if (category[band] < 7) {
if (unpack_SQVH(q, p, category[band], subband_coef_index, subband_coef_sign)) {
index = 7;
for (j = 0; j < p->total_subbands; j++)
category[band + j] = 7;
}
}
if (index >= 7) {
memset(subband_coef_index, 0, sizeof(subband_coef_index));
memset(subband_coef_sign, 0, sizeof(subband_coef_sign));
}
q->scalar_dequant(q, index, quant_index_table[band],
subband_coef_index, subband_coef_sign,
&mlt_buffer[band * SUBBAND_SIZE]);
}
/* FIXME: should this be removed, or moved into loop above? */
if (p->total_subbands * SUBBAND_SIZE >= q->samples_per_channel)
return;
}
static int mono_decode(COOKContext *q, COOKSubpacket *p, float *mlt_buffer)
{
int category_index[128] = { 0 };
int category[128] = { 0 };
int quant_index_table[102];
int res, i;
if ((res = decode_envelope(q, p, quant_index_table)) < 0)
return res;
q->num_vectors = get_bits(&q->gb, p->log2_numvector_size);
categorize(q, p, quant_index_table, category, category_index);
expand_category(q, category, category_index);
for (i=0; i<p->total_subbands; i++) {
if (category[i] > 7)
return AVERROR_INVALIDDATA;
}
decode_vectors(q, p, category, quant_index_table, mlt_buffer);
return 0;
}
/**
* the actual requantization of the timedomain samples
*
* @param q pointer to the COOKContext
* @param buffer pointer to the timedomain buffer
* @param gain_index index for the block multiplier
* @param gain_index_next index for the next block multiplier
*/
static void interpolate_float(COOKContext *q, float *buffer,
int gain_index, int gain_index_next)
{
int i;
float fc1, fc2;
fc1 = pow2tab[gain_index + 63];
if (gain_index == gain_index_next) { // static gain
for (i = 0; i < q->gain_size_factor; i++)
buffer[i] *= fc1;
} else { // smooth gain
fc2 = q->gain_table[11 + (gain_index_next - gain_index)];
for (i = 0; i < q->gain_size_factor; i++) {
buffer[i] *= fc1;
fc1 *= fc2;
}
}
}
/**
* Apply transform window, overlap buffers.
*
* @param q pointer to the COOKContext
* @param inbuffer pointer to the mltcoefficients
* @param gains_ptr current and previous gains
* @param previous_buffer pointer to the previous buffer to be used for overlapping
*/
static void imlt_window_float(COOKContext *q, float *inbuffer,
cook_gains *gains_ptr, float *previous_buffer)
{
const float fc = pow2tab[gains_ptr->previous[0] + 63];
int i;
/* The weird thing here, is that the two halves of the time domain
* buffer are swapped. Also, the newest data, that we save away for
* next frame, has the wrong sign. Hence the subtraction below.
* Almost sounds like a complex conjugate/reverse data/FFT effect.
*/
/* Apply window and overlap */
for (i = 0; i < q->samples_per_channel; i++)
inbuffer[i] = inbuffer[i] * fc * q->mlt_window[i] -
previous_buffer[i] * q->mlt_window[q->samples_per_channel - 1 - i];
}
/**
* The modulated lapped transform, this takes transform coefficients
* and transforms them into timedomain samples.
* Apply transform window, overlap buffers, apply gain profile
* and buffer management.
