mirror of
https://gitee.com/openharmony/third_party_ffmpeg
synced 2024-11-24 11:49:48 +00:00
00c3b67b8a
Also break some long lines, remove codec function placeholder comments and add spaces in sample/pixel format lists. Signed-off-by: Martin Storsjö <martin@martin.st>
447 lines
14 KiB
C
447 lines
14 KiB
C
/*
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* WMA compatible encoder
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* Copyright (c) 2007 Michael Niedermayer
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*
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* This file is part of Libav.
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*
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* Libav is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* Libav is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with Libav; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "avcodec.h"
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#include "internal.h"
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#include "wma.h"
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#undef NDEBUG
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#include <assert.h>
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static int encode_init(AVCodecContext * avctx){
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WMACodecContext *s = avctx->priv_data;
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int i, flags1, flags2;
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uint8_t *extradata;
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s->avctx = avctx;
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if(avctx->channels > MAX_CHANNELS) {
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av_log(avctx, AV_LOG_ERROR, "too many channels: got %i, need %i or fewer",
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avctx->channels, MAX_CHANNELS);
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return AVERROR(EINVAL);
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}
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if (avctx->sample_rate > 48000) {
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av_log(avctx, AV_LOG_ERROR, "sample rate is too high: %d > 48kHz",
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avctx->sample_rate);
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return AVERROR(EINVAL);
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}
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if(avctx->bit_rate < 24*1000) {
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av_log(avctx, AV_LOG_ERROR, "bitrate too low: got %i, need 24000 or higher\n",
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avctx->bit_rate);
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return AVERROR(EINVAL);
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}
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/* extract flag infos */
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flags1 = 0;
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flags2 = 1;
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if (avctx->codec->id == CODEC_ID_WMAV1) {
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extradata= av_malloc(4);
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avctx->extradata_size= 4;
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AV_WL16(extradata, flags1);
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AV_WL16(extradata+2, flags2);
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} else if (avctx->codec->id == CODEC_ID_WMAV2) {
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extradata= av_mallocz(10);
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avctx->extradata_size= 10;
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AV_WL32(extradata, flags1);
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AV_WL16(extradata+4, flags2);
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}else
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assert(0);
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avctx->extradata= extradata;
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s->use_exp_vlc = flags2 & 0x0001;
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s->use_bit_reservoir = flags2 & 0x0002;
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s->use_variable_block_len = flags2 & 0x0004;
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if (avctx->channels == 2)
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s->ms_stereo = 1;
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ff_wma_init(avctx, flags2);
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/* init MDCT */
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for(i = 0; i < s->nb_block_sizes; i++)
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ff_mdct_init(&s->mdct_ctx[i], s->frame_len_bits - i + 1, 0, 1.0);
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s->block_align = avctx->bit_rate * (int64_t)s->frame_len /
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(avctx->sample_rate * 8);
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s->block_align = FFMIN(s->block_align, MAX_CODED_SUPERFRAME_SIZE);
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avctx->block_align = s->block_align;
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avctx->bit_rate = avctx->block_align * 8LL * avctx->sample_rate /
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s->frame_len;
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//av_log(NULL, AV_LOG_ERROR, "%d %d %d %d\n", s->block_align, avctx->bit_rate, s->frame_len, avctx->sample_rate);
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avctx->frame_size = avctx->delay = s->frame_len;
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#if FF_API_OLD_ENCODE_AUDIO
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avctx->coded_frame = &s->frame;
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avcodec_get_frame_defaults(avctx->coded_frame);
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#endif
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return 0;
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}
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static void apply_window_and_mdct(AVCodecContext * avctx, const signed short * audio, int len) {
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WMACodecContext *s = avctx->priv_data;
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int window_index= s->frame_len_bits - s->block_len_bits;
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FFTContext *mdct = &s->mdct_ctx[window_index];
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int i, j, channel;
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const float * win = s->windows[window_index];
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int window_len = 1 << s->block_len_bits;
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float n = window_len/2;
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for (channel = 0; channel < avctx->channels; channel++) {
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memcpy(s->output, s->frame_out[channel], sizeof(float)*window_len);
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j = channel;
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for (i = 0; i < len; i++, j += avctx->channels){
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s->output[i+window_len] = audio[j] / n * win[window_len - i - 1];
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s->frame_out[channel][i] = audio[j] / n * win[i];
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}
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mdct->mdct_calc(mdct, s->coefs[channel], s->output);
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}
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}
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//FIXME use for decoding too
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static void init_exp(WMACodecContext *s, int ch, const int *exp_param){
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int n;
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const uint16_t *ptr;
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float v, *q, max_scale, *q_end;
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ptr = s->exponent_bands[s->frame_len_bits - s->block_len_bits];
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q = s->exponents[ch];
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q_end = q + s->block_len;
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max_scale = 0;
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while (q < q_end) {
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/* XXX: use a table */
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v = pow(10, *exp_param++ * (1.0 / 16.0));
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max_scale= FFMAX(max_scale, v);
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n = *ptr++;
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do {
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*q++ = v;
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} while (--n);
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}
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s->max_exponent[ch] = max_scale;
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}
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static void encode_exp_vlc(WMACodecContext *s, int ch, const int *exp_param){
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int last_exp;
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const uint16_t *ptr;
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float *q, *q_end;
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ptr = s->exponent_bands[s->frame_len_bits - s->block_len_bits];
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q = s->exponents[ch];
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q_end = q + s->block_len;
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if (s->version == 1) {
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last_exp= *exp_param++;
