From a67c21f093202f142438689d3f7cfbdf4ea82eea Mon Sep 17 00:00:00 2001 From: Lennart Poettering Date: Sun, 28 Oct 2007 19:13:50 +0000 Subject: merge 'lennart' branch back into trunk. git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@1971 fefdeb5f-60dc-0310-8127-8f9354f1896f --- src/pulsecore/ffmpeg/Makefile | 13 ++ src/pulsecore/ffmpeg/avcodec.h | 82 ++++++++++ src/pulsecore/ffmpeg/dsputil.h | 1 + src/pulsecore/ffmpeg/resample2.c | 324 +++++++++++++++++++++++++++++++++++++++ 4 files changed, 420 insertions(+) create mode 100644 src/pulsecore/ffmpeg/Makefile create mode 100644 src/pulsecore/ffmpeg/avcodec.h create mode 100644 src/pulsecore/ffmpeg/dsputil.h create mode 100644 src/pulsecore/ffmpeg/resample2.c (limited to 'src/pulsecore/ffmpeg') diff --git a/src/pulsecore/ffmpeg/Makefile b/src/pulsecore/ffmpeg/Makefile new file mode 100644 index 00000000..316beb72 --- /dev/null +++ b/src/pulsecore/ffmpeg/Makefile @@ -0,0 +1,13 @@ +# This is a dirty trick just to ease compilation with emacs +# +# This file is not intended to be distributed or anything +# +# So: don't touch it, even better ignore it! + +all: + $(MAKE) -C ../.. + +clean: + $(MAKE) -C ../.. clean + +.PHONY: all clean diff --git a/src/pulsecore/ffmpeg/avcodec.h b/src/pulsecore/ffmpeg/avcodec.h new file mode 100644 index 00000000..696fc986 --- /dev/null +++ b/src/pulsecore/ffmpeg/avcodec.h @@ -0,0 +1,82 @@ +/* + * copyright (c) 2001 Fabrice Bellard + * + * This file is part of FFmpeg. + * + * FFmpeg 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, + * 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 + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef AVCODEC_H +#define AVCODEC_H + +/* Just a heavily bastardized version of the original file from + * ffmpeg, just enough to get resample2.c to compile without + * modification -- Lennart */ + +#if !defined(PACKAGE) && defined(HAVE_CONFIG_H) +#include +#endif + +#include +#include +#include +#include +#include +#include + +#define av_mallocz(l) calloc(1, (l)) +#define av_malloc(l) malloc(l) +#define av_realloc(p,l) realloc((p),(l)) +#define av_free(p) free(p) + +static inline void av_freep(void *k) { + void **p = k; + + if (p) { + free(*p); + *p = NULL; + } +} + +static inline int av_clip(int a, int amin, int amax) +{ + if (a < amin) return amin; + else if (a > amax) return amax; + else return a; +} + +#define av_log(a,b,c) + +#define FFABS(a) ((a) >= 0 ? (a) : (-(a))) +#define FFSIGN(a) ((a) > 0 ? 1 : -1) + +#define FFMAX(a,b) ((a) > (b) ? (a) : (b)) +#define FFMIN(a,b) ((a) > (b) ? (b) : (a)) + +struct AVResampleContext; +struct AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_length, int log2_phase_count, int linear, double cutoff); +int av_resample(struct AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx); +void av_resample_compensate(struct AVResampleContext *c, int sample_delta, int compensation_distance); +void av_resample_close(struct AVResampleContext *c); +void av_build_filter(int16_t *filter, double factor, int tap_count, int phase_count, int scale, int type); + +/* + * crude lrintf for non-C99 systems. + */ +#ifndef HAVE_LRINTF +#define lrintf(x) ((long int)(x)) +#endif + +#endif /* AVCODEC_H */ diff --git a/src/pulsecore/ffmpeg/dsputil.h b/src/pulsecore/ffmpeg/dsputil.h new file mode 100644 index 00000000..8da742d0 --- /dev/null +++ b/src/pulsecore/ffmpeg/dsputil.h @@ -0,0 +1 @@ +/* empty file, just here to allow us to compile an unmodified resampler2.c */ diff --git a/src/pulsecore/ffmpeg/resample2.c b/src/pulsecore/ffmpeg/resample2.c new file mode 100644 index 00000000..da1443d9 --- /dev/null +++ b/src/pulsecore/ffmpeg/resample2.c @@ -0,0 +1,324 @@ +/* + * audio resampling + * Copyright (c) 2004 Michael Niedermayer + * + * This file is part of FFmpeg. + * + * FFmpeg 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, + * 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 + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +/** + * @file resample2.c + * audio resampling + * @author Michael Niedermayer + */ + +#include "avcodec.h" +#include "dsputil.h" + +#ifndef CONFIG_RESAMPLE_HP +#define FILTER_SHIFT 15 + +#define FELEM int16_t +#define FELEM2 int32_t +#define FELEML int64_t +#define FELEM_MAX INT16_MAX +#define FELEM_MIN INT16_MIN +#define WINDOW_TYPE 