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authorKenan Gillet <kenan.gillet@gmail.com>2008-10-24 21:29:23 +0000
committerDiego Biurrun <diego@biurrun.de>2008-10-24 21:29:23 +0000
commit4599d22c0cd7fa952bc02375d2899d8c9a31b9ae (patch)
tree23797fb97eb41bdfd65c7d2fbfdb7c5d1c0562dd /libavcodec/acelp_filters.h
parent60c25a4beb1d0df699104ecb8a536d9a09703d2f (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.h44
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).