*
* @param q pointer to the COOKContext
* @param inbuffer pointer to the mltcoefficients
* @param gains_ptr current and previous gains
* @param previous_buffer pointer to the previous buffer to be used for overlapping
*/
static void imlt_gain(COOKContext *q, float *inbuffer,
cook_gains *gains_ptr, float *previous_buffer)
{
float *buffer0 = q->mono_mdct_output;
float *buffer1 = q->mono_mdct_output + q->samples_per_channel;
int i;
/* Inverse modified discrete cosine transform */
q->mdct_ctx.imdct_calc(&q->mdct_ctx, q->mono_mdct_output, inbuffer);
q->imlt_window(q, buffer1, gains_ptr, previous_buffer);
/* Apply gain profile */
for (i = 0; i < 8; i++)
if (gains_ptr->now[i] || gains_ptr->now[i + 1])
q->interpolate(q, &buffer1[q->gain_size_factor * i],
gains_ptr->now[i], gains_ptr->now[i + 1]);
/* Save away the current to be previous block. */
memcpy(previous_buffer, buffer0,
q->samples_per_channel * sizeof(*previous_buffer));
}
/**
* function for getting the jointstereo coupling information
*
* @param q pointer to the COOKContext
* @param decouple_tab decoupling array
*/
static int decouple_info(COOKContext *q, COOKSubpacket *p, int *decouple_tab)
{
int i;
int vlc = get_bits1(&q->gb);
int start = cplband[p->js_subband_start];
int end = cplband[p->subbands - 1];
int length = end - start + 1;
if (start > end)
return 0;
if (vlc)
for (i = 0; i < length; i++)
decouple_tab[start + i] = get_vlc2(&q->gb,
p->channel_coupling.table,
p->channel_coupling.bits, 2);
else
for (i = 0; i < length; i++) {
int v = get_bits(&q->gb, p->js_vlc_bits);
if (v == (1<<p->js_vlc_bits)-1) {
av_log(q->avctx, AV_LOG_ERROR, "decouple value too large\n");
return AVERROR_INVALIDDATA;
}
decouple_tab[start + i] = v;
}
return 0;
}
/**
* function decouples a pair of signals from a single signal via multiplication.
*
* @param q pointer to the COOKContext
* @param subband index of the current subband
* @param f1 multiplier for channel 1 extraction
* @param f2 multiplier for channel 2 extraction
* @param decode_buffer input buffer
* @param mlt_buffer1 pointer to left channel mlt coefficients
* @param mlt_buffer2 pointer to right channel mlt coefficients
*/
static void decouple_float(COOKContext *q,
COOKSubpacket *p,
int subband,
float f1, float f2,
float *decode_buffer,
float *mlt_buffer1, float *mlt_buffer2)
{
int j, tmp_idx;
for (j = 0; j < SUBBAND_SIZE; j++) {
tmp_idx = ((p->js_subband_start + subband) * SUBBAND_SIZE) + j;
mlt_buffer1[SUBBAND_SIZE * subband + j] = f1 * decode_buffer[tmp_idx];
mlt_buffer2[SUBBAND_SIZE * subband + j] = f2 * decode_buffer[tmp_idx];
}
}
/**
* function for decoding joint stereo data
*
* @param q pointer to the COOKContext
* @param mlt_buffer1 pointer to left channel mlt coefficients
* @param mlt_buffer2 pointer to right channel mlt coefficients
*/
static int joint_decode(COOKContext *q, COOKSubpacket *p,
float *mlt_buffer_left, float *mlt_buffer_right)
{
int i, j, res;
int decouple_tab[SUBBAND_SIZE] = { 0 };
float *decode_buffer = q->decode_buffer_0;
int idx, cpl_tmp;
float f1, f2;
const float *cplscale;
memset(decode_buffer, 0, sizeof(q->decode_buffer_0));
/* Make sure the buffers are zeroed out. */
memset(mlt_buffer_left, 0, 1024 * sizeof(*mlt_buffer_left));
memset(mlt_buffer_right, 0, 1024 * sizeof(*mlt_buffer_right));
if ((res = decouple_info(q, p, decouple_tab)) < 0)
return res;
if ((res = mono_decode(q, p, decode_buffer)) < 0)
return res;
/* The two channels are stored interleaved in decode_buffer. */
for (i = 0; i < p->js_subband_start; i++) {
for (j = 0; j < SUBBAND_SIZE; j++) {
mlt_buffer_left[i * 20 + j] = decode_buffer[i * 40 + j];
mlt_buffer_right[i * 20 + j] = decode_buffer[i * 40 + 20 + j];
}
}
/* When we reach js_subband_start (the higher frequencies)
the coefficients are stored in a coupling scheme. */
idx = (1 << p->js_vlc_bits) - 1;
for (i = p->js_subband_start; i < p->subbands; i++) {
cpl_tmp = cplband[i];
idx -= decouple_tab[cpl_tmp];
cplscale = q->cplscales[p->js_vlc_bits - 2]; // choose decoupler table
f1 = cplscale[decouple_tab[cpl_tmp] + 1];
f2 = cplscale[idx];
q->decouple(q, p, i, f1, f2, decode_buffer,
mlt_buffer_left, mlt_buffer_right);
idx = (1 << p->js_vlc_bits) - 1;
}
return 0;
}
/**
* First part of subpacket decoding:
* decode raw stream bytes and read gain info.
