RepoMirrors
/
ffmpeg
mirror of https://git.ffmpeg.org/ffmpeg.git
1
0
Fork 0
Code Issues Projects Releases Wiki Activity
ffmpeg/libavcodec/wmaenc.c

447 lines
14 KiB
C
Raw Blame History

/*
* 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 "internal.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;
if (avctx->channels == 2)
s->ms_stereo = 1;
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 = avctx->delay = s->frame_len;
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = &s->frame;
avcodec_get_frame_defaults(avctx->coded_frame);
#endif
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);
}
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;
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, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
WMACodecContext *s = avctx->priv_data;
const int16_t *samples = (const int16_t *)frame->data[0];
int i, total_gain, ret;
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, frame->nb_samples);
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 ((ret = ff_alloc_packet(avpkt, 2 * MAX_CODED_SUPERFRAME_SIZE))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
#if 1
total_gain= 128;
for(i=64; i; i>>=1){
int error = encode_frame(s, s->coefs, avpkt->data, avpkt->size,
total_gain - i);
if(error<0)
total_gain-= i;
}
#else
total_gain= 90;
best = encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain);
for(i=32; i; i>>=1){
int scoreL = encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain - i);
int scoreR = encode_frame(s, s->coefs, avpkt->data, avpkt->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
if ((i = encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain)) >= 0) {
av_log(avctx, AV_LOG_ERROR, "required frame size too large. please "
"use a higher bit rate.\n");
return AVERROR(EINVAL);
}
assert((put_bits_count(&s->pb) & 7) == 0);
while (i++)
put_bits(&s->pb, 8, 'N');
flush_put_bits(&s->pb);
if (frame->pts != AV_NOPTS_VALUE)
avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->delay);
avpkt->size = s->block_align;
*got_packet_ptr = 1;
return 0;
}
AVCodec ff_wmav1_encoder = {
.name = "wmav1",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_WMAV1,
.priv_data_size = sizeof(WMACodecContext),
.init = encode_init,
.encode2 = 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,
.encode2 = 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"),
};
Reference in New Issue View Git Blame Copy Permalink