*
* Copyright (c) 2008 Vladimir Voroshilov
*
- * 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
*/
-#ifndef FFMPEG_ACELP_FILTERS_H
-#define FFMPEG_ACELP_FILTERS_H
+#ifndef AVCODEC_ACELP_FILTERS_H
+#define AVCODEC_ACELP_FILTERS_H
#include <stdint.h>
/**
- * low-pass FIR (Finite Impulse Response) filter coefficients
- *
- * A similar filter is named b30 in G.729.
- *
- * G.729 specification says:
- * b30 is based on Hamming windowed sinc functions, truncated at +/-29 and
- * padded with zeros at +/-30 b30[30]=0.
- * The filter has a cut-off frequency (-3 dB) at 3600 Hz in the oversampled
- * domain.
- *
- * After some analysis, I found this approximation:
- *
- * PI * x
- * Hamm(x,N) = 0.53836-0.46164*cos(--------)
- * N-1
- * ---
- * 2
- *
- * PI * x
- * Hamm'(x,k) = Hamm(x - k, 2*k+1) = 0.53836 + 0.46164*cos(--------)
- * k
- *
- * sin(PI * x)
- * Sinc(x) = ----------- (normalized sinc function)
- * PI * x
- *
- * h(t,B) = 2 * B * Sinc(2 * B * t) (impulse response of sinc low-pass filter)
- *
- * b(k,B, n) = Hamm'(n, k) * h(n, B)
- *
- *
- * 3600
- * B = ----
- * 8000
- *
- * 3600 - cut-off frequency
- * 8000 - sampling rate
- * k - filter order
- *
- * ff_acelp_interp_filter[6*i+j] = b(10, 3600/8000, i+j/6)
- *
- * The filter assumes the following order of fractions (X - integer delay):
- *
- * 1/3 precision: X 1/3 2/3 X 1/3 2/3 X
- * 1/6 precision: X 1/6 2/6 3/6 4/6 5/6 X 1/6 2/6 3/6 4/6 5/6 X
- *
- * The filter can be used for 1/3 precision, too, by
- * passing 2*pitch_delay_frac as third parameter to the interpolation routine.
+ * low-pass Finite Impulse Response filter coefficients.
*
+ * Hamming windowed sinc filter with cutoff freq 3/40 of the sampling freq,
+ * the coefficients are scaled by 2^15.
+ * This array only contains the right half of the filter.
+ * This filter is likely identical to the one used in G.729, though this
+ * could not be determined from the original comments with certainty.
*/
extern const int16_t ff_acelp_interp_filter[61];
/**
- * Generic interpolation routine.
- * @param out [out] buffer for interpolated data
+ * Generic FIR interpolation routine.
+ * @param[out] out buffer for interpolated data
* @param in input data
* @param filter_coeffs interpolation filter coefficients (0.15)
- * @param precision filter is able to interpolate with 1/precision precision of pitch delay
- * @param pitch_delay_frac pitch delay, fractional part [0..precision-1]
+ * @param precision sub sample factor, that is the precision of the position
+ * @param frac_pos fractional part of position [0..precision-1]
* @param filter_length filter length
- * @param length length of speech data to process
+ * @param length length of output
*
- * filter_coeffs contains coefficients of the positive half of the symmetric
+ * filter_coeffs contains coefficients of the right half of the symmetric
* interpolation filter. filter_coeffs[0] should the central (unpaired) coefficient.
* See ff_acelp_interp_filter for an example.
*
*/
-void ff_acelp_interpolate(
- int16_t* out,
- const int16_t* in,
- const int16_t* filter_coeffs,
- int precision,
- int pitch_delay_frac,
- 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 subframe_size);
+void ff_acelp_interpolate(int16_t* out, const int16_t* in,
+ const int16_t* filter_coeffs, int precision,
+ int frac_pos, int filter_length, int length);
/**
- * 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.
+ * Floating point version of ff_acelp_interpolate()
*/
-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);
+void ff_acelp_interpolatef(float *out, const float *in,
+ const float *filter_coeffs, int precision,
+ int frac_pos, int filter_length, int length);
-/**
- * Calculates coefficients of weighted A(z/weight) filter.
- * @param out [out] weighted A(z/weight) result
- * filter (-0x8000 <= (3.12) < 0x8000)
- * @param in source filter (-0x8000 <= (3.12) < 0x8000)
- * @param weight_pow array containing weight^i (-0x8000 <= (0.15) < 0x8000)
- * @param filter_length filter length (11 for 10th order LP filter)
- *
- * out[i]=weight_pow[i]*in[i] , i=0..9
- */
-void ff_acelp_weighted_filter(
- int16_t *out,
- const int16_t* in,
- const int16_t *weight_pow,
- int filter_length);
/**
* high-pass filtering and upscaling (4.2.5 of G.729).
- * @param out [out] output buffer for filtered speech data
- * @param hpf_f [in/out] past filtered data from previous (2 items long)
+ * @param[out] out output buffer for filtered speech data
+ * @param[in,out] hpf_f past filtered data from previous (2 items long)
* frames (-0x20000000 <= (14.13) < 0x20000000)
* @param in speech data to process
* @param length input data size
* out[i] = 0.93980581 * in[i] - 1.8795834 * in[i-1] + 0.93980581 * in[i-2] +
* 1.9330735 * out[i-1] - 0.93589199 * out[i-2]
*
- * The filter has a cut-off frequency of 100Hz
+ * The filter has a cut-off frequency of 1/80 of the sampling freq
*
* @note Two items before the top of the out buffer must contain two items from the
* tail of the previous subframe.
* fixed-point all coefficients are the same as in G.729. Thus this
* routine can be used for the fixed-point AMR decoder, too.
*/
-void ff_acelp_high_pass_filter(
- int16_t* out,
- int hpf_f[2],
- const int16_t* in,
- int length);
+void ff_acelp_high_pass_filter(int16_t* out, int hpf_f[2],
+ const int16_t* in, int length);
+
+/**
+ * Apply an order 2 rational transfer function in-place.
+ *
+ * @param out output buffer for filtered speech samples
+ * @param in input buffer containing speech data (may be the same as out)
+ * @param zero_coeffs z^-1 and z^-2 coefficients of the numerator
+ * @param pole_coeffs z^-1 and z^-2 coefficients of the denominator
+ * @param gain scale factor for final output
+ * @param mem intermediate values used by filter (should be 0 initially)
+ * @param n number of samples
+ */
+void ff_acelp_apply_order_2_transfer_function(float *out, const float *in,
+ const float zero_coeffs[2],
+ const float pole_coeffs[2],
+ float gain,
+ float mem[2], int n);
+
+/**
+ * Apply tilt compensation filter, 1 - tilt * z-1.
+ *
+ * @param mem pointer to the filter's state (one single float)
+ * @param tilt tilt factor
+ * @param samples array where the filter is applied
+ * @param size the size of the samples array
+ */
+void ff_tilt_compensation(float *mem, float tilt, float *samples, int size);
+
-#endif /* FFMPEG_ACELP_FILTERS_H */
+#endif /* AVCODEC_ACELP_FILTERS_H */
projects / ffmpeg / blobdiff