Mercurial > libavcodec.hg
changeset 8049:611a21e4b01b libavcodec
Split off celp_filters.[ch] from acelp_filters.[ch] for the QCELP decoder.
patch by Kenan Gillet, kenan.gillet gmail com
| author | diego |
|---|---|
| date | Fri, 24 Oct 2008 21:29:23 +0000 |
| parents | ecb1962c12f3 |
| children | 4eac1822bc65 |
| files | Makefile acelp_filters.c acelp_filters.h celp_filters.c celp_filters.h ra144.c |
| diffstat | 6 files changed, 162 insertions(+), 107 deletions(-) [+] |
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--- a/Makefile Fri Oct 24 21:20:29 2008 +0000 +++ b/Makefile Fri Oct 24 21:29:23 2008 +0000 @@ -156,7 +156,7 @@ OBJS-$(CONFIG_QPEG_DECODER) += qpeg.o OBJS-$(CONFIG_QTRLE_DECODER) += qtrle.o OBJS-$(CONFIG_QTRLE_ENCODER) += qtrleenc.o -OBJS-$(CONFIG_RA_144_DECODER) += ra144.o acelp_filters.o +OBJS-$(CONFIG_RA_144_DECODER) += ra144.o celp_filters.o OBJS-$(CONFIG_RA_288_DECODER) += ra288.o OBJS-$(CONFIG_RAWVIDEO_DECODER) += rawdec.o OBJS-$(CONFIG_RAWVIDEO_ENCODER) += rawenc.o
--- a/acelp_filters.c Fri Oct 24 21:20:29 2008 +0000 +++ b/acelp_filters.c Fri Oct 24 21:29:23 2008 +0000 @@ -81,65 +81,6 @@ } } -void ff_acelp_convolve_circ( - int16_t* fc_out, - const int16_t* fc_in, - const int16_t* filter, - int len) -{ - int i, k; - - memset(fc_out, 0, len * sizeof(int16_t)); - - /* Since there are few pulses over an entire subframe (i.e. almost - all fc_in[i] are zero) it is faster to loop over fc_in first. */ - for(i=0; i<len; i++) - { - if(fc_in[i]) - { - for(k=0; k<i; k++) - fc_out[k] += (fc_in[i] * filter[len + k - i]) >> 15; - - for(k=i; k<len; k++) - fc_out[k] += (fc_in[i] * filter[ k - i]) >> 15; - } - } -} - -int ff_acelp_lp_synthesis_filter( - int16_t *out, - const int16_t* filter_coeffs, - const int16_t* in, - int buffer_length, - int filter_length, - int stop_on_overflow, - int rounder) -{ - int i,n; - - // These two lines are to avoid a -1 subtraction in the main loop - filter_length++; - filter_coeffs--; - - for(n=0; n<buffer_length; n++) - { - int sum = rounder; - for(i=1; i<filter_length; i++) - sum -= filter_coeffs[i] * out[n-i]; - - sum = (sum >> 12) + in[n]; - - if(sum + 0x8000 > 0xFFFFU) - { - if(stop_on_overflow) - return 1; - sum = (sum >> 31) ^ 32767; - } - out[n] = sum; - } - - return 0; -} void ff_acelp_high_pass_filter( int16_t* out,
--- a/acelp_filters.h Fri Oct 24 21:20:29 2008 +0000 +++ b/acelp_filters.h Fri Oct 24 21:29:23 2008 +0000 @@ -60,50 +60,6 @@ int filter_length, int length); -/** - * Circularly convolve fixed vector with a phase dispersion impulse - * response filter (D.6.2 of G.729 and 6.1.5 of AMR). - * @param fc_out vector with filter applied - * @param fc_in source vector - * @param filter phase filter coefficients - * - * fc_out[n] = sum(i,0,len-1){ fc_in[i] * filter[(len + n - i)%len] } - * - * \note fc_in and fc_out should not overlap! - */ -void ff_acelp_convolve_circ( - int16_t* fc_out, - const int16_t* fc_in, - const int16_t* filter, - int len); - -/** - * LP synthesis filter. - * @param out [out] pointer to output buffer - * @param filter_coeffs filter coefficients (-0x8000 <= (3.12) < 0x8000) - * @param in input signal - * @param buffer_length amount of data to process - * @param filter_length filter length (10 for 10th order LP filter) - * @param stop_on_overflow 1 - return immediately if overflow occurs - * 0 - ignore overflows - * @param rounder the amount to add for rounding (usually 0x800 or 0xfff) - * - * @return 1 if overflow occurred, 0 - otherwise - * - * @note Output buffer must contain 10 samples of past - * speech data before pointer. - * - * Routine applies 1/A(z) filter to given speech data. - */ -int ff_acelp_lp_synthesis_filter( - int16_t *out, - const int16_t* filter_coeffs, - const int16_t* in, - int buffer_length, - int filter_length, - int stop_on_overflow, - int rounder); - /** * high-pass filtering and upscaling (4.2.5 of G.729).