*
* @param q pointer to the COOKContext
* @param inbuffer pointer to raw stream data
* @param gains_ptr array of current/prev gain pointers
*/
static inline void decode_bytes_and_gain(COOKContext *q, COOKSubpacket *p,
const uint8_t *inbuffer,
cook_gains *gains_ptr)
{
int offset;
offset = decode_bytes(inbuffer, q->decoded_bytes_buffer,
p->bits_per_subpacket / 8);
init_get_bits(&q->gb, q->decoded_bytes_buffer + offset,
p->bits_per_subpacket);
decode_gain_info(&q->gb, gains_ptr->now);
/* Swap current and previous gains */
FFSWAP(int *, gains_ptr->now, gains_ptr->previous);
}
/**
* Saturate the output signal and interleave.
*
* @param q pointer to the COOKContext
* @param out pointer to the output vector
*/
static void saturate_output_float(COOKContext *q, float *out)
{
q->adsp.vector_clipf(out, q->mono_mdct_output + q->samples_per_channel,
FFALIGN(q->samples_per_channel, 8), -1.0f, 1.0f);
}
/**
* Final part of subpacket decoding:
* Apply modulated lapped transform, gain compensation,
* clip and convert to integer.
*
* @param q pointer to the COOKContext
* @param decode_buffer pointer to the mlt coefficients
* @param gains_ptr array of current/prev gain pointers
* @param previous_buffer pointer to the previous buffer to be used for overlapping
* @param out pointer to the output buffer
*/
static inline void mlt_compensate_output(COOKContext *q, float *decode_buffer,
cook_gains *gains_ptr, float *previous_buffer,
float *out)
{
imlt_gain(q, decode_buffer, gains_ptr, previous_buffer);
if (out)
q->saturate_output(q, out);
}
/**
* Cook subpacket decoding. This function returns one decoded subpacket,
* usually 1024 samples per channel.
*
* @param q pointer to the COOKContext
* @param inbuffer pointer to the inbuffer
* @param outbuffer pointer to the outbuffer
*/
static int decode_subpacket(COOKContext *q, COOKSubpacket *p,
const uint8_t *inbuffer, float **outbuffer)
{
int sub_packet_size = p->size;
int res;
memset(q->decode_buffer_1, 0, sizeof(q->decode_buffer_1));
decode_bytes_and_gain(q, p, inbuffer, &p->gains1);
if (p->joint_stereo) {
if ((res = joint_decode(q, p, q->decode_buffer_1, q->decode_buffer_2)) < 0)
return res;
} else {
if ((res = mono_decode(q, p, q->decode_buffer_1)) < 0)
return res;
if (p->num_channels == 2) {
decode_bytes_and_gain(q, p, inbuffer + sub_packet_size / 2, &p->gains2);
if ((res = mono_decode(q, p, q->decode_buffer_2)) < 0)
return res;
}
}
mlt_compensate_output(q, q->decode_buffer_1, &p->gains1,
p->mono_previous_buffer1,
outbuffer ? outbuffer[p->ch_idx] : NULL);
if (p->num_channels == 2) {
if (p->joint_stereo)
mlt_compensate_output(q, q->decode_buffer_2, &p->gains1,
p->mono_previous_buffer2,
outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
else
mlt_compensate_output(q, q->decode_buffer_2, &p->gains2,
p->mono_previous_buffer2,
outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
}
return 0;
}
static int cook_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
AVFrame *frame = data;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
COOKContext *q = avctx->priv_data;
float **samples = NULL;
int i, ret;
int offset = 0;
int chidx = 0;
if (buf_size < avctx->block_align)
return buf_size;
/* get output buffer */
if (q->discarded_packets >= 2) {
