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https://gitee.com/openharmony/third_party_ffmpeg
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vorbis cosmetics: mdct0,mdct1 => mdct[2]
Originally committed as revision 5978 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Loren Merritt
parent
88db1a15c5
commit
0e0adae0ba
@ -161,8 +161,8 @@ static void vorbis_free(vorbis_context *vc) {
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av_freep(&vc->residues);
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av_freep(&vc->modes);
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ff_mdct_end(&vc->mdct0);
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ff_mdct_end(&vc->mdct1);
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ff_mdct_end(&vc->mdct[0]);
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ff_mdct_end(&vc->mdct[1]);
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for(i=0;i<vc->codebook_count;++i) {
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av_free(vc->codebooks[i].codevectors);
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@ -194,8 +194,8 @@ static void vorbis_free(vorbis_context *vc) {
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av_freep(&vc->mappings);
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if(vc->exp_bias){
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av_freep(&vc->swin);
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av_freep(&vc->lwin);
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av_freep(&vc->win[0]);
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av_freep(&vc->win[1]);
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}
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}
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@ -762,7 +762,7 @@ static void create_map( vorbis_context * vc, uint_fast8_t floor_number )
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for (blockflag=0;blockflag<2;++blockflag)
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{
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n=(blockflag ? vc->blocksize_1 : vc->blocksize_0) / 2;
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n=(blockflag ? vc->blocksize[1] : vc->blocksize[0]) / 2;
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floors[floor_number].data.t0.map[blockflag]=
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av_malloc((n+1) * sizeof(int_fast32_t)); // n+sentinel
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@ -877,33 +877,30 @@ static int vorbis_parse_id_hdr(vorbis_context *vc){
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vc->bitrate_minimum=get_bits_long_le(gb, 32);
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bl0=get_bits(gb, 4);
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bl1=get_bits(gb, 4);
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vc->blocksize_0=(1<<bl0);
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vc->blocksize_1=(1<<bl1);
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vc->blocksize[0]=(1<<bl0);
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vc->blocksize[1]=(1<<bl1);
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if (bl0>13 || bl0<6 || bl1>13 || bl1<6 || bl1<bl0) {
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av_log(vc->avccontext, AV_LOG_ERROR, " Vorbis id header packet corrupt (illegal blocksize). \n");
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return 3;
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}
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// output format int16
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if (vc->blocksize_1/2 * vc->audio_channels * 2 >
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if (vc->blocksize[1]/2 * vc->audio_channels * 2 >
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AVCODEC_MAX_AUDIO_FRAME_SIZE) {
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av_log(vc->avccontext, AV_LOG_ERROR, "Vorbis channel count makes "
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"output packets too large.\n");
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return 4;
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}
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vc->swin=vwin[bl0-6];
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vc->lwin=vwin[bl1-6];
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vc->win[0]=vwin[bl0-6];
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vc->win[1]=vwin[bl1-6];
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if(vc->exp_bias){
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int i;
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float *win;
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win = av_malloc(vc->blocksize_0/2 * sizeof(float));
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for(i=0; i<vc->blocksize_0/2; i++)
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win[i] = vc->swin[i] * (1<<15);
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vc->swin = win;
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win = av_malloc(vc->blocksize_1/2 * sizeof(float));
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for(i=0; i<vc->blocksize_1/2; i++)
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win[i] = vc->lwin[i] * (1<<15);
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vc->lwin = win;
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int i, j;
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for(j=0; j<2; j++){
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float *win = av_malloc(vc->blocksize[j]/2 * sizeof(float));
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for(i=0; i<vc->blocksize[j]/2; i++)
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win[i] = vc->win[j][i] * (1<<15);
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vc->win[j] = win;
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}
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}
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if ((get_bits1(gb)) == 0) {
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@ -911,24 +908,24 @@ static int vorbis_parse_id_hdr(vorbis_context *vc){
