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

1 files changed, 72 insertions, 0 deletions

diff --git a/libavcodec/celp_filters.h b/libavcodec/celp_filters.h
new file mode 100644
index 0000000000..cb73aa8869
--- /dev/null
+++ b/libavcodec/celp_filters.h
@@ -0,0 +1,72 @@
+/*
+ * various filters for CELP-based codecs
+ *
+ * Copyright (c) 2008 Vladimir Voroshilov
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg 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,
+ * 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
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_CELP_FILTERS_H
+#define AVCODEC_CELP_FILTERS_H
+
+#include <stdint.h>
+
+/**
+ * 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_celp_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_celp_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);
+
+#endif /* AVCODEC_CELP_FILTERS_H */