[SCM] libav/experimental: increase audio buffer to 1second of 48khz float audio

[siretart at users.alioth.debian.org]

The following commit has been merged in the experimental branch:
commit 8170e5fb4988ddbb4aec8f2e50974dcff0f1158c
Author: Alex Beregszaszi <alex at rtfs.hu>
Date:   Mon Feb 13 12:00:27 2006 +0000

    increase audio buffer to 1second of 48khz float audio
    
    Originally committed as revision 5013 to svn://svn.ffmpeg.org/ffmpeg/trunk

diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index d0d5e64..10167d7 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -21,8 +21,8 @@ extern "C" {
 #define AV_STRINGIFY(s)         AV_TOSTRING(s)
 #define AV_TOSTRING(s) #s
 
-#define LIBAVCODEC_VERSION_INT  ((51<<16)+(4<<8)+0)
-#define LIBAVCODEC_VERSION      51.4.0
+#define LIBAVCODEC_VERSION_INT  ((51<<16)+(5<<8)+0)
+#define LIBAVCODEC_VERSION      51.5.0
 #define LIBAVCODEC_BUILD        LIBAVCODEC_VERSION_INT
 
 #define LIBAVCODEC_IDENT        "Lavc" AV_STRINGIFY(LIBAVCODEC_VERSION)
@@ -270,7 +270,7 @@ enum SampleFormat {
 };
 
 /* in bytes */
-#define AVCODEC_MAX_AUDIO_FRAME_SIZE 131072
+#define AVCODEC_MAX_AUDIO_FRAME_SIZE 192000 // 1 second of 48khz 32bit audio
 
 /**
  * Required number of additionally allocated bytes at the end of the input bitstream for decoding.

-- 
Libav/FFmpeg packaging