--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/celp_filters.c Fri Oct 24 21:29:23 2008 +0000 @@ -0,0 +1,86 @@ +/* + * various filters for ACELP-based codecs + * + * Copyright (c) 2008 Vladimir Voroshilov + * + * 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 + */ + +#include <inttypes.h> + +#include "avcodec.h" +#include "celp_filters.h" + +void ff_celp_convolve_circ( + int16_t* fc_out, + const int16_t* fc_in, + const int16_t* filter, + int len) +{ + int i, k; + + memset(fc_out, 0, len * sizeof(int16_t)); + + /* Since there are few pulses over an entire subframe (i.e. almost + all fc_in[i] are zero) it is faster to loop over fc_in first. */ + for(i=0; i<len; i++) + { + if(fc_in[i]) + { + for(k=0; k<i; k++) + fc_out[k] += (fc_in[i] * filter[len + k - i]) >> 15; + + for(k=i; k<len; k++) + fc_out[k] += (fc_in[i] * filter[ k - i]) >> 15; + } + } +} + +int ff_celp_lp_synthesis_filter( + int16_t *out, + const int16_t* filter_coeffs, + const int16_t* in, + int buffer_length, + int filter_length, + int stop_on_overflow, + int rounder) +{ + int i,n; + + // These two lines are to avoid a -1 subtraction in the main loop + filter_length++; + filter_coeffs--; + + for(n=0; n<buffer_length; n++) + { + int sum = rounder; + for(i=1; i<filter_length; i++) + sum -= filter_coeffs[i] * out[n-i]; + + sum = (sum >> 12) + in[n]; + + if(sum + 0x8000 > 0xFFFFU) + { + if(stop_on_overflow) + return 1; + sum = (sum >> 31) ^ 32767; + } + out[n] = sum; + } + + return 0; +}
--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/celp_filters.h Fri Oct 24 21:29:23 2008 +0000 @@ -0,0 +1,72 @@ +/* + * various filters for CELP-based codecs + * + * Copyright (c) 2008 Vladimir Voroshilov + * + * 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_CELP_FILTERS_H +#define AVCODEC_CELP_FILTERS_H + +#include <stdint.h> + +/** + * Circularly convolve fixed vector with a phase dispersion impulse + * response filter (D.6.2 of G.729 and 6.1.5 of AMR). + * @param fc_out vector with filter applied + * @param fc_in source vector + * @param filter phase filter coefficients + * + * fc_out[n] = sum(i,0,len-1){ fc_in[i] * filter[(len + n - i)%len] } + * + * \note fc_in and fc_out should not overlap! + */ +void ff_celp_convolve_circ( + int16_t* fc_out, + const int16_t* fc_in, + const int16_t* filter, + int len); + +/** + * LP synthesis filter. + * @param out [out] pointer to output buffer + * @param filter_coeffs filter coefficients (-0x8000 <= (3.12) < 0x8000) + * @param in input signal + * @param buffer_length amount of data to process + * @param filter_length filter length (10 for 10th order LP filter) + * @param stop_on_overflow 1 - return immediately if overflow occurs + * 0 - ignore overflows + * @param rounder the amount to add for rounding (usually 0x800 or 0xfff) + * + * @return 1 if overflow occurred, 0 - otherwise + * + * @note Output buffer must contain 10 samples of past + * speech data before pointer. + * + * Routine applies 1/A(z) filter to given speech data. + */ +int ff_celp_lp_synthesis_filter( + int16_t *out, + const int16_t* filter_coeffs, + const int16_t* in, + int buffer_length, + int filter_length, + int stop_on_overflow, + int rounder); + +#endif /* AVCODEC_CELP_FILTERS_H */
--- a/ra144.c Fri Oct 24 21:20:29 2008 +0000 +++ b/ra144.c Fri Oct 24 21:29:23 2008 +0000 @@ -25,7 +25,7 @@ #include "avcodec.h" #include "bitstream.h" #include "ra144.h" -#include "acelp_filters.h" +#include "celp_filters.h" #define NBLOCKS 4 ///< number of subblocks within a block #define BLOCKSIZE 40 ///< subblock size in 16-bit words @@ -201,8 +201,8 @@ memcpy(ractx->curr_sblock, ractx->curr_sblock + 40, 10*sizeof(*ractx->curr_sblock)); - if (ff_acelp_lp_synthesis_filter(ractx->curr_sblock + 10, lpc_coefs, - block, BLOCKSIZE, 10, 1, 0xfff)) + if (ff_celp_lp_synthesis_filter(ractx->curr_sblock + 10, lpc_coefs, + block, BLOCKSIZE, 10, 1, 0xfff)) memset(ractx->curr_sblock, 0, 50*sizeof(*ractx->curr_sblock)); }