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FFmpeg/libavcodec/wmadec.c
Michael Niedermayer 874c5b02c4 Merge commit '594d4d5df3c70404168701dd5c90b7e6e5587793' 
* commit '594d4d5df3c70404168701dd5c90b7e6e5587793':
  lavc: add a wrapper for AVCodecContext.get_buffer().

Conflicts:
	libavcodec/4xm.c
	libavcodec/8svx.c
	libavcodec/bmv.c
	libavcodec/cljr.c
	libavcodec/cscd.c
	libavcodec/dnxhddec.c
	libavcodec/dpcm.c
	libavcodec/dpx.c
	libavcodec/eacmv.c
	libavcodec/eamad.c
	libavcodec/frwu.c
	libavcodec/g723_1.c
	libavcodec/gifdec.c
	libavcodec/idcinvideo.c
	libavcodec/iff.c
	libavcodec/indeo3.c
	libavcodec/internal.h
	libavcodec/interplayvideo.c
	libavcodec/kmvc.c
	libavcodec/mpc7.c
	libavcodec/mpegaudiodec.c
	libavcodec/pcx.c
	libavcodec/pngdec.c
	libavcodec/pnmdec.c
	libavcodec/rl2.c
	libavcodec/snow.c
	libavcodec/targa.c
	libavcodec/tscc.c
	libavcodec/txd.c
	libavcodec/utils.c
	libavcodec/v210dec.c
	libavcodec/vb.c
	libavcodec/vmdav.c
	libavcodec/vp56.c
	libavcodec/vqavideo.c
	libavcodec/wavpack.c
	libavcodec/wnv1.c
	libavcodec/xl.c
	libavcodec/yop.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-12-05 15:18:12 +01:00

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/*
* WMA compatible decoder
* Copyright (c) 2002 The FFmpeg Project
*
* 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
* WMA compatible decoder.
* This decoder handles Microsoft Windows Media Audio data, versions 1 & 2.
* WMA v1 is identified by audio format 0x160 in Microsoft media files
* (ASF/AVI/WAV). WMA v2 is identified by audio format 0x161.
*
* To use this decoder, a calling application must supply the extra data
* bytes provided with the WMA data. These are the extra, codec-specific
* bytes at the end of a WAVEFORMATEX data structure. Transmit these bytes
* to the decoder using the extradata[_size] fields in AVCodecContext. There
* should be 4 extra bytes for v1 data and 6 extra bytes for v2 data.
*/
#include "avcodec.h"
#include "internal.h"
#include "wma.h"
#undef NDEBUG
#include <assert.h>
#define EXPVLCBITS 8
#define EXPMAX ((19+EXPVLCBITS-1)/EXPVLCBITS)
#define HGAINVLCBITS 9
#define HGAINMAX ((13+HGAINVLCBITS-1)/HGAINVLCBITS)
static void wma_lsp_to_curve_init(WMACodecContext *s, int frame_len);
#ifdef TRACE
static void dump_floats(WMACodecContext *s, const char *name, int prec, const float *tab, int n)
{
int i;
tprintf(s->avctx, "%s[%d]:\n", name, n);
for(i=0;i<n;i++) {
if ((i & 7) == 0)
tprintf(s->avctx, "%4d: ", i);
tprintf(s->avctx, " %8.*f", prec, tab[i]);
if ((i & 7) == 7)
tprintf(s->avctx, "\n");
}
if ((i & 7) != 0)
tprintf(s->avctx, "\n");
}
#endif
static int wma_decode_init(AVCodecContext * avctx)
{
WMACodecContext *s = avctx->priv_data;
int i, flags2;
uint8_t *extradata;
s->avctx = avctx;
/* extract flag infos */
flags2 = 0;
extradata = avctx->extradata;
if (avctx->codec->id == AV_CODEC_ID_WMAV1 && avctx->extradata_size >= 4) {
flags2 = AV_RL16(extradata+2);
} else if (avctx->codec->id == AV_CODEC_ID_WMAV2 && avctx->extradata_size >= 6) {
flags2 = AV_RL16(extradata+4);
}
s->use_exp_vlc = flags2 & 0x0001;
s->use_bit_reservoir = flags2 & 0x0002;
