diff options
author | Kenan Gillet <kenan.gillet@gmail.com> | 2008-10-24 21:29:23 +0000 |
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committer | Diego Biurrun <diego@biurrun.de> | 2008-10-24 21:29:23 +0000 |
commit | 4599d22c0cd7fa952bc02375d2899d8c9a31b9ae (patch) | |
tree | 23797fb97eb41bdfd65c7d2fbfdb7c5d1c0562dd /libavcodec/acelp_filters.h | |
parent | 60c25a4beb1d0df699104ecb8a536d9a09703d2f (diff) |
Split off celp_filters.[ch] from acelp_filters.[ch] for the QCELP decoder.
patch by Kenan Gillet, kenan.gillet gmail com
Originally committed as revision 15680 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavcodec/acelp_filters.h')
-rw-r--r-- | libavcodec/acelp_filters.h | 44 |
1 files changed, 0 insertions, 44 deletions
diff --git a/libavcodec/acelp_filters.h b/libavcodec/acelp_filters.h index b2f05bc9f0..e1e3d685ae 100644 --- a/libavcodec/acelp_filters.h +++ b/libavcodec/acelp_filters.h @@ -60,50 +60,6 @@ void ff_acelp_interpolate( 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). |