9 +#elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE) +#define FILTER_SHIFT 30 + +#define FELEM int32_t +#define FELEM2 int64_t +#define FELEML int64_t +#define FELEM_MAX INT32_MAX +#define FELEM_MIN INT32_MIN +#define WINDOW_TYPE 12 +#else +#define FILTER_SHIFT 0 + +#define FELEM double +#define FELEM2 double +#define FELEML double +#define WINDOW_TYPE 24 +#endif + + +typedef struct AVResampleContext{ + FELEM *filter_bank; + int filter_length; + int ideal_dst_incr; + int dst_incr; + int index; + int frac; + int src_incr; + int compensation_distance; + int phase_shift; + int phase_mask; + int linear; +}AVResampleContext; + +/** + * 0th order modified bessel function of the first kind. + */ +static double bessel(double x){ + double v=1; + double t=1; + int i; + + x= x*x/4; + for(i=1; i<50; i++){ + t *= x/(i*i); + v += t; + } + return v; +} + +/** + * builds 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 + */ +void av_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]; + const int center= (tap_count-1)/2; + + /* if upsampling, only need to interpolate, no filter */ + if (factor > 1.0) + factor = 1.0; + + for(ph=0;phphase_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[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]; + + c->src_incr= out_rate; + c->ideal_dst_incr= c->dst_incr= in_rate * phase_count; + c->index= -phase_count*((c->filter_length-1)/2); + + return c; +} + +void av_resample_close(AVResampleContext *c){ + av_freep(&c->filter_bank); + av_freep(&c); +} + +/** + * Compensates samplerate/timestamp drift. The compensation is done by changing + * the resampler parameters, so no audible clicks or similar distortions ocur + * @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 occured + */ +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; + int frac= c->frac; + int dst_incr_frac= c->dst_incr % c->src_incr; + int dst_incr= c->dst_incr / c->src_incr; + int compensation_distance= c->compensation_distance; + + if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){ + int64_t index2= ((int64_t)index)<<32; + int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr; + dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr); + + for(dst_index=0; dst_index < dst_size; dst_index++){ + dst[dst_index] = src[index2>>32]; + index2 += incr; + } + frac += dst_index * dst_incr_frac; + index += dst_index * dst_incr; + index += frac / c->src_incr; + frac %= c->src_incr; + }else{ + for(dst_index=0; dst_index < dst_size; dst_index++){ + FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask); + int sample_index= index >> c->phase_shift; + FELEM2 val=0; + + if(sample_index < 0){ + for(i=0; ifilter_length; i++) + val += src[FFABS(sample_index + i) % src_size] * filter[i]; + }else if(sample_index + c->filter_length > src_size){ + break; + }else if(c->linear){ + FELEM2 v2=0; + for(i=0; ifilter_length; i++){ + val += src[sample_index + i] * (FELEM2)filter[i]; + v2 += src[sample_index + i] * (FELEM2)filter[i + c->filter_length]; + } + val+=(v2-val)*(FELEML)frac / c->src_incr; + }else{ + for(i=0; ifilter_length; i++){ + val += src[sample_index + i] * (FELEM2)filter[i]; + } + } + +#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE + dst[dst_index] = av_clip_int16(lrintf(val)); +#else + val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT; + dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val; +#endif + + frac += dst_incr_frac; + index += dst_incr; + if(frac >= c->src_incr){ + frac -= c->src_incr; + index++; + } + + if(dst_index + 1 == compensation_distance){ + compensation_distance= 0; + dst_incr_frac= c->ideal_dst_incr % c->src_incr; + dst_incr= c->ideal_dst_incr / c->src_incr; + } + } + } + *consumed= FFMAX(index, 0) >> c->phase_shift; + if(index>=0) index &= c->phase_mask; + + if(compensation_distance){ + compensation_distance -= dst_index; + assert(compensation_distance > 0); + } + if(update_ctx){ + c->frac= frac; + c->index= index; + c->dst_incr= dst_incr_frac + c->src_incr*dst_incr; + c->compensation_distance= compensation_distance; + } +#if 0 + if(update_ctx && !c->compensation_distance){ +#undef rand + av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2); +av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance); + } +#endif + + return dst_index; +} -- cgit