FFmpeg/libavcodec/wmaenc.c
Justin Ruggles 1ec075cfec wmaenc: limit allowed sample rate to 48kHz
ff_wma_init() allows up to 50kHz, but this generates an exponent band
size table that requires 65 bands. The code assumes 25 bands in many
places, and using sample rates higher than 48kHz will lead to buffer
overwrites.

CC:libav-stable@libav.org
2012-03-03 18:20:10 -05:00

426 lines
14 KiB
C

/*
* WMA compatible encoder
* Copyright (c) 2007 Michael Niedermayer
*
* 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
*/
#include "avcodec.h"
#include "wma.h"
#undef NDEBUG
#include <assert.h>
static int encode_init(AVCodecContext * avctx){
WMACodecContext *s = avctx->priv_data;
int i, flags1, flags2;
uint8_t *extradata;
s->avctx = avctx;
if(avctx->channels > MAX_CHANNELS) {
av_log(avctx, AV_LOG_ERROR, "too many channels: got %i, need %i or fewer",
avctx->channels, MAX_CHANNELS);
return AVERROR(EINVAL);
}
if (avctx->sample_rate > 48000) {
av_log(avctx, AV_LOG_ERROR, "sample rate is too high: %d > 48kHz",
avctx->sample_rate);
return AVERROR(EINVAL);
}
if(avctx->bit_rate < 24*1000) {
av_log(avctx, AV_LOG_ERROR, "bitrate too low: got %i, need 24000 or higher\n",
avctx->bit_rate);
return AVERROR(EINVAL);
}
/* extract flag infos */
flags1 = 0;
flags2 = 1;
if (avctx->codec->id == CODEC_ID_WMAV1) {
extradata= av_malloc(4);
avctx->extradata_size= 4;
AV_WL16(extradata, flags1);
AV_WL16(extradata+2, flags2);
} else if (avctx->codec->id == CODEC_ID_WMAV2) {
extradata= av_mallocz(10);
avctx->extradata_size= 10;
AV_WL32(extradata, flags1);
AV_WL16(extradata+4, flags2);
}else
assert(0);
avctx->extradata= extradata;
s->use_exp_vlc = flags2 & 0x0001;
s->use_bit_reservoir = flags2 & 0x0002;
s->use_variable_block_len = flags2 & 0x0004;
ff_wma_init(avctx, flags2);
/* init MDCT */
for(i = 0; i < s->nb_block_sizes; i++)
ff_mdct_init(&s->mdct_ctx[i], s->frame_len_bits - i + 1, 0, 1.0);
s->block_align = avctx->bit_rate * (int64_t)s->frame_len /
(avctx->sample_rate * 8);
s->block_align = FFMIN(s->block_align, MAX_CODED_SUPERFRAME_SIZE);
avctx->block_align = s->block_align;
avctx->bit_rate = avctx->block_align * 8LL * avctx->sample_rate /
s->frame_len;
//av_log(NULL, AV_LOG_ERROR, "%d %d %d %d\n", s->block_align, avctx->bit_rate, s->frame_len, avctx->sample_rate);
avctx->frame_size= s->frame_len;
return 0;
}
static void apply_window_and_mdct(AVCodecContext * avctx, const signed short * audio, int len) {
WMACodecContext *s = avctx->priv_data;
int window_index= s->frame_len_bits - s->block_len_bits;
FFTContext *mdct = &s->mdct_ctx[window_index];
int i, j, channel;
const float * win = s->windows[window_index];
int window_len = 1 << s->block_len_bits;
float n = window_len/2;
for (channel = 0; channel < avctx->channels; channel++) {
memcpy(s->output, s->frame_out[channel], sizeof(float)*window_len);
j = channel;
for (i = 0; i < len; i++, j += avctx->channels){
s->output[i+window_len] = audio[j] / n * win[window_len - i - 1];
s->frame_out[channel][i] = audio[j] / n * win[i];
}
mdct->mdct_calc(mdct, s->coefs[channel], s->output);
}
}
//FIXME use for decoding too
static void init_exp(WMACodecContext *s, int ch, const int *exp_param){
int n;
const uint16_t *ptr;
float v, *q, max_scale, *q_end;
ptr = s->exponent_bands[s->frame_len_bits - s->block_len_bits];
q = s->exponents[ch];
q_end = q + s->block_len;
max_scale = 0;
while (q < q_end) {
/* XXX: use a table */
v = pow(10, *exp_param++ * (1.0 / 16.0));
