PCM data in AVI is always little endian

Original commit message from CVS:
PCM data in AVI is always little endian

diff --git a/gst/avi/gstavidemux.c b/gst/avi/gstavidemux.c
index 894d157..f7ab3af 100644 (file)
--- a/gst/avi/gstavidemux.c
+++ b/gst/avi/gstavidemux.c
@@ -127,7 +127,7 @@ GST_PAD_TEMPLATE_FACTORY (src_audio_templ,
     "audio/raw",
       "format",           GST_PROPS_STRING ("int"),
       "law",              GST_PROPS_INT (0),
-      "endianness",       GST_PROPS_INT (G_BYTE_ORDER),
+      "endianness",       GST_PROPS_INT (G_LITTLE_ENDIAN),
       "signed",           GST_PROPS_LIST (
                            GST_PROPS_BOOLEAN (TRUE),
                            GST_PROPS_BOOLEAN (FALSE)
@@ -786,7 +786,7 @@ gst_avi_demux_strf_auds (GstAviDemux *avi_demux)
                               gst_props_new (
                                 "format",     GST_PROPS_STRING ("int"),
                                 "law",        GST_PROPS_INT (0),
-                                "endianness", GST_PROPS_INT (G_BYTE_ORDER),
+                                "endianness", GST_PROPS_INT (G_LITTLE_ENDIAN),
                                 "signed",     GST_PROPS_BOOLEAN ((GUINT16_FROM_LE (strf->size) != 8)),
                                 "width",      GST_PROPS_INT ((GUINT16_FROM_LE (strf->blockalign)*8) /
                                                               GUINT16_FROM_LE (strf->channels)),

diff --git a/gst/avi/gstavimux.c b/gst/avi/gstavimux.c
index 750a512..ef8d4a6 100644 (file)
--- a/gst/avi/gstavimux.c
+++ b/gst/avi/gstavimux.c
@@ -419,19 +419,27 @@ gst_avimux_sinkconnect (GstPad *pad, GstCaps *vscaps)
     }
     else if (!strcmp (mimetype, "audio/x-mp3"))
     {
-      gint layer;
+      gint layer = 3;
 
-      gst_caps_get_int(caps, "layer", &layer);
+      if (gst_caps_has_property(caps, "layer"))
+        gst_caps_get_int(caps, "layer", &layer);
+      else
+        GST_DEBUG(GST_CAT_PLUGIN_INFO,
+                  "No layer specified, assuming layer 3");
 
       /* we don't need to do anything here, compressed mp3 contains it all */
       avimux->auds.format      = (layer == 3?
                                    GST_RIFF_WAVE_FORMAT_MPEGL3 : 
                                   GST_RIFF_WAVE_FORMAT_MPEGL12);
+      avimux->auds_hdr.scale = avimux->auds_hdr.samplesize = 1;
+      avimux->auds_hdr.rate = avimux->auds.av_bps = 0;
       goto done;
     }
     else if (!strcmp (mimetype, "application/x-ogg"))
     {
       avimux->auds.format = GST_RIFF_WAVE_FORMAT_VORBIS1;
+      avimux->auds_hdr.scale = avimux->auds_hdr.samplesize = 1;
