Fix issue357

Do what the spec says, insane or not:
"
Format 0 (uncompressed) and Format 3 (uncompressed little-endian) are similar. Both encode
uncompressed audio samples. For 8-bit samples, the two formats are identical. For 16-bit
samples, the two formats differ in byte ordering. In Format 0, 16-bit samples are encoded and
decoded according to the native byte ordering of the platform on which the encoder and Flash
Player, respectively, are running. In Format 3, 16-bit samples are always encoded in little-endian
order (least significant byte first), and are byte-swapped if necessary in Flash Player before
playback. Format 0 is clearly disadvantageous because it introduces a playback platform
dependency. For 16-bit samples, Format 3 is highly preferable to Format 0 for SWF version 4
or later.
"


git-svn-id: file:///var/local/repositories/ffmpeg/trunk@12184 9553f0bf-9b14-0410-a0b8-cfaf0461ba5b

libavformat/flvdec.c

@@ -42,7 +42,13 @@ static void flv_set_audio_codec(AVFormatContext *s, AVStream *astream, int flv_c
     switch(flv_codecid) {
         //no distinction between S16 and S8 PCM codec flags
         case FLV_CODECID_PCM:
-            acodec->codec_id = acodec->bits_per_sample == 8 ? CODEC_ID_PCM_S8 : CODEC_ID_PCM_S16BE; break;
+            acodec->codec_id = acodec->bits_per_sample == 8 ? CODEC_ID_PCM_S8 :
+#ifdef WORDS_BIGENDIAN
+                                CODEC_ID_PCM_S16BE;
+#else
+                                CODEC_ID_PCM_S16LE;
+#endif
+            break;
         case FLV_CODECID_PCM_LE:
             acodec->codec_id = acodec->bits_per_sample == 8 ? CODEC_ID_PCM_S8 : CODEC_ID_PCM_S16LE; break;
         case FLV_CODECID_ADPCM: acodec->codec_id = CODEC_ID_ADPCM_SWF;                              break;