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assert(last_exp-10 >= 0 && last_exp-10 < 32);
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put_bits(&s->pb, 5, last_exp - 10);
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q+= *ptr++;
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}else
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last_exp = 36;
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while (q < q_end) {
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int exp = *exp_param++;
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int code = exp - last_exp + 60;
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assert(code >= 0 && code < 120);
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put_bits(&s->pb, ff_aac_scalefactor_bits[code], ff_aac_scalefactor_code[code]);
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/* XXX: use a table */
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q+= *ptr++;
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last_exp= exp;
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}
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}
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static int encode_block(WMACodecContext *s, float (*src_coefs)[BLOCK_MAX_SIZE], int total_gain){
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int v, bsize, ch, coef_nb_bits, parse_exponents;
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float mdct_norm;
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int nb_coefs[MAX_CHANNELS];
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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};
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//FIXME remove duplication relative to decoder
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if (s->use_variable_block_len) {
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assert(0); //FIXME not implemented
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}else{
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/* fixed block len */
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s->next_block_len_bits = s->frame_len_bits;
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s->prev_block_len_bits = s->frame_len_bits;
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s->block_len_bits = s->frame_len_bits;
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}
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s->block_len = 1 << s->block_len_bits;
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// assert((s->block_pos + s->block_len) <= s->frame_len);
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bsize = s->frame_len_bits - s->block_len_bits;
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//FIXME factor
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v = s->coefs_end[bsize] - s->coefs_start;
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for(ch = 0; ch < s->nb_channels; ch++)
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nb_coefs[ch] = v;
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{
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int n4 = s->block_len / 2;
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mdct_norm = 1.0 / (float)n4;
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if (s->version == 1) {
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mdct_norm *= sqrt(n4);
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}
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}
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if (s->nb_channels == 2) {
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put_bits(&s->pb, 1, !!s->ms_stereo);
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}
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for(ch = 0; ch < s->nb_channels; ch++) {
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s->channel_coded[ch] = 1; //FIXME only set channel_coded when needed, instead of always
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if (s->channel_coded[ch]) {
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init_exp(s, ch, fixed_exp);
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}
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}
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for(ch = 0; ch < s->nb_channels; ch++) {
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if (s->channel_coded[ch]) {
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WMACoef *coefs1;
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float *coefs, *exponents, mult;
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int i, n;
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coefs1 = s->coefs1[ch];
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exponents = s->exponents[ch];
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mult = pow(10, total_gain * 0.05) / s->max_exponent[ch];
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mult *= mdct_norm;
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coefs = src_coefs[ch];
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if (s->use_noise_coding && 0) {
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assert(0); //FIXME not implemented
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} else {
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coefs += s->coefs_start;
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n = nb_coefs[ch];
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for(i = 0;i < n; i++){
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double t= *coefs++ / (exponents[i] * mult);
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if(t<-32768 || t>32767)
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return -1;
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coefs1[i] = lrint(t);
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}
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}
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}
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}
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v = 0;
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for(ch = 0; ch < s->nb_channels; ch++) {
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int a = s->channel_coded[ch];
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put_bits(&s->pb, 1, a);
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v |= a;
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}
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if (!v)
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return 1;
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for(v= total_gain-1; v>=127; v-= 127)
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put_bits(&s->pb, 7, 127);
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put_bits(&s->pb, 7, v);
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coef_nb_bits= ff_wma_total_gain_to_bits(total_gain);
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if (s->use_noise_coding) {
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for(ch = 0; ch < s->nb_channels; ch++) {
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if (s->channel_coded[ch]) {
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int i, n;
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n = s->exponent_high_sizes[bsize];
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for(i=0;i<n;i++) {
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put_bits(&s->pb, 1, s->high_band_coded[ch][i]= 0);
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if (0)
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nb_coefs[ch] -= s->exponent_high_bands[bsize][i];
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}
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}
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}
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}
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parse_exponents = 1;
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if (s->block_len_bits != s->frame_len_bits) {
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put_bits(&s->pb, 1, parse_exponents);
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}
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if (parse_exponents) {
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for(ch = 0; ch < s->nb_channels; ch++) {
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if (s->channel_coded[ch]) {
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if (s->use_exp_vlc) {
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encode_exp_vlc(s, ch, fixed_exp);
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} else {
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assert(0); //FIXME not implemented
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// encode_exp_lsp(s, ch);
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}
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}
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}
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} else {
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assert(0); //FIXME not implemented
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}
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for(ch = 0; ch < s->nb_channels; ch++) {
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if (s->channel_coded[ch]) {
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int run, tindex;
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WMACoef *ptr, *eptr;
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tindex = (ch == 1 && s->ms_stereo);
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ptr = &s->coefs1[ch][0];
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eptr = ptr + nb_coefs[ch];
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run=0;
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for(;ptr < eptr; ptr++){
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if(*ptr){
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int level= *ptr;
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int abs_level= FFABS(level);
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int code= 0;