s->use_variable_block_len = flags2 & 0x0004;
if(avctx->codec->id == AV_CODEC_ID_WMAV2 && avctx->extradata_size >= 8){
if(AV_RL16(extradata+4)==0xd && s->use_variable_block_len){
av_log(avctx, AV_LOG_WARNING, "Disabling use_variable_block_len, if this fails contact the ffmpeg developers and send us the file\n");
s->use_variable_block_len= 0; // this fixes issue1503
}
}
if(ff_wma_init(avctx, flags2)<0)
return -1;
/* init MDCT */
for(i = 0; i < s->nb_block_sizes; i++)
ff_mdct_init(&s->mdct_ctx[i], s->frame_len_bits - i + 1, 1, 1.0 / 32768.0);
if (s->use_noise_coding) {
init_vlc(&s->hgain_vlc, HGAINVLCBITS, sizeof(ff_wma_hgain_huffbits),
ff_wma_hgain_huffbits, 1, 1,
ff_wma_hgain_huffcodes, 2, 2, 0);
}
if (s->use_exp_vlc) {
init_vlc(&s->exp_vlc, EXPVLCBITS, sizeof(ff_aac_scalefactor_bits), //FIXME move out of context
ff_aac_scalefactor_bits, 1, 1,
ff_aac_scalefactor_code, 4, 4, 0);
} else {
wma_lsp_to_curve_init(s, s->frame_len);
}
avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
return 0;
}
/**
* compute x^-0.25 with an exponent and mantissa table. We use linear
* interpolation to reduce the mantissa table size at a small speed
* expense (linear interpolation approximately doubles the number of
* bits of precision).
*/
static inline float pow_m1_4(WMACodecContext *s, float x)
{
union {
float f;
unsigned int v;
} u, t;
unsigned int e, m;
float a, b;
u.f = x;
e = u.v >> 23;
m = (u.v >> (23 - LSP_POW_BITS)) & ((1 << LSP_POW_BITS) - 1);
/* build interpolation scale: 1 <= t < 2. */
t.v = ((u.v << LSP_POW_BITS) & ((1 << 23) - 1)) | (127 << 23);
a = s->lsp_pow_m_table1[m];
b = s->lsp_pow_m_table2[m];
return s->lsp_pow_e_table[e] * (a + b * t.f);
}
static void wma_lsp_to_curve_init(WMACodecContext *s, int frame_len)
{
float wdel, a, b;
int i, e, m;
wdel = M_PI / frame_len;
for(i=0;i<frame_len;i++)
s->lsp_cos_table[i] = 2.0f * cos(wdel * i);
/* tables for x^-0.25 computation */
for(i=0;i<256;i++) {
e = i - 126;
s->lsp_pow_e_table[i] = pow(2.0, e * -0.25);
}
/* NOTE: these two tables are needed to avoid two operations in
pow_m1_4 */
b = 1.0;
for(i=(1 << LSP_POW_BITS) - 1;i>=0;i--) {
m = (1 << LSP_POW_BITS) + i;
a = (float)m * (0.5 / (1 << LSP_POW_BITS));
a = pow(a, -0.25);
s->lsp_pow_m_table1[i] = 2 * a - b;
s->lsp_pow_m_table2[i] = b - a;
b = a;
}
}
/**
* NOTE: We use the same code as Vorbis here
* @todo optimize it further with SSE/3Dnow
*/
static void wma_lsp_to_curve(WMACodecContext *s,
float *out, float *val_max_ptr,
int n, float *lsp)
{
int i, j;
float p, q, w, v, val_max;
val_max = 0;
for(i=0;i<n;i++) {
p = 0.5f;
q = 0.5f;
w = s->lsp_cos_table[i];
for(j=1;j<NB_LSP_COEFS;j+=2){
q *= w - lsp[j - 1];
p *= w - lsp[j];
}
p *= p * (2.0f - w);
q *= q * (2.0f + w);
v = p + q;
v = pow_m1_4(s, v);
if (v > val_max)
val_max = v;
out[i] = v;
}
*val_max_ptr = val_max;
}
/**
* decode exponents coded with LSP coefficients (same idea as Vorbis)
*/
static void decode_exp_lsp(WMACodecContext *s, int ch)
{
float lsp_coefs[NB_LSP_COEFS];
int val, i;
for(i = 0; i < NB_LSP_COEFS; i++) {
if (i == 0 || i >= 8)
val = get_bits(&s->gb, 3);
else