max_scale= FFMAX(max_scale, v);
n = *ptr++;
do {
*q++ = v;
} while (--n);
}
s->max_exponent[ch] = max_scale;
}
static void encode_exp_vlc(WMACodecContext *s, int ch, const int *exp_param){
int last_exp;
const uint16_t *ptr;
float *q, *q_end;
ptr = s->exponent_bands[s->frame_len_bits - s->block_len_bits];
q = s->exponents[ch];
q_end = q + s->block_len;
if (s->version == 1) {
last_exp= *exp_param++;
assert(last_exp-10 >= 0 && last_exp-10 < 32);
put_bits(&s->pb, 5, last_exp - 10);
q+= *ptr++;
}else
last_exp = 36;
while (q < q_end) {
int exp = *exp_param++;
int code = exp - last_exp + 60;
assert(code >= 0 && code < 120);
put_bits(&s->pb, ff_aac_scalefactor_bits[code], ff_aac_scalefactor_code[code]);
/* XXX: use a table */
q+= *ptr++;
last_exp= exp;
}
}
static int encode_block(WMACodecContext *s, float (*src_coefs)[BLOCK_MAX_SIZE], int total_gain){
int v, bsize, ch, coef_nb_bits, parse_exponents;
float mdct_norm;
int nb_coefs[MAX_CHANNELS];
static const int fixed_exp[25]={20,20,20,20,20,20,20,20,20,20,20,20,20,20,20,20,20,20,20,20,20,20,20,20,20};
//FIXME remove duplication relative to decoder
if (s->use_variable_block_len) {
assert(0); //FIXME not implemented
}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;
}
s->block_len = 1 << s->block_len_bits;
// assert((s->block_pos + s->block_len) <= s->frame_len);
bsize = s->frame_len_bits - s->block_len_bits;
//FIXME factor
v = s->coefs_end[bsize] - s->coefs_start;
for(ch = 0; ch < s->nb_channels; ch++)
nb_coefs[ch] = v;
{
int n4 = s->block_len / 2;
mdct_norm = 1.0 / (float)n4;
if (s->version == 1) {
mdct_norm *= sqrt(n4);
}
}
if (s->nb_channels == 2) {
put_bits(&s->pb, 1, s->ms_stereo= 1);
}
for(ch = 0; ch < s->nb_channels; ch++) {
s->channel_coded[ch] = 1; //FIXME only set channel_coded when needed, instead of always
if (s->channel_coded[ch]) {
init_exp(s, ch, fixed_exp);
}
}
for(ch = 0; ch < s->nb_channels; ch++) {
if (s->channel_coded[ch]) {
WMACoef *coefs1;
float *coefs, *exponents, mult;
int i, n;
coefs1 = s->coefs1[ch];
exponents = s->exponents[ch];
mult = pow(10, total_gain * 0.05) / s->max_exponent[ch];
mult *= mdct_norm;
coefs = src_coefs[ch];
if (s->use_noise_coding && 0) {
assert(0); //FIXME not implemented
} else {
coefs += s->coefs_start;
n = nb_coefs[ch];
for(i = 0;i < n; i++){
double t= *coefs++ / (exponents[i] * mult);
if(t<-32768 || t>32767)
return -1;
coefs1[i] = lrint(t);
}
}
}
}
v = 0;
for(ch = 0; ch < s->nb_channels; ch++) {
int a = s->channel_coded[ch];
put_bits(&s->pb, 1, a);
v |= a;
}
if (!v)
return 1;
for(v= total_gain-1; v>=127; v-= 127)
put_bits(&s->pb, 7, 127);
put_bits(&s->pb, 7, v);
coef_nb_bits= ff_wma_total_gain_to_bits(total_gain);
if (s->use_noise_coding) {
for(ch = 0; ch < s->nb_channels; ch++) {
if (s->channel_coded[ch]) {
int i, n;
n = s->exponent_high_sizes[bsize];
for(i=0;i<n;i++) {
put_bits(&s->pb, 1, s->high_band_coded[ch][i]= 0);
if (0)
nb_coefs[ch] -= s->exponent_high_bands[bsize][i];
}
}
}
}
parse_exponents = 1;
if (s->block_len_bits != s->frame_len_bits) {
put_bits(&s->pb, 1, parse_exponents);
}
if (parse_exponents) {
for(ch = 0; ch < s->nb_channels; ch++) {
if (s->channel_coded[ch]) {
if (s->use_exp_vlc) {
encode_exp_vlc(s, ch, fixed_exp);
} else {
assert(0); //FIXME not implemented
// encode_exp_lsp(s, ch);
}
}
}
} else {
assert(0); //FIXME not implemented
}
for(ch = 0; ch < s->nb_channels; ch++) {
if (s->channel_coded[ch]) {
int run, tindex;
WMACoef *ptr, *eptr;
tindex = (ch == 1 && s->ms_stereo);