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if(abs_level <= s->coef_vlcs[tindex]->max_level){
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if(run < s->coef_vlcs[tindex]->levels[abs_level-1])
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code= run + s->int_table[tindex][abs_level-1];
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}
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assert(code < s->coef_vlcs[tindex]->n);
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put_bits(&s->pb, s->coef_vlcs[tindex]->huffbits[code], s->coef_vlcs[tindex]->huffcodes[code]);
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if(code == 0){
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if(1<<coef_nb_bits <= abs_level)
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return -1;
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put_bits(&s->pb, coef_nb_bits, abs_level);
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put_bits(&s->pb, s->frame_len_bits, run);
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}
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put_bits(&s->pb, 1, level < 0); //FIXME the sign is fliped somewhere
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run=0;
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}else{
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run++;
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}
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}
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if(run)
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put_bits(&s->pb, s->coef_vlcs[tindex]->huffbits[1], s->coef_vlcs[tindex]->huffcodes[1]);
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}
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if (s->version == 1 && s->nb_channels >= 2) {
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avpriv_align_put_bits(&s->pb);
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}
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}
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return 0;
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}
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static int encode_frame(WMACodecContext *s, float (*src_coefs)[BLOCK_MAX_SIZE], uint8_t *buf, int buf_size, int total_gain){
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init_put_bits(&s->pb, buf, buf_size);
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if (s->use_bit_reservoir) {
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assert(0);//FIXME not implemented
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}else{
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if(encode_block(s, src_coefs, total_gain) < 0)
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return INT_MAX;
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}
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avpriv_align_put_bits(&s->pb);
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return put_bits_count(&s->pb)/8 - s->block_align;
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}
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static int encode_superframe(AVCodecContext *avctx, AVPacket *avpkt,
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const AVFrame *frame, int *got_packet_ptr)
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{
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WMACodecContext *s = avctx->priv_data;
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const int16_t *samples = (const int16_t *)frame->data[0];
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int i, total_gain, ret;
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s->block_len_bits= s->frame_len_bits; //required by non variable block len
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s->block_len = 1 << s->block_len_bits;
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apply_window_and_mdct(avctx, samples, frame->nb_samples);
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if (s->ms_stereo) {
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float a, b;
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int i;
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for(i = 0; i < s->block_len; i++) {
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a = s->coefs[0][i]*0.5;
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b = s->coefs[1][i]*0.5;
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s->coefs[0][i] = a + b;
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s->coefs[1][i] = a - b;
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}
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}
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if ((ret = ff_alloc_packet(avpkt, 2 * MAX_CODED_SUPERFRAME_SIZE))) {
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av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
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return ret;
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}
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#if 1
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total_gain= 128;
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for(i=64; i; i>>=1){
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int error = encode_frame(s, s->coefs, avpkt->data, avpkt->size,
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total_gain - i);
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if(error<0)
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total_gain-= i;
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}
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#else
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total_gain= 90;
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best = encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain);
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for(i=32; i; i>>=1){
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int scoreL = encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain - i);
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int scoreR = encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain + i);
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av_log(NULL, AV_LOG_ERROR, "%d %d %d (%d)\n", scoreL, best, scoreR, total_gain);
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if(scoreL < FFMIN(best, scoreR)){
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best = scoreL;
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total_gain -= i;
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}else if(scoreR < best){
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best = scoreR;
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total_gain += i;
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}
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}
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#endif
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if ((i = encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain)) >= 0) {
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av_log(avctx, AV_LOG_ERROR, "required frame size too large. please "
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"use a higher bit rate.\n");
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return AVERROR(EINVAL);
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}
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assert((put_bits_count(&s->pb) & 7) == 0);
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while (i++)
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put_bits(&s->pb, 8, 'N');
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flush_put_bits(&s->pb);
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if (frame->pts != AV_NOPTS_VALUE)
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avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->delay);
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avpkt->size = s->block_align;
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*got_packet_ptr = 1;
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return 0;
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}
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AVCodec ff_wmav1_encoder = {
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.name = "wmav1",
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.type = AVMEDIA_TYPE_AUDIO,
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.id = CODEC_ID_WMAV1,
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.priv_data_size = sizeof(WMACodecContext),
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.init = encode_init,
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.encode2 = encode_superframe,
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.close = ff_wma_end,
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.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
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AV_SAMPLE_FMT_NONE },
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.long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 1"),
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};
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AVCodec ff_wmav2_encoder = {
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.name = "wmav2",
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.type = AVMEDIA_TYPE_AUDIO,
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.id = CODEC_ID_WMAV2,
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.priv_data_size = sizeof(WMACodecContext),
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.init = encode_init,
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.encode2 = encode_superframe,
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.close = ff_wma_end,
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.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
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AV_SAMPLE_FMT_NONE },
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.long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 2"),
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};
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