intended for use by RealAudio 2.0 (28.8k) and G.728 decoders. --- libavcodec/g728_template.c | 65 ++++++++++++++++++++++++++++++++++++++ libavcodec/ra288.c | 50 ++--------------------------- 2 files changed, 68 insertions(+), 47 deletions(-) create mode 100644 libavcodec/g728_template.c
diff --git a/libavcodec/g728_template.c b/libavcodec/g728_template.c
new file mode 100644
index 0000000000..72eb5fde80
--- /dev/null
+++ b/libavcodec/g728_template.c
@@ -0,0 +1,65 @@
+/*
+ * G.728 / RealAudio 2.0 (28.8K) decoder
+ *
+ * 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
+ */
+
+static void convolve(float *tgt, const float *src, int len, int n)
+{
+ for (; n >= 0; n--)
+ tgt[n] = avpriv_scalarproduct_float_c(src, src - n, len);
+
+}
+
+/**
+ * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
+ *
+ * @param order filter order
+ * @param n input length
+ * @param non_rec number of non-recursive samples
+ * @param out filter output
+ * @param hist pointer to the input history of the filter
+ * @param out pointer to the non-recursive part of the output
+ * @param out2 pointer to the recursive part of the output
+ * @param window pointer to the windowing function table
+ */
+static void do_hybrid_window(void (*vector_fmul)(float *dst, const float
*src0, const float *src1, int len),
+ int order, int n, int non_rec, float *out,
+ float *hist, float *out2, const float *window)
+{
+ int i;
+ float buffer1[MAX_BACKWARD_FILTER_ORDER + 1];
+ float buffer2[MAX_BACKWARD_FILTER_ORDER + 1];
+ LOCAL_ALIGNED(32, float, work, [FFALIGN(MAX_BACKWARD_FILTER_ORDER +
+ MAX_BACKWARD_FILTER_LEN +
+ MAX_BACKWARD_FILTER_NONREC, 16)]);
+
+ av_assert2(order>=0);
+
+ vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 16));
+
+ convolve(buffer1, work + order , n , order);
+ convolve(buffer2, work + order + n, non_rec, order);
+
+ for (i=0; i <= order; i++) {
+ out2[i] = out2[i] * ATTEN + buffer1[i];
+ out [i] = out2[i] + buffer2[i];
+ }
+
+ /* Multiply by the white noise correcting factor (WNCF). */
+ *out *= 257.0 / 256.0;
+}
diff --git a/libavcodec/ra288.c b/libavcodec/ra288.c
index 5b186a7a3d..9c69f49d81 100644
--- a/libavcodec/ra288.c
+++ b/libavcodec/ra288.c
@@ -37,6 +37,8 @@
#define MAX_BACKWARD_FILTER_ORDER 36
#define MAX_BACKWARD_FILTER_LEN 40
#define MAX_BACKWARD_FILTER_NONREC 35
+#define ATTEN 0.5625
+#include "g728_template.c"
#define RA288_BLOCK_SIZE 5
#define RA288_BLOCKS_PER_FRAME 32
@@ -87,13 +89,6 @@ static av_cold int ra288_decode_init(AVCodecContext *avctx)
return 0;
}
-static void convolve(float *tgt, const float *src, int len, int n)
-{
- for (; n >= 0; n--)
- tgt[n] = avpriv_scalarproduct_float_c(src, src - n, len);
-
-}
-
static void decode(RA288Context *ractx, float gain, int cb_coef)
{
int i;
@@ -131,45 +126,6 @@ static void decode(RA288Context *ractx, float gain, int
cb_coef)
ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36);
}
-/**
- * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
- *
- * @param order filter order
- * @param n input length
- * @param non_rec number of non-recursive samples
- * @param out filter output
- * @param hist pointer to the input history of the filter
- * @param out pointer to the non-recursive part of the output
- * @param out2 pointer to the recursive part of the output
- * @param window pointer to the windowing function table
- */
-static void do_hybrid_window(RA288Context *ractx,
- int order, int n, int non_rec, float *out,
- float *hist, float *out2, const float *window)
-{
- int i;
- float buffer1[MAX_BACKWARD_FILTER_ORDER + 1];
- float buffer2[MAX_BACKWARD_FILTER_ORDER + 1];
- LOCAL_ALIGNED(32, float, work, [FFALIGN(MAX_BACKWARD_FILTER_ORDER +
- MAX_BACKWARD_FILTER_LEN +
- MAX_BACKWARD_FILTER_NONREC, 16)]);
-
- av_assert2(order>=0);
-
- ractx->vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 16));
-
- convolve(buffer1, work + order , n , order);
- convolve(buffer2, work + order + n, non_rec, order);
-
- for (i=0; i <= order; i++) {
- out2[i] = out2[i] * 0.5625 + buffer1[i];
- out [i] = out2[i] + buffer2[i];
- }
-
- /* Multiply by the white noise correcting factor (WNCF). */
- *out *= 257.0 / 256.0;
-}
-
/**
* Backward synthesis filter, find the LPC coefficients from past speech data.
*/
@@ -180,7 +136,7 @@ static void backward_filter(RA288Context *ractx,
{
float temp[MAX_BACKWARD_FILTER_ORDER+1];
- do_hybrid_window(ractx, order, n, non_rec, temp, hist, rec, window);
+ do_hybrid_window(ractx->vector_fmul, order, n, non_rec, temp, hist, rec,
window);
if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))
ractx->vector_fmul(lpc, lpc, tab, FFALIGN(order, 16));
--
2.45.2
-- Peter
(A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B)
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