frame->nb_samples = q->samples_per_channel;
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
samples = (float **)frame->extended_data;
}
/* estimate subpacket sizes */
q->subpacket[0].size = avctx->block_align;
for (i = 1; i < q->num_subpackets; i++) {
q->subpacket[i].size = 2 * buf[avctx->block_align - q->num_subpackets + i];
q->subpacket[0].size -= q->subpacket[i].size + 1;
if (q->subpacket[0].size < 0) {
av_log(avctx, AV_LOG_DEBUG,
"frame subpacket size total > avctx->block_align!\n");
return AVERROR_INVALIDDATA;
}
}
/* decode supbackets */
for (i = 0; i < q->num_subpackets; i++) {
q->subpacket[i].bits_per_subpacket = (q->subpacket[i].size * 8) >>
q->subpacket[i].bits_per_subpdiv;
q->subpacket[i].ch_idx = chidx;
av_log(avctx, AV_LOG_DEBUG,
"subpacket[%i] size %i js %i %i block_align %i\n",
i, q->subpacket[i].size, q->subpacket[i].joint_stereo, offset,
avctx->block_align);
if ((ret = decode_subpacket(q, &q->subpacket[i], buf + offset, samples)) < 0)
return ret;
offset += q->subpacket[i].size;
chidx += q->subpacket[i].num_channels;
av_log(avctx, AV_LOG_DEBUG, "subpacket[%i] %i %i\n",
i, q->subpacket[i].size * 8, get_bits_count(&q->gb));
}
/* Discard the first two frames: no valid audio. */
if (q->discarded_packets < 2) {
q->discarded_packets++;
*got_frame_ptr = 0;
return avctx->block_align;
}
*got_frame_ptr = 1;
return avctx->block_align;
}
static void dump_cook_context(COOKContext *q)
{
//int i=0;
#define PRINT(a, b) ff_dlog(q->avctx, " %s = %d\n", a, b);
ff_dlog(q->avctx, "COOKextradata\n");
ff_dlog(q->avctx, "cookversion=%x\n", q->subpacket[0].cookversion);
if (q->subpacket[0].cookversion > STEREO) {
PRINT("js_subband_start", q->subpacket[0].js_subband_start);
PRINT("js_vlc_bits", q->subpacket[0].js_vlc_bits);
}
ff_dlog(q->avctx, "COOKContext\n");
PRINT("nb_channels", q->avctx->channels);
PRINT("bit_rate", (int)q->avctx->bit_rate);
PRINT("sample_rate", q->avctx->sample_rate);
PRINT("samples_per_channel", q->subpacket[0].samples_per_channel);
PRINT("subbands", q->subpacket[0].subbands);
PRINT("js_subband_start", q->subpacket[0].js_subband_start);
PRINT("log2_numvector_size", q->subpacket[0].log2_numvector_size);
PRINT("numvector_size", q->subpacket[0].numvector_size);
PRINT("total_subbands", q->subpacket[0].total_subbands);
}
/**
* Cook initialization
*
* @param avctx pointer to the AVCodecContext
*/
static av_cold int cook_decode_init(AVCodecContext *avctx)
{
COOKContext *q = avctx->priv_data;
GetByteContext gb;
int s = 0;
unsigned int channel_mask = 0;
int samples_per_frame = 0;
int ret;
q->avctx = avctx;
/* Take care of the codec specific extradata. */
if (avctx->extradata_size < 8) {
av_log(avctx, AV_LOG_ERROR, "Necessary extradata missing!\n");
return AVERROR_INVALIDDATA;
}
av_log(avctx, AV_LOG_DEBUG, "codecdata_length=%d\n", avctx->extradata_size);
bytestream2_init(&gb, avctx->extradata, avctx->extradata_size);
/* Take data from the AVCodecContext (RM container). */
if (!avctx->channels) {
av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
return AVERROR_INVALIDDATA;
}
/* Initialize RNG. */
av_lfg_init(&q->random_state, 0);
ff_audiodsp_init(&q->adsp);
while (bytestream2_get_bytes_left(&gb)) {
/* 8 for mono, 16 for stereo, ? for multichannel
