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-rw-r--r--libavcodec/acelp_filters.c125
1 files changed, 125 insertions, 0 deletions
diff --git a/libavcodec/acelp_filters.c b/libavcodec/acelp_filters.c
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+++ b/libavcodec/acelp_filters.c
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+/*
+ * various filters for ACELP-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
+ */
+
+#include <inttypes.h>
+
+#include "avcodec.h"
+#include "acelp_filters.h"
+#define FRAC_BITS 13
+#include "mathops.h"
+
+void ff_acelp_convolve_circ(
+ int16_t* fc_out,
+ const int16_t* fc_in,
+ const int16_t* filter,
+ int subframe_size)
+{
+ int i, k;
+
+ memset(fc_out, 0, subframe_size * sizeof(int16_t));
+
+ /* Since there are few pulses over entire subframe (i.e. almost all
+ fc_in[i] are zero, in case of G.729D the buffer contains two non-zero
+ samples before the call to ff_acelp_enhance_harmonics, and (due to
+ pitch_delay bounded to [20; 143]) a maximum four non-zero samples
+ for a total of 40 after the call to it), it is faster to swap two loops
+ and process non-zero samples only. This will reduce the number of
+ multiplications from 40*40 to 4*40 for G.729D */
+ for(i=0; i<subframe_size; i++)
+ {
+ if(fc_in[i])
+ {
+ for(k=0; k<i; k++)
+ fc_out[k] += (fc_in[i] * filter[subframe_size + k - i]) >> 15;
+
+ for(k=i; k<subframe_size; k++)
+ fc_out[k] += (fc_in[i] * filter[k - i]) >> 15;
+ }
+ }
+}
+
+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 i,n;
+
+ for(n=0; n<buffer_length; n++)
+ {
+ int sum = 0x800;
+ for(i=1; i<filter_length; i++)
+ sum -= filter_coeffs[i] * out[n-i];
+
+ sum = (sum >> 12) + in[n];
+
+ /* Check for overflow */
+ if(sum + 0x8000 > 0xFFFFU)
+ {
+ if(stop_on_overflow)
+ return 1;
+ sum = (sum >> 31) ^ 32767;
+ }
+ out[n] = sum;
+ }
+
+ return 0;
+}
+
+void ff_acelp_weighted_filter(
+ int16_t *out,
+ const int16_t* in,
+ const int16_t *weight_pow,
+ int filter_length)
+{
+ int n;
+ for(n=0; n<filter_length; n++)
+ out[n] = (in[n] * weight_pow[n] + 0x4000) >> 15; /* (3.12) = (0.15) * (3.12) with rounding */
+}
+
+void ff_acelp_high_pass_filter(
+ int16_t* out,
+ int hpf_f[2],
+ const int16_t* in,
+ int length)
+{
+ int i;
+ int tmp;
+
+ for(i=0; i<length; i++)
+ {
+ tmp = MULL(hpf_f[0], 15836); /* (14.13) = (13.13) * (1.13) */
+ tmp += MULL(hpf_f[1], -7667); /* (13.13) = (13.13) * (0.13) */
+ tmp += 7699 * (in[i] - 2*in[i-1] + in[i-2]); /* (14.13) = (0.13) * (14.0) */
+
+ /* Multiplication by 2 with rounding can cause short type
+ overflow, thus clipping is required. */
+
+ out[i] = av_clip_int16((tmp + 0x800) >> 12); /* (15.0) = 2 * (13.13) = (14.13) */
+
+ hpf_f[1] = hpf_f[0];
+ hpf_f[0] = tmp;
+ }
+}
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