ptr = &s->coefs1[ch][0];
eptr = ptr + nb_coefs[ch];
run=0;
for(;ptr < eptr; ptr++){
if(*ptr){
int level= *ptr;
int abs_level= FFABS(level);
int code= 0;
if(abs_level <= s->coef_vlcs[tindex]->max_level){
if(run < s->coef_vlcs[tindex]->levels[abs_level-1])
code= run + s->int_table[tindex][abs_level-1];
}
assert(code < s->coef_vlcs[tindex]->n);
put_bits(&s->pb, s->coef_vlcs[tindex]->huffbits[code], s->coef_vlcs[tindex]->huffcodes[code]);
if(code == 0){
if(1<<coef_nb_bits <= abs_level)
return -1;
//Workaround minor rounding differences for the regression tests, FIXME we should find and replace the problematic float by fixpoint for reg tests
if(abs_level == 0x71B && (s->avctx->flags & CODEC_FLAG_BITEXACT)) abs_level=0x71A;
put_bits(&s->pb, coef_nb_bits, abs_level);
put_bits(&s->pb, s->frame_len_bits, run);
}
put_bits(&s->pb, 1, level < 0); //FIXME the sign is fliped somewhere
run=0;
}else{
run++;
}
}
if(run)
put_bits(&s->pb, s->coef_vlcs[tindex]->huffbits[1], s->coef_vlcs[tindex]->huffcodes[1]);
}
if (s->version == 1 && s->nb_channels >= 2) {
avpriv_align_put_bits(&s->pb);
}
}
return 0;
}
static int encode_frame(WMACodecContext *s, float (*src_coefs)[BLOCK_MAX_SIZE], uint8_t *buf, int buf_size, int total_gain){
init_put_bits(&s->pb, buf, buf_size);
if (s->use_bit_reservoir) {
assert(0);//FIXME not implemented
}else{
if(encode_block(s, src_coefs, total_gain) < 0)
return INT_MAX;
}
avpriv_align_put_bits(&s->pb);
return put_bits_count(&s->pb)/8 - s->block_align;
}
static int encode_superframe(AVCodecContext *avctx,
unsigned char *buf, int buf_size, void *data){
WMACodecContext *s = avctx->priv_data;
const short *samples = data;
int i, total_gain;
s->block_len_bits= s->frame_len_bits; //required by non variable block len
s->block_len = 1 << s->block_len_bits;
apply_window_and_mdct(avctx, samples, avctx->frame_size);
if (s->ms_stereo) {
float a, b;
int i;
for(i = 0; i < s->block_len; i++) {
a = s->coefs[0][i]*0.5;
b = s->coefs[1][i]*0.5;
s->coefs[0][i] = a + b;
s->coefs[1][i] = a - b;
}
}
#if 1
total_gain= 128;
for(i=64; i; i>>=1){
int error= encode_frame(s, s->coefs, buf, buf_size, total_gain-i);
if(error<0)
total_gain-= i;
}
#else
total_gain= 90;
best= encode_frame(s, s->coefs, buf, buf_size, total_gain);
for(i=32; i; i>>=1){
int scoreL= encode_frame(s, s->coefs, buf, buf_size, total_gain-i);
int scoreR= encode_frame(s, s->coefs, buf, buf_size, total_gain+i);
av_log(NULL, AV_LOG_ERROR, "%d %d %d (%d)\n", scoreL, best, scoreR, total_gain);
if(scoreL < FFMIN(best, scoreR)){
best = scoreL;
total_gain -= i;
}else if(scoreR < best){
best = scoreR;
total_gain += i;
}
}
#endif
encode_frame(s, s->coefs, buf, buf_size, total_gain);
assert((put_bits_count(&s->pb) & 7) == 0);
i= s->block_align - (put_bits_count(&s->pb)+7)/8;
assert(i>=0);
while(i--)
put_bits(&s->pb, 8, 'N');
flush_put_bits(&s->pb);
return put_bits_ptr(&s->pb) - s->pb.buf;
}
AVCodec ff_wmav1_encoder = {
.name = "wmav1",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_WMAV1,
.priv_data_size = sizeof(WMACodecContext),
.init = encode_init,
.encode = encode_superframe,
.close = ff_wma_end,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 1"),
};
AVCodec ff_wmav2_encoder = {
.name = "wmav2",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_WMAV2,
.priv_data_size = sizeof(WMACodecContext),
.init = encode_init,
.encode = encode_superframe,
.close = ff_wma_end,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 2"),
};