val = get_bits(&s->gb, 4);
lsp_coefs[i] = ff_wma_lsp_codebook[i][val];
}
wma_lsp_to_curve(s, s->exponents[ch], &s->max_exponent[ch],
s->block_len, lsp_coefs);
}
/** pow(10, i / 16.0) for i in -60..95 */
static const float pow_tab[] = {
1.7782794100389e-04, 2.0535250264571e-04,
2.3713737056617e-04, 2.7384196342644e-04,
3.1622776601684e-04, 3.6517412725484e-04,
4.2169650342858e-04, 4.8696752516586e-04,
5.6234132519035e-04, 6.4938163157621e-04,
7.4989420933246e-04, 8.6596432336006e-04,
1.0000000000000e-03, 1.1547819846895e-03,
1.3335214321633e-03, 1.5399265260595e-03,
1.7782794100389e-03, 2.0535250264571e-03,
2.3713737056617e-03, 2.7384196342644e-03,
3.1622776601684e-03, 3.6517412725484e-03,
4.2169650342858e-03, 4.8696752516586e-03,
5.6234132519035e-03, 6.4938163157621e-03,
7.4989420933246e-03, 8.6596432336006e-03,
1.0000000000000e-02, 1.1547819846895e-02,
1.3335214321633e-02, 1.5399265260595e-02,
1.7782794100389e-02, 2.0535250264571e-02,
2.3713737056617e-02, 2.7384196342644e-02,
3.1622776601684e-02, 3.6517412725484e-02,
4.2169650342858e-02, 4.8696752516586e-02,
5.6234132519035e-02, 6.4938163157621e-02,
7.4989420933246e-02, 8.6596432336007e-02,
1.0000000000000e-01, 1.1547819846895e-01,
1.3335214321633e-01, 1.5399265260595e-01,
1.7782794100389e-01, 2.0535250264571e-01,
2.3713737056617e-01, 2.7384196342644e-01,
3.1622776601684e-01, 3.6517412725484e-01,
4.2169650342858e-01, 4.8696752516586e-01,
5.6234132519035e-01, 6.4938163157621e-01,
7.4989420933246e-01, 8.6596432336007e-01,
1.0000000000000e+00, 1.1547819846895e+00,
1.3335214321633e+00, 1.5399265260595e+00,
1.7782794100389e+00, 2.0535250264571e+00,
2.3713737056617e+00, 2.7384196342644e+00,
3.1622776601684e+00, 3.6517412725484e+00,
4.2169650342858e+00, 4.8696752516586e+00,
5.6234132519035e+00, 6.4938163157621e+00,
7.4989420933246e+00, 8.6596432336007e+00,
1.0000000000000e+01, 1.1547819846895e+01,
1.3335214321633e+01, 1.5399265260595e+01,
1.7782794100389e+01, 2.0535250264571e+01,
2.3713737056617e+01, 2.7384196342644e+01,
3.1622776601684e+01, 3.6517412725484e+01,
4.2169650342858e+01, 4.8696752516586e+01,
5.6234132519035e+01, 6.4938163157621e+01,
7.4989420933246e+01, 8.6596432336007e+01,
1.0000000000000e+02, 1.1547819846895e+02,
1.3335214321633e+02, 1.5399265260595e+02,
1.7782794100389e+02, 2.0535250264571e+02,
2.3713737056617e+02, 2.7384196342644e+02,
3.1622776601684e+02, 3.6517412725484e+02,
4.2169650342858e+02, 4.8696752516586e+02,
5.6234132519035e+02, 6.4938163157621e+02,
7.4989420933246e+02, 8.6596432336007e+02,
1.0000000000000e+03, 1.1547819846895e+03,
1.3335214321633e+03, 1.5399265260595e+03,
1.7782794100389e+03, 2.0535250264571e+03,
2.3713737056617e+03, 2.7384196342644e+03,
3.1622776601684e+03, 3.6517412725484e+03,
4.2169650342858e+03, 4.8696752516586e+03,
5.6234132519035e+03, 6.4938163157621e+03,
7.4989420933246e+03, 8.6596432336007e+03,
1.0000000000000e+04, 1.1547819846895e+04,
1.3335214321633e+04, 1.5399265260595e+04,
1.7782794100389e+04, 2.0535250264571e+04,
2.3713737056617e+04, 2.7384196342644e+04,
3.1622776601684e+04, 3.6517412725484e+04,
4.2169650342858e+04, 4.8696752516586e+04,
