X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Facelp_filters.h;h=8b6060fa1502974d57b14f8dfdfb5255a81e1edf;hb=fc4d335f6d5c4f211505b79371a8588a48376d21;hp=8c0689211564e67870993f140b1a234d9223af0b;hpb=40a655e306d253b393f465a437be566d28f4b25e;p=ffmpeg

diff --git a/libavcodec/acelp_filters.h b/libavcodec/acelp_filters.h
index 8c068921156..8b6060fa150 100644
--- a/libavcodec/acelp_filters.h
+++ b/libavcodec/acelp_filters.h
@@ -25,6 +25,83 @@
 
 #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 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).
@@ -48,9 +125,10 @@ void ff_acelp_convolve_circ(
  * \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
  *
@@ -65,7 +143,8 @@ int ff_acelp_lp_synthesis_filter(
         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.
@@ -112,4 +191,4 @@ void ff_acelp_high_pass_filter(
         const int16_t* in,
         int length);
 
-#endif // FFMPEG_ACELP_FILTERS_H
+#endif /* FFMPEG_ACELP_FILTERS_H */