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-rw-r--r--libavcodec/acelp_filters.c54
-rw-r--r--libavcodec/acelp_filters.h77
2 files changed, 131 insertions, 0 deletions
diff --git a/libavcodec/acelp_filters.c b/libavcodec/acelp_filters.c
index 7e33a674f1..2aedbddab3 100644
--- a/libavcodec/acelp_filters.c
+++ b/libavcodec/acelp_filters.c
@@ -27,6 +27,60 @@
#define FRAC_BITS 13
#include "mathops.h"
+const int16_t ff_acelp_interp_filter[61] =
+{ /* (0.15) */
+ 29443, 28346, 25207, 20449, 14701, 8693,
+ 3143, -1352, -4402, -5865, -5850, -4673,
+ -2783, -672, 1211, 2536, 3130, 2991,
+ 2259, 1170, 0, -1001, -1652, -1868,
+ -1666, -1147, -464, 218, 756, 1060,
+ 1099, 904, 550, 135, -245, -514,
+ -634, -602, -451, -231, 0, 191,
+ 308, 340, 296, 198, 78, -36,
+ -120, -163, -165, -132, -79, -19,
+ 34, 73, 91, 89, 70, 38,
+ 0,
+};
+
+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)
+{
+ int n, i;
+
+ assert(pitch_delay_frac >= 0 && pitch_delay_frac < precision);
+
+ for(n=0; n<length; n++)
+ {
+ int idx = 0;
+ int v = 0x4000;
+
+ for(i=0; i<filter_length;)
+ {
+
+ /* The reference G.729 and AMR fixed point code performs clipping after
+ each of the two following accumulations.
+ Since clipping affects only the synthetic OVERFLOW test without
+ causing an int type overflow, it was moved outside the loop. */
+
+ /* R(x):=ac_v[-k+x]
+ v += R(n-i)*ff_acelp_interp_filter(t+6i)
+ v += R(n+i+1)*ff_acelp_interp_filter(6-t+6i) */
+
+ v += in[n + i] * filter_coeffs[idx + pitch_delay_frac];
+ idx += precision;
+ i++;
+ v += in[n - i] * filter_coeffs[idx - pitch_delay_frac];
+ }
+ out[n] = av_clip_int16(v >> 15);
+ }
+}
+
void ff_acelp_convolve_circ(
int16_t* fc_out,
const int16_t* fc_in,
diff --git a/libavcodec/acelp_filters.h b/libavcodec/acelp_filters.h
index 9e497caec6..167faf72c3 100644
--- a/libavcodec/acelp_filters.h
+++ b/libavcodec/acelp_filters.h
@@ -26,6 +26,83 @@
#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.
+ *
+ */
+extern const int16_t ff_acelp_interp_filter[61];
+
+/**
+ * \brief Generic 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 filter_length filter length
+ * \param length length of speech data to process
+ *
+ * filter_coeffs contains coefficients of the positive half of the symmetric
+ * interpolation filter. filter_coeffs[0] should the central (unpaired) coefficient.
+ * See ff_acelp_interp_filter fot 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);
+
+/**
* \brief 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