avformat: remove avio_flush() calls from the end of write_packet functions

[ffmpeg] / libavcodec / g729postfilter.c

diff --git a/libavcodec/g729postfilter.c b/libavcodec/g729postfilter.c
index d9076ec7357f0eeee8ed42d069f1128559b0b95c..fc9a8d54cc56ca395607191daae2121c5c121a4d 100644 (file)
--- a/libavcodec/g729postfilter.c
+++ b/libavcodec/g729postfilter.c
@@ -156,7 +156,7 @@ static int16_t long_term_filter(AudioDSPContext *adsp, int pitch_delay_int,
             sig_scaled[i] = residual[i] >> shift;
     else
         for (i = 0; i < subframe_size + RES_PREV_DATA_SIZE; i++)
-            sig_scaled[i] = residual[i] << -shift;
+            sig_scaled[i] = (unsigned)residual[i] << -shift;
 
     /* Start of best delay searching code */
     gain_num = 0;
@@ -201,8 +201,8 @@ static int16_t long_term_filter(AudioDSPContext *adsp, int pitch_delay_int,
         }
         if (corr_int_num) {
             /* Compute denominator of pseudo-normalized correlation R'(0). */
-            corr_int_den = adsp->scalarproduct_int16(sig_scaled - best_delay_int + RES_PREV_DATA_SIZE,
-                                                    sig_scaled - best_delay_int + RES_PREV_DATA_SIZE,
+            corr_int_den = adsp->scalarproduct_int16(sig_scaled + RES_PREV_DATA_SIZE - best_delay_int,
+                                                     sig_scaled + RES_PREV_DATA_SIZE - best_delay_int,
                                                     subframe_size);
 
             /* Compute signals with non-integer delay k (with 1/8 precision),
@@ -346,7 +346,7 @@ static int16_t long_term_filter(AudioDSPContext *adsp, int pitch_delay_int,
         L_temp1 = gain_long_num * gain_long_num;
         L_temp1 = MULL(L_temp1, gain_den, FRAC_BITS);
 
-        tmp = ((sh_gain_long_num - sh_gain_num) << 1) - (sh_gain_long_den - sh_gain_den);
+        tmp = ((sh_gain_long_num - sh_gain_num) * 2) - (sh_gain_long_den - sh_gain_den);
         if (tmp > 0)
             L_temp0 >>= tmp;
         else
@@ -367,7 +367,7 @@ static int16_t long_term_filter(AudioDSPContext *adsp, int pitch_delay_int,
         /* Rescale selected signal to original value. */
         if (shift > 0)
             for (i = 0; i < subframe_size; i++)
-                selected_signal[i] <<= shift;
+                selected_signal[i] *= 1 << shift;
         else
             for (i = 0; i < subframe_size; i++)
                 selected_signal[i] >>= -shift;
@@ -464,7 +464,7 @@ static int16_t get_tilt_comp(AudioDSPContext *adsp, int16_t *lp_gn,
             speech[i] = (speech[i] * temp + 0x4000) >> 15;
     }
 
-    return -(rh1 << 15) / rh0;
+    return -(rh1 * (1 << 15)) / rh0;
 }
 
 /**
@@ -500,14 +500,14 @@ static int16_t apply_tilt_comp(int16_t* out, int16_t* res_pst, int refl_coeff,
     tmp = res_pst[subframe_size - 1];
 
     for (i = subframe_size - 1; i >= 1; i--) {
-        tmp2 = (res_pst[i] << 15) + ((gt * res_pst[i-1]) << 1);
-        tmp2 = (tmp2 + 0x4000) >> 15;
+        tmp2 = (gt * res_pst[i-1]) * 2 + 0x4000;
+        tmp2 = res_pst[i] + (tmp2 >> 15);
 
         tmp2 = (tmp2 * ga * 2 + fact) >> sh_fact;
         out[i] = tmp2;
     }
-    tmp2 = (res_pst[0] << 15) + ((gt * ht_prev_data) << 1);
-    tmp2 = (tmp2 + 0x4000) >> 15;
+    tmp2 = (gt * ht_prev_data) * 2 + 0x4000;
+    tmp2 = res_pst[0] + (tmp2 >> 15);
     tmp2 = (tmp2 * ga * 2 + fact) >> sh_fact;
     out[0] = tmp2;