5.6234132519035e+04, 6.4938163157621e+04,
7.4989420933246e+04, 8.6596432336007e+04,
1.0000000000000e+05, 1.1547819846895e+05,
1.3335214321633e+05, 1.5399265260595e+05,
1.7782794100389e+05, 2.0535250264571e+05,
2.3713737056617e+05, 2.7384196342644e+05,
3.1622776601684e+05, 3.6517412725484e+05,
4.2169650342858e+05, 4.8696752516586e+05,
5.6234132519035e+05, 6.4938163157621e+05,
7.4989420933246e+05, 8.6596432336007e+05,
};
/**
* decode exponents coded with VLC codes
*/
static int decode_exp_vlc(WMACodecContext *s, int ch)
{
int last_exp, n, code;
const uint16_t *ptr;
float v, max_scale;
uint32_t *q, *q_end, iv;
const float *ptab = pow_tab + 60;
const uint32_t *iptab = (const uint32_t*)ptab;
ptr = s->exponent_bands[s->frame_len_bits - s->block_len_bits];
q = (uint32_t *)s->exponents[ch];
q_end = q + s->block_len;
max_scale = 0;
if (s->version == 1) {
last_exp = get_bits(&s->gb, 5) + 10;
v = ptab[last_exp];
iv = iptab[last_exp];
max_scale = v;
n = *ptr++;
switch (n & 3) do {
case 0: *q++ = iv;
case 3: *q++ = iv;
case 2: *q++ = iv;
case 1: *q++ = iv;
} while ((n -= 4) > 0);
}else
last_exp = 36;
while (q < q_end) {
code = get_vlc2(&s->gb, s->exp_vlc.table, EXPVLCBITS, EXPMAX);
if (code < 0){
av_log(s->avctx, AV_LOG_ERROR, "Exponent vlc invalid\n");
return -1;
}
/* NOTE: this offset is the same as MPEG4 AAC ! */
last_exp += code - 60;
if ((unsigned)last_exp + 60 >= FF_ARRAY_ELEMS(pow_tab)) {
av_log(s->avctx, AV_LOG_ERROR, "Exponent out of range: %d\n",
last_exp);
return -1;
}
v = ptab[last_exp];
iv = iptab[last_exp];
if (v > max_scale)
max_scale = v;
n = *ptr++;
switch (n & 3) do {
case 0: *q++ = iv;
case 3: *q++ = iv;
case 2: *q++ = iv;
case 1: *q++ = iv;
} while ((n -= 4) > 0);
}
s->max_exponent[ch] = max_scale;
return 0;
}
/**
* Apply MDCT window and add into output.
*
* We ensure that when the windows overlap their squared sum
* is always 1 (MDCT reconstruction rule).
*/
static void wma_window(WMACodecContext *s, float *out)
{
float *in = s->output;
int block_len, bsize, n;
/* left part */
if (s->block_len_bits <= s->prev_block_len_bits) {
block_len = s->block_len;
bsize = s->frame_len_bits - s->block_len_bits;
s->dsp.vector_fmul_add(out, in, s->windows[bsize],
out, block_len);
} else {
block_len = 1 << s->prev_block_len_bits;
n = (s->block_len - block_len) / 2;
bsize = s->frame_len_bits - s->prev_block_len_bits;
s->dsp.vector_fmul_add(out+n, in+n, s->windows[bsize],
out+n, block_len);
memcpy(out+n+block_len, in+n+block_len, n*sizeof(float));
}
out += s->block_len;
in += s->block_len;
/* right part */
if (s->block_len_bits <= s->next_block_len_bits) {
block_len = s->block_len;
bsize = s->frame_len_bits - s->block_len_bits;
s->dsp.vector_fmul_reverse(out, in, s->windows[bsize], block_len);
} else {
block_len = 1 << s->next_block_len_bits;
n = (s->block_len - block_len) / 2;
bsize = s->frame_len_bits - s->next_block_len_bits;
memcpy(out, in, n*sizeof(float));
s->dsp.vector_fmul_reverse(out+n, in+n, s->windows[bsize], block_len);
memset(out+n+block_len, 0, n*sizeof(float));
}
}
/**
* @return 0 if OK. 1 if last block of frame. return -1 if
* unrecorrable error.