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return 2;
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}
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vc->channel_residues=(float *)av_malloc((vc->blocksize_1/2)*vc->audio_channels * sizeof(float));
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vc->channel_floors=(float *)av_malloc((vc->blocksize_1/2)*vc->audio_channels * sizeof(float));
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vc->saved=(float *)av_malloc((vc->blocksize_1/2)*vc->audio_channels * sizeof(float));
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vc->ret=(float *)av_malloc((vc->blocksize_1/2)*vc->audio_channels * sizeof(float));
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vc->buf=(float *)av_malloc(vc->blocksize_1 * sizeof(float));
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vc->buf_tmp=(float *)av_malloc(vc->blocksize_1 * sizeof(float));
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vc->channel_residues=(float *)av_malloc((vc->blocksize[1]/2)*vc->audio_channels * sizeof(float));
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vc->channel_floors=(float *)av_malloc((vc->blocksize[1]/2)*vc->audio_channels * sizeof(float));
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vc->saved=(float *)av_malloc((vc->blocksize[1]/2)*vc->audio_channels * sizeof(float));
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vc->ret=(float *)av_malloc((vc->blocksize[1]/2)*vc->audio_channels * sizeof(float));
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vc->buf=(float *)av_malloc(vc->blocksize[1] * sizeof(float));
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vc->buf_tmp=(float *)av_malloc(vc->blocksize[1] * sizeof(float));
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vc->saved_start=0;
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ff_mdct_init(&vc->mdct0, bl0, 1);
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ff_mdct_init(&vc->mdct1, bl1, 1);
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ff_mdct_init(&vc->mdct[0], bl0, 1);
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ff_mdct_init(&vc->mdct[1], bl1, 1);
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AV_DEBUG(" vorbis version %d \n audio_channels %d \n audio_samplerate %d \n bitrate_max %d \n bitrate_nom %d \n bitrate_min %d \n blk_0 %d blk_1 %d \n ",
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vc->version, vc->audio_channels, vc->audio_samplerate, vc->bitrate_maximum, vc->bitrate_nominal, vc->bitrate_minimum, vc->blocksize_0, vc->blocksize_1);
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vc->version, vc->audio_channels, vc->audio_samplerate, vc->bitrate_maximum, vc->bitrate_nominal, vc->bitrate_minimum, vc->blocksize[0], vc->blocksize[1]);
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/*
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BLK=vc->blocksize_0;
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BLK=vc->blocksize[0];
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for(i=0;i<BLK/2;++i) {
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vc->swin[i]=sin(0.5*3.14159265358*(sin(((float)i+0.5)/(float)BLK*3.14159265358))*(sin(((float)i+0.5)/(float)BLK*3.14159265358)));
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vc->win[0][i]=sin(0.5*3.14159265358*(sin(((float)i+0.5)/(float)BLK*3.14159265358))*(sin(((float)i+0.5)/(float)BLK*3.14159265358)));
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}
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*/
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@ -1545,7 +1542,7 @@ static int vorbis_parse_audio_packet(vorbis_context *vc) {
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next_window=get_bits1(gb);
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}
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blocksize=vc->modes[mode_number].blockflag ? vc->blocksize_1 : vc->blocksize_0;
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blocksize=vc->blocksize[vc->modes[mode_number].blockflag];
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memset(ch_res_ptr, 0, sizeof(float)*vc->audio_channels*blocksize/2); //FIXME can this be removed ?
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memset(ch_floor_ptr, 0, sizeof(float)*vc->audio_channels*blocksize/2); //FIXME can this be removed ?
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@ -1618,10 +1615,10 @@ static int vorbis_parse_audio_packet(vorbis_context *vc) {
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for(j=0;j<vc->audio_channels;++j) {
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uint_fast8_t step=vc->audio_channels;
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uint_fast16_t k;
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float *saved=vc->saved+j*vc->blocksize_1/2;
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float *saved=vc->saved+j*vc->blocksize[1]/2;
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float *ret=vc->ret;
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const float *lwin=vc->lwin;
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const float *swin=vc->swin;
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const float *lwin=vc->win[1];
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const float *swin=vc->win[0];
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float *buf=vc->buf;
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float *buf_tmp=vc->buf_tmp;
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@ -1629,20 +1626,20 @@ static int vorbis_parse_audio_packet(vorbis_context *vc) {
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saved_start=vc->saved_start;
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vc->mdct0.fft.imdct_calc(vc->modes[mode_number].blockflag ? &vc->mdct1 : &vc->mdct0, buf, ch_floor_ptr, buf_tmp);
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vc->mdct[0].fft.imdct_calc(&vc->mdct[vc->modes[mode_number].blockflag], buf, ch_floor_ptr, buf_tmp);
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//FIXME process channels together, to allow faster simd vector_fmul_add_add?
