[FFmpeg-devel] [PATCHv3 1/8] avcodec/g728_template: do_hybrid_window() template

[Peter Ross <pross@xvid.org>]

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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..b60122ee37
--- /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] = ff_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 aa499506b7..a1ee3f7eba 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] = ff_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));
-- 
2.47.2

-- Peter
(A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B)

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