#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 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);
+
/**
* \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 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 (11 for 10th order LP filter)
+ * \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
*
const int16_t* in,
int buffer_length,
int filter_length,
- int stop_on_overflow);
+ int stop_on_overflow,
+ int rounder);
/**
* \brief Calculates coefficients of weighted A(z/weight) filter.
const int16_t* in,
int length);
-#endif // FFMPEG_ACELP_FILTERS_H
+#endif /* FFMPEG_ACELP_FILTERS_H */