X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fresample2.c;h=48c20c2cbb18b0ac132089e4539379b3eaa5e8f3;hb=f769cfedd89db30e3d0fbf654138956779ea8053;hp=ed59448a49d2d3b5cf64dcf35f0cc3ca689f3a9e;hpb=d9526386990bccd61d8773e993e47fceb55b0174;p=ffmpeg diff --git a/libavcodec/resample2.c b/libavcodec/resample2.c index ed59448a49d..48c20c2cbb1 100644 --- a/libavcodec/resample2.c +++ b/libavcodec/resample2.c @@ -2,25 +2,25 @@ * audio resampling * Copyright (c) 2004 Michael Niedermayer * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * 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. * - * FFmpeg is distributed in the hope that it will be useful, + * 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 FFmpeg; if not, write to the Free Software + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** - * @file resample2.c + * @file * audio resampling * @author Michael Niedermayer */ @@ -57,6 +57,7 @@ typedef struct AVResampleContext{ + const AVClass *av_class; FELEM *filter_bank; int filter_length; int ideal_dst_incr; @@ -75,11 +76,13 @@ typedef struct AVResampleContext{ */ static double bessel(double x){ double v=1; + double lastv=0; double t=1; int i; x= x*x/4; - for(i=1; i<50; i++){ + for(i=1; v != lastv; i++){ + lastv=v; t *= x/(i*i); v += t; } @@ -87,16 +90,21 @@ static double bessel(double x){ } /** - * builds a polyphase filterbank. + * Build a polyphase filterbank. * @param factor resampling factor * @param scale wanted sum of coefficients for each filter * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16 + * @return 0 on success, negative on error */ -void av_build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){ +static int build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){ int ph, i; - double x, y, w, tab[tap_count]; + double x, y, w; + double *tab = av_malloc(tap_count * sizeof(*tab)); const int center= (tap_count-1)/2; + if (!tab) + return AVERROR(ENOMEM); + /* if upsampling, only need to interpolate, no filter */ if (factor > 1.0) factor = 1.0; @@ -173,24 +181,29 @@ void av_build_filter(FELEM *filter, double factor, int tap_count, int phase_coun } } #endif + + av_free(tab); + return 0; } -/** - * Initializes an audio resampler. - * Note, if either rate is not an integer then simply scale both rates up so they are. - */ AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){ AVResampleContext *c= av_mallocz(sizeof(AVResampleContext)); double factor= FFMIN(out_rate * cutoff / in_rate, 1.0); int phase_count= 1<phase_shift= phase_shift; c->phase_mask= phase_count-1; c->linear= linear; c->filter_length= FFMAX((int)ceil(filter_size/factor), 1); c->filter_bank= av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM)); - av_build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<filter_bank) + goto error; + if (build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM)); c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1]; @@ -199,6 +212,10 @@ AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size, c->index= -phase_count*((c->filter_length-1)/2); return c; +error: + av_free(c->filter_bank); + av_free(c); + return NULL; } void av_resample_close(AVResampleContext *c){ @@ -206,33 +223,12 @@ void av_resample_close(AVResampleContext *c){ av_freep(&c); } -/** - * Compensates samplerate/timestamp drift. The compensation is done by changing - * the resampler parameters, so no audible clicks or similar distortions occur - * @param compensation_distance distance in output samples over which the compensation should be performed - * @param sample_delta number of output samples which should be output less - * - * example: av_resample_compensate(c, 10, 500) - * here instead of 510 samples only 500 samples would be output - * - * note, due to rounding the actual compensation might be slightly different, - * especially if the compensation_distance is large and the in_rate used during init is small - */ void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){ // sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr; c->compensation_distance= compensation_distance; c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance; } -/** - * resamples. - * @param src an array of unconsumed samples - * @param consumed the number of samples of src which have been consumed are returned here - * @param src_size the number of unconsumed samples available - * @param dst_size the amount of space in samples available in dst - * @param update_ctx if this is 0 then the context wont be modified, that way several channels can be resampled with the same context - * @return the number of samples written in dst or -1 if an error occurred - */ int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){ int dst_index, i; int index= c->index;