*/
static int wma_decode_block(WMACodecContext *s)
{
int n, v, a, ch, bsize;
int coef_nb_bits, total_gain;
int nb_coefs[MAX_CHANNELS];
float mdct_norm;
FFTContext *mdct;
#ifdef TRACE
tprintf(s->avctx, "***decode_block: %d:%d\n", s->frame_count - 1, s->block_num);
#endif
/* compute current block length */
if (s->use_variable_block_len) {
n = av_log2(s->nb_block_sizes - 1) + 1;
if (s->reset_block_lengths) {
s->reset_block_lengths = 0;
v = get_bits(&s->gb, n);
if (v >= s->nb_block_sizes){
av_log(s->avctx, AV_LOG_ERROR, "prev_block_len_bits %d out of range\n", s->frame_len_bits - v);
return -1;
}
s->prev_block_len_bits = s->frame_len_bits - v;
v = get_bits(&s->gb, n);
if (v >= s->nb_block_sizes){
av_log(s->avctx, AV_LOG_ERROR, "block_len_bits %d out of range\n", s->frame_len_bits - v);
return -1;
}
s->block_len_bits = s->frame_len_bits - v;
} else {
/* update block lengths */
s->prev_block_len_bits = s->block_len_bits;
s->block_len_bits = s->next_block_len_bits;
}
v = get_bits(&s->gb, n);
if (v >= s->nb_block_sizes){
av_log(s->avctx, AV_LOG_ERROR, "next_block_len_bits %d out of range\n", s->frame_len_bits - v);
return -1;
}
s->next_block_len_bits = s->frame_len_bits - v;
} else {
/* fixed block len */
s->next_block_len_bits = s->frame_len_bits;
s->prev_block_len_bits = s->frame_len_bits;
s->block_len_bits = s->frame_len_bits;
}
if (s->frame_len_bits - s->block_len_bits >= s->nb_block_sizes){
av_log(s->avctx, AV_LOG_ERROR, "block_len_bits not initialized to a valid value\n");
return -1;
}
/* now check if the block length is coherent with the frame length */
s->block_len = 1 << s->block_len_bits;
if ((s->block_pos + s->block_len) > s->frame_len){
av_log(s->avctx, AV_LOG_ERROR, "frame_len overflow\n");
return -1;
}
if (s->avctx->channels == 2) {
s->ms_stereo = get_bits1(&s->gb);
}
v = 0;
for(ch = 0; ch < s->avctx->channels; ch++) {
a = get_bits1(&s->gb);
s->channel_coded[ch] = a;
v |= a;
}
bsize = s->frame_len_bits - s->block_len_bits;
/* if no channel coded, no need to go further */
/* XXX: fix potential framing problems */
if (!v)
goto next;
/* read total gain and extract corresponding number of bits for
coef escape coding */
total_gain = 1;
for(;;) {
a = get_bits(&s->gb, 7);
total_gain += a;
if (a != 127)
break;
}
coef_nb_bits= ff_wma_total_gain_to_bits(total_gain);
/* compute number of coefficients */
n = s->coefs_end[bsize] - s->coefs_start;
for(ch = 0; ch < s->avctx->channels; ch++)
nb_coefs[ch] = n;
/* complex coding */
if (s->use_noise_coding) {
for(ch = 0; ch < s->avctx->channels; ch++) {
if (s->channel_coded[ch]) {
int i, n, a;
n = s->exponent_high_sizes[bsize];
for(i=0;i<n;i++) {
a = get_bits1(&s->gb);
s->high_band_coded[ch][i] = a;
/* if noise coding, the coefficients are not transmitted */
if (a)
nb_coefs[ch] -= s->exponent_high_bands[bsize][i];
}
}
}
for(ch = 0; ch < s->avctx->channels; ch++) {
if (s->channel_coded[ch]) {
int i, n, val, code;
n = s->exponent_high_sizes[bsize];
val = (int)0x80000000;
for(i=0;i<n;i++) {
if (s->high_band_coded[ch][i]) {
if (val == (int)0x80000000) {
val = get_bits(&s->gb, 7) - 19;
} else {
code = get_vlc2(&s->gb, s->hgain_vlc.table, HGAINVLCBITS, HGAINMAX);
if (code < 0){
av_log(s->avctx, AV_LOG_ERROR, "hgain vlc invalid\n");
return -1;
}
val += code - 18;
}
s->high_band_values[ch][i] = val;
}
}
}
}
}
/* exponents can be reused in short blocks. */
if ((s->block_len_bits == s->frame_len_bits) ||