Swap to right endianness so we don't need to care later on. */
q->subpacket[s].cookversion = bytestream2_get_be32(&gb);
samples_per_frame = bytestream2_get_be16(&gb);
q->subpacket[s].subbands = bytestream2_get_be16(&gb);
bytestream2_get_be32(&gb); // Unknown unused
q->subpacket[s].js_subband_start = bytestream2_get_be16(&gb);
if (q->subpacket[s].js_subband_start >= 51) {
av_log(avctx, AV_LOG_ERROR, "js_subband_start %d is too large\n", q->subpacket[s].js_subband_start);
return AVERROR_INVALIDDATA;
}
q->subpacket[s].js_vlc_bits = bytestream2_get_be16(&gb);
/* Initialize extradata related variables. */
q->subpacket[s].samples_per_channel = samples_per_frame / avctx->channels;
q->subpacket[s].bits_per_subpacket = avctx->block_align * 8;
/* Initialize default data states. */
q->subpacket[s].log2_numvector_size = 5;
q->subpacket[s].total_subbands = q->subpacket[s].subbands;
q->subpacket[s].num_channels = 1;
/* Initialize version-dependent variables */
av_log(avctx, AV_LOG_DEBUG, "subpacket[%i].cookversion=%x\n", s,
q->subpacket[s].cookversion);
q->subpacket[s].joint_stereo = 0;
switch (q->subpacket[s].cookversion) {
case MONO:
if (avctx->channels != 1) {
avpriv_request_sample(avctx, "Container channels != 1");
return AVERROR_PATCHWELCOME;
}
av_log(avctx, AV_LOG_DEBUG, "MONO\n");
break;
case STEREO:
if (avctx->channels != 1) {
q->subpacket[s].bits_per_subpdiv = 1;
q->subpacket[s].num_channels = 2;
}
av_log(avctx, AV_LOG_DEBUG, "STEREO\n");
break;
case JOINT_STEREO:
if (avctx->channels != 2) {
avpriv_request_sample(avctx, "Container channels != 2");
return AVERROR_PATCHWELCOME;
}
av_log(avctx, AV_LOG_DEBUG, "JOINT_STEREO\n");
if (avctx->extradata_size >= 16) {
q->subpacket[s].total_subbands = q->subpacket[s].subbands +
q->subpacket[s].js_subband_start;
q->subpacket[s].joint_stereo = 1;
q->subpacket[s].num_channels = 2;
}
if (q->subpacket[s].samples_per_channel > 256) {
q->subpacket[s].log2_numvector_size = 6;
}
if (q->subpacket[s].samples_per_channel > 512) {
q->subpacket[s].log2_numvector_size = 7;
}
break;
case MC_COOK:
av_log(avctx, AV_LOG_DEBUG, "MULTI_CHANNEL\n");
channel_mask |= q->subpacket[s].channel_mask = bytestream2_get_be32(&gb);
if (av_get_channel_layout_nb_channels(q->subpacket[s].channel_mask) > 1) {
q->subpacket[s].total_subbands = q->subpacket[s].subbands +
q->subpacket[s].js_subband_start;
q->subpacket[s].joint_stereo = 1;
q->subpacket[s].num_channels = 2;
q->subpacket[s].samples_per_channel = samples_per_frame >> 1;
if (q->subpacket[s].samples_per_channel > 256) {
q->subpacket[s].log2_numvector_size = 6;
}
if (q->subpacket[s].samples_per_channel > 512) {
q->subpacket[s].log2_numvector_size = 7;
}
} else
q->subpacket[s].samples_per_channel = samples_per_frame;
break;
default:
avpriv_request_sample(avctx, "Cook version %d",
q->subpacket[s].cookversion);
return AVERROR_PATCHWELCOME;
}
if (s > 1 && q->subpacket[s].samples_per_channel != q->samples_per_channel) {
av_log(avctx, AV_LOG_ERROR, "different number of samples per channel!\n");
return AVERROR_INVALIDDATA;
} else
q->samples_per_channel = q->subpacket[0].samples_per_channel;
/* Initialize variable relations */
q->subpacket[s].numvector_size = (1 << q->subpacket[s].log2_numvector_size);
