Mercurial > libavcodec.hg

changeset 5366:59a9c1e471fc libavcodec

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Yet more cosmetics
author vitor
date Wed, 18 Jul 2007 14:36:30 +0000
parents 8feb956bdbee
children 1a1f93a4e726
files alac.c
diffstat 1 files changed, 4 insertions(+), 9 deletions(-) [+]
line wrap: on
line diff
--- a/alac.c	Wed Jul 18 14:29:14 2007 +0000
+++ b/alac.c	Wed Jul 18 14:36:30 2007 +0000
@@ -321,9 +321,7 @@
             int32_t val;
 
             val = buffer_out[i] + error_buffer[i+1];
-
             val = SIGN_EXTENDED32(val, readsamplesize);
-
             buffer_out[i+1] = val;
         }
 
@@ -342,7 +340,6 @@
     }
 #endif
 
-
     /* general case */
     if (predictor_coef_num > 0) {
         for (i = predictor_coef_num + 1; i < output_size; i++) {
@@ -561,8 +558,8 @@
                                                prediction_quantitization[chan]);
             } else {
                 av_log(avctx, AV_LOG_ERROR, "FIXME: unhandled prediction type: %i\n", prediction_type[chan]);
-                /* i think the only other prediction type (or perhaps this is just a
-                 * boolean?) runs adaptive fir twice.. like:
+                /* I think the only other prediction type (or perhaps this is
+                 * just a boolean?) runs adaptive fir twice.. like:
                  * predictor_decompress_fir_adapt(predictor_error, tempout, ...)
                  * predictor_decompress_fir_adapt(predictor_error, outputsamples ...)
                  * little strange..
@@ -573,7 +570,7 @@
         /* not compressed, easy case */
         if (alac->setinfo_sample_size <= 16) {
             int i, chan;
-            for (chan = 0; chan < channels; chan++) {
+            for (chan = 0; chan < channels; chan++)
                 for (i = 0; i < outputsamples; i++) {
                     int32_t audiobits;
 
@@ -582,10 +579,9 @@
 
                     alac->outputsamples_buffer[chan][i] = audiobits;
                 }
-            }
         } else {
             int i, chan;
-            for (chan = 0; chan < channels; chan++) {
+            for (chan = 0; chan < channels; chan++)
                 for (i = 0; i < outputsamples; i++) {
                     int32_t audiobits;
 
@@ -598,7 +594,6 @@
 
                     alac->outputsamples_buffer[chan][i] = audiobits;
                 }
-            }
         }
         /* wasted_bytes = 0; */
         interlacing_shift = 0;