get_bits1(&s->gb)) {
for(ch = 0; ch < s->avctx->channels; ch++) {
if (s->channel_coded[ch]) {
if (s->use_exp_vlc) {
if (decode_exp_vlc(s, ch) < 0)
return -1;
} else {
decode_exp_lsp(s, ch);
}
s->exponents_bsize[ch] = bsize;
}
}
}
/* parse spectral coefficients : just RLE encoding */
for (ch = 0; ch < s->avctx->channels; ch++) {
if (s->channel_coded[ch]) {
int tindex;
WMACoef* ptr = &s->coefs1[ch][0];
/* special VLC tables are used for ms stereo because
there is potentially less energy there */
tindex = (ch == 1 && s->ms_stereo);
memset(ptr, 0, s->block_len * sizeof(WMACoef));
ff_wma_run_level_decode(s->avctx, &s->gb, &s->coef_vlc[tindex],
s->level_table[tindex], s->run_table[tindex],
0, ptr, 0, nb_coefs[ch],
s->block_len, s->frame_len_bits, coef_nb_bits);
}
if (s->version == 1 && s->avctx->channels >= 2) {
align_get_bits(&s->gb);
}
}
/* normalize */
{
int n4 = s->block_len / 2;
mdct_norm = 1.0 / (float)n4;
if (s->version == 1) {
mdct_norm *= sqrt(n4);
}
}
/* finally compute the MDCT coefficients */
for (ch = 0; ch < s->avctx->channels; ch++) {
if (s->channel_coded[ch]) {
WMACoef *coefs1;
float *coefs, *exponents, mult, mult1, noise;
int i, j, n, n1, last_high_band, esize;
float exp_power[HIGH_BAND_MAX_SIZE];
coefs1 = s->coefs1[ch];
exponents = s->exponents[ch];
esize = s->exponents_bsize[ch];
mult = pow(10, total_gain * 0.05) / s->max_exponent[ch];
mult *= mdct_norm;
coefs = s->coefs[ch];
if (s->use_noise_coding) {
mult1 = mult;
/* very low freqs : noise */
for(i = 0;i < s->coefs_start; i++) {
*coefs++ = s->noise_table[s->noise_index] *
exponents[i<<bsize>>esize] * mult1;
s->noise_index = (s->noise_index + 1) & (NOISE_TAB_SIZE - 1);
}
n1 = s->exponent_high_sizes[bsize];
/* compute power of high bands */
exponents = s->exponents[ch] +
(s->high_band_start[bsize]<<bsize>>esize);
last_high_band = 0; /* avoid warning */
for(j=0;j<n1;j++) {
n = s->exponent_high_bands[s->frame_len_bits -
s->block_len_bits][j];
if (s->high_band_coded[ch][j]) {
float e2, v;
e2 = 0;
for(i = 0;i < n; i++) {
v = exponents[i<<bsize>>esize];
e2 += v * v;
}
exp_power[j] = e2 / n;
last_high_band = j;
tprintf(s->avctx, "%d: power=%f (%d)\n", j, exp_power[j], n);
}
exponents += n<<bsize>>esize;
}
/* main freqs and high freqs */
exponents = s->exponents[ch] + (s->coefs_start<<bsize>>esize);
for(j=-1;j<n1;j++) {
if (j < 0) {
n = s->high_band_start[bsize] -
s->coefs_start;
} else {
n = s->exponent_high_bands[s->frame_len_bits -
s->block_len_bits][j];
}
if (j >= 0 && s->high_band_coded[ch][j]) {
/* use noise with specified power */
mult1 = sqrt(exp_power[j] / exp_power[last_high_band]);
/* XXX: use a table */
mult1 = mult1 * pow(10, s->high_band_values[ch][j] * 0.05);
mult1 = mult1 / (s->max_exponent[ch] * s->noise_mult);
mult1 *= mdct_norm;
for(i = 0;i < n; i++) {
noise = s->noise_table[s->noise_index];
s->noise_index = (s->noise_index + 1) & (NOISE_TAB_SIZE - 1);
*coefs++ = noise *
exponents[i<<bsize>>esize] * mult1;
}
exponents += n<<bsize>>esize;
} else {
/* coded values + small noise */
for(i = 0;i < n; i++) {
noise = s->noise_table[s->noise_index];
s->noise_index = (s->noise_index + 1) & (NOISE_TAB_SIZE - 1);
*coefs++ = ((*coefs1++) + noise) *
exponents[i<<bsize>>esize] * mult;
}
exponents += n<<bsize>>esize;
}
}
/* very high freqs : noise */
n = s->block_len - s->coefs_end[bsize];
mult1 = mult * exponents[((-1<<bsize))>>esize];