/* Try to catch some obviously faulty streams, otherwise it might be exploitable */
if (q->subpacket[s].total_subbands > 53) {
avpriv_request_sample(avctx, "total_subbands > 53");
return AVERROR_PATCHWELCOME;
}
if ((q->subpacket[s].js_vlc_bits > 6) ||
(q->subpacket[s].js_vlc_bits < 2 * q->subpacket[s].joint_stereo)) {
av_log(avctx, AV_LOG_ERROR, "js_vlc_bits = %d, only >= %d and <= 6 allowed!\n",
q->subpacket[s].js_vlc_bits, 2 * q->subpacket[s].joint_stereo);
return AVERROR_INVALIDDATA;
}
if (q->subpacket[s].subbands > 50) {
avpriv_request_sample(avctx, "subbands > 50");
return AVERROR_PATCHWELCOME;
}
if (q->subpacket[s].subbands == 0) {
avpriv_request_sample(avctx, "subbands = 0");
return AVERROR_PATCHWELCOME;
}
q->subpacket[s].gains1.now = q->subpacket[s].gain_1;
q->subpacket[s].gains1.previous = q->subpacket[s].gain_2;
q->subpacket[s].gains2.now = q->subpacket[s].gain_3;
q->subpacket[s].gains2.previous = q->subpacket[s].gain_4;
if (q->num_subpackets + q->subpacket[s].num_channels > q->avctx->channels) {
av_log(avctx, AV_LOG_ERROR, "Too many subpackets %d for channels %d\n", q->num_subpackets, q->avctx->channels);
return AVERROR_INVALIDDATA;
}
q->num_subpackets++;
s++;
if (s > FFMIN(MAX_SUBPACKETS, avctx->block_align)) {
avpriv_request_sample(avctx, "subpackets > %d", FFMIN(MAX_SUBPACKETS, avctx->block_align));
return AVERROR_PATCHWELCOME;
}
}
/* Generate tables */
init_pow2table();
init_gain_table(q);
init_cplscales_table(q);
if ((ret = init_cook_vlc_tables(q)))
return ret;
if (avctx->block_align >= UINT_MAX / 2)
return AVERROR(EINVAL);
/* Pad the databuffer with:
DECODE_BYTES_PAD1 or DECODE_BYTES_PAD2 for decode_bytes(),
AV_INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */
q->decoded_bytes_buffer =
av_mallocz(avctx->block_align
+ DECODE_BYTES_PAD1(avctx->block_align)
+ AV_INPUT_BUFFER_PADDING_SIZE);
if (!q->decoded_bytes_buffer)
return AVERROR(ENOMEM);
/* Initialize transform. */
if ((ret = init_cook_mlt(q)))
return ret;
/* Initialize COOK signal arithmetic handling */
if (1) {
q->scalar_dequant = scalar_dequant_float;
q->decouple = decouple_float;
q->imlt_window = imlt_window_float;
q->interpolate = interpolate_float;
q->saturate_output = saturate_output_float;
}
/* Try to catch some obviously faulty streams, otherwise it might be exploitable */
if (q->samples_per_channel != 256 && q->samples_per_channel != 512 &&
q->samples_per_channel != 1024) {
avpriv_request_sample(avctx, "samples_per_channel = %d",
q->samples_per_channel);
return AVERROR_PATCHWELCOME;
}
avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
if (channel_mask)
avctx->channel_layout = channel_mask;
else
avctx->channel_layout = (avctx->channels == 2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
dump_cook_context(q);
return 0;
}
AVCodec ff_cook_decoder = {
.name = "cook",
.long_name = NULL_IF_CONFIG_SMALL("Cook / Cooker / Gecko (RealAudio G2)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_COOK,
.priv_data_size = sizeof(COOKContext),
.init = cook_decode_init,
.close = cook_decode_close,
.decode = cook_decode_frame,
.capabilities = AV_CODEC_CAP_DR1,
.caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
};
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