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if (vc->modes[mode_number].blockflag) {
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// -- overlap/add
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if (previous_window) {
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vc->dsp.vector_fmul_add_add(ret+j, buf, lwin, saved, vc->add_bias, vc->blocksize_1/2, step);
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retlen=vc->blocksize_1/2;
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vc->dsp.vector_fmul_add_add(ret+j, buf, lwin, saved, vc->add_bias, vc->blocksize[1]/2, step);
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retlen=vc->blocksize[1]/2;
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} else {
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int len = (vc->blocksize_1-vc->blocksize_0)/4;
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int len = (vc->blocksize[1]-vc->blocksize[0])/4;
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buf += len;
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vc->dsp.vector_fmul_add_add(ret+j, buf, swin, saved, vc->add_bias, vc->blocksize_0/2, step);
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k = vc->blocksize_0/2*step + j;
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buf += vc->blocksize_0/2;
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vc->dsp.vector_fmul_add_add(ret+j, buf, swin, saved, vc->add_bias, vc->blocksize[0]/2, step);
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k = vc->blocksize[0]/2*step + j;
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buf += vc->blocksize[0]/2;
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if(vc->exp_bias){
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for(i=0; i<len; i++, k+=step)
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((uint32_t*)ret)[k] = ((uint32_t*)buf)[i] + vc->exp_bias; // ret[k]=buf[i]*(1<<bias)
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@ -1651,19 +1648,19 @@ static int vorbis_parse_audio_packet(vorbis_context *vc) {
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ret[k] = buf[i] + fadd_bias;
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}
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buf=vc->buf;
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retlen=vc->blocksize_0/2+len;
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retlen=vc->blocksize[0]/2+len;
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}
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// -- save
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if (next_window) {
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buf += vc->blocksize_1/2;
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vc->dsp.vector_fmul_reverse(saved, buf, lwin, vc->blocksize_1/2);
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buf += vc->blocksize[1]/2;
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vc->dsp.vector_fmul_reverse(saved, buf, lwin, vc->blocksize[1]/2);
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saved_start=0;
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} else {
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saved_start=(vc->blocksize_1-vc->blocksize_0)/4;
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buf += vc->blocksize_1/2;
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saved_start=(vc->blocksize[1]-vc->blocksize[0])/4;
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buf += vc->blocksize[1]/2;
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for(i=0; i<saved_start; i++)
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((uint32_t*)saved)[i] = ((uint32_t*)buf)[i] + vc->exp_bias;
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vc->dsp.vector_fmul_reverse(saved+saved_start, buf+saved_start, swin, vc->blocksize_0/2);
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vc->dsp.vector_fmul_reverse(saved+saved_start, buf+saved_start, swin, vc->blocksize[0]/2);
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}
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} else {
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// --overlap/add
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@ -1674,11 +1671,11 @@ static int vorbis_parse_audio_packet(vorbis_context *vc) {
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for(k=j, i=0;i<saved_start;++i, k+=step)
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ret[k] = saved[i];
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}
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vc->dsp.vector_fmul_add_add(ret+k, buf, swin, saved+saved_start, vc->add_bias, vc->blocksize_0/2, step);
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retlen=saved_start+vc->blocksize_0/2;
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vc->dsp.vector_fmul_add_add(ret+k, buf, swin, saved+saved_start, vc->add_bias, vc->blocksize[0]/2, step);
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retlen=saved_start+vc->blocksize[0]/2;
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// -- save
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buf += vc->blocksize_0/2;
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vc->dsp.vector_fmul_reverse(saved, buf, swin, vc->blocksize_0/2);
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buf += vc->blocksize[0]/2;
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vc->dsp.vector_fmul_reverse(saved, buf, swin, vc->blocksize[0]/2);
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saved_start=0;
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}
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}
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@ -89,8 +89,7 @@ typedef struct vorbis_context_s {
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GetBitContext gb;
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DSPContext dsp;
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MDCTContext mdct0;
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MDCTContext mdct1;
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MDCTContext mdct[2];
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uint_fast8_t first_frame;
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uint_fast32_t version;
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uint_fast8_t audio_channels;
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@ -98,10 +97,8 @@ typedef struct vorbis_context_s {
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uint_fast32_t bitrate_maximum;
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uint_fast32_t bitrate_nominal;
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uint_fast32_t bitrate_minimum;
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uint_fast32_t blocksize_0;
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uint_fast32_t blocksize_1;
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const float * swin;
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const float * lwin;
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uint_fast32_t blocksize[2];
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const float * win[2];
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uint_fast16_t codebook_count;
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vorbis_codebook *codebooks;
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uint_fast8_t floor_count;
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