for(i = 0; i < n; i++) {
*coefs++ = s->noise_table[s->noise_index] * mult1;
s->noise_index = (s->noise_index + 1) & (NOISE_TAB_SIZE - 1);
}
} else {
/* XXX: optimize more */
for(i = 0;i < s->coefs_start; i++)
*coefs++ = 0.0;
n = nb_coefs[ch];
for(i = 0;i < n; i++) {
*coefs++ = coefs1[i] * exponents[i<<bsize>>esize] * mult;
}
n = s->block_len - s->coefs_end[bsize];
for(i = 0;i < n; i++)
*coefs++ = 0.0;
}
}
}
#ifdef TRACE
for (ch = 0; ch < s->avctx->channels; ch++) {
if (s->channel_coded[ch]) {
dump_floats(s, "exponents", 3, s->exponents[ch], s->block_len);
dump_floats(s, "coefs", 1, s->coefs[ch], s->block_len);
}
}
#endif
if (s->ms_stereo && s->channel_coded[1]) {
/* nominal case for ms stereo: we do it before mdct */
/* no need to optimize this case because it should almost
never happen */
if (!s->channel_coded[0]) {
tprintf(s->avctx, "rare ms-stereo case happened\n");
memset(s->coefs[0], 0, sizeof(float) * s->block_len);
s->channel_coded[0] = 1;
}
s->dsp.butterflies_float(s->coefs[0], s->coefs[1], s->block_len);
}
next:
mdct = &s->mdct_ctx[bsize];
for (ch = 0; ch < s->avctx->channels; ch++) {
int n4, index;
n4 = s->block_len / 2;
if(s->channel_coded[ch]){
mdct->imdct_calc(mdct, s->output, s->coefs[ch]);
}else if(!(s->ms_stereo && ch==1))
memset(s->output, 0, sizeof(s->output));
/* multiply by the window and add in the frame */
index = (s->frame_len / 2) + s->block_pos - n4;
wma_window(s, &s->frame_out[ch][index]);
}
/* update block number */
s->block_num++;
s->block_pos += s->block_len;
if (s->block_pos >= s->frame_len)
return 1;
else
return 0;
}
/* decode a frame of frame_len samples */
static int wma_decode_frame(WMACodecContext *s, float **samples,
int samples_offset)
{
int ret, ch;
#ifdef TRACE
tprintf(s->avctx, "***decode_frame: %d size=%d\n", s->frame_count++, s->frame_len);
#endif
/* read each block */
s->block_num = 0;
s->block_pos = 0;
for(;;) {
ret = wma_decode_block(s);
if (ret < 0)
return -1;
if (ret)
break;
}
for (ch = 0; ch < s->avctx->channels; ch++) {
/* copy current block to output */
memcpy(samples[ch] + samples_offset, s->frame_out[ch],
s->frame_len * sizeof(*s->frame_out[ch]));
/* prepare for next block */
memmove(&s->frame_out[ch][0], &s->frame_out[ch][s->frame_len],
s->frame_len * sizeof(*s->frame_out[ch]));
#ifdef TRACE
dump_floats(s, "samples", 6, samples[ch] + samples_offset, s->frame_len);
#endif
}
return 0;
}
static int wma_decode_superframe(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
WMACodecContext *s = avctx->priv_data;
int nb_frames, bit_offset, i, pos, len, ret;
uint8_t *q;
float **samples;
int samples_offset;
tprintf(avctx, "***decode_superframe:\n");
if(buf_size==0){
s->last_superframe_len = 0;
return 0;
}
if (buf_size < avctx->block_align) {
av_log(avctx, AV_LOG_ERROR,
"Input packet size too small (%d < %d)\n",
buf_size, avctx->block_align);
return AVERROR_INVALIDDATA;
}
if(avctx->block_align)
buf_size = avctx->block_align;
init_get_bits(&s->gb, buf, buf_size*8);
if (s->use_bit_reservoir) {
/* read super frame header */
skip_bits(&s->gb, 4); /* super frame index */
nb_frames = get_bits(&s->gb, 4) - (s->last_superframe_len <= 0);
} else {
nb_frames = 1;
}
/* get output buffer */
s->frame.nb_samples = nb_frames * s->frame_len;
if ((ret = ff_get_buffer(avctx, &s->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples = (float **)s->frame.extended_data;
samples_offset = 0;
if (s->use_bit_reservoir) {
bit_offset = get_bits(&s->gb, s->byte_offset_bits + 3);
if (bit_offset > get_bits_left(&s->gb)) {
av_log(avctx, AV_LOG_ERROR,
"Invalid last frame bit offset %d > buf size %d (%d)\n",
bit_offset, get_bits_left(&s->gb), buf_size);
goto fail;
}
if (s->last_superframe_len > 0) {
/* add bit_offset bits to last frame */
if ((s->last_superframe_len + ((bit_offset + 7) >> 3)) >
MAX_CODED_SUPERFRAME_SIZE)
goto fail;
q = s->last_superframe + s->last_superframe_len;
len = bit_offset;
while (len > 7) {
*q++ = (get_bits)(&s->gb, 8);
len -= 8;
}
if (len > 0) {
*q++ = (get_bits)(&s->gb, len) << (8 - len);
}
memset(q, 0, FF_INPUT_BUFFER_PADDING_SIZE);
/* XXX: bit_offset bits into last frame */
init_get_bits(&s->gb, s->last_superframe, s->last_superframe_len * 8 + bit_offset);
/* skip unused bits */
if (s->last_bitoffset > 0)
skip_bits(&s->gb, s->last_bitoffset);
/* this frame is stored in the last superframe and in the
current one */
if (wma_decode_frame(s, samples, samples_offset) < 0)
goto fail;
samples_offset += s->frame_len;
nb_frames--;
}
/* read each frame starting from bit_offset */
pos = bit_offset + 4 + 4 + s->byte_offset_bits + 3;
if (pos >= MAX_CODED_SUPERFRAME_SIZE * 8 || pos > buf_size * 8)
return AVERROR_INVALIDDATA;
init_get_bits(&s->gb, buf + (pos >> 3), (buf_size - (pos >> 3))*8);
len = pos & 7;
if (len > 0)
skip_bits(&s->gb, len);
s->reset_block_lengths = 1;
for(i=0;i<nb_frames;i++) {
if (wma_decode_frame(s, samples, samples_offset) < 0)
goto fail;
samples_offset += s->frame_len;
}
/* we copy the end of the frame in the last frame buffer */
pos = get_bits_count(&s->gb) + ((bit_offset + 4 + 4 + s->byte_offset_bits + 3) & ~7);
s->last_bitoffset = pos & 7;
pos >>= 3;
len = buf_size - pos;
if (len > MAX_CODED_SUPERFRAME_SIZE || len < 0) {
av_log(s->avctx, AV_LOG_ERROR, "len %d invalid\n", len);
goto fail;
}
s->last_superframe_len = len;
memcpy(s->last_superframe, buf + pos, len);
} else {
/* single frame decode */
if (wma_decode_frame(s, samples, samples_offset) < 0)
goto fail;
samples_offset += s->frame_len;
}
av_dlog(s->avctx, "%d %d %d %d outbytes:%td eaten:%d\n",
s->frame_len_bits, s->block_len_bits, s->frame_len, s->block_len,
(int8_t *)samples - (int8_t *)data, avctx->block_align);
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
return buf_size;
fail:
/* when error, we reset the bit reservoir */
s->last_superframe_len = 0;
return -1;
}
static av_cold void flush(AVCodecContext *avctx)
{
WMACodecContext *s = avctx->priv_data;
s->last_bitoffset=
s->last_superframe_len= 0;
}
#if CONFIG_WMAV1_DECODER
AVCodec ff_wmav1_decoder = {
.name = "wmav1",
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_WMAV1,
.priv_data_size = sizeof(WMACodecContext),
.init = wma_decode_init,
.close = ff_wma_end,
.decode = wma_decode_superframe,
.flush = flush,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 1"),
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
};
#endif
#if CONFIG_WMAV2_DECODER
AVCodec ff_wmav2_decoder = {
.name = "wmav2",
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_WMAV2,
.priv_data_size = sizeof(WMACodecContext),
.init = wma_decode_init,
.close = ff_wma_end,
.decode = wma_decode_superframe,
.flush = flush,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 2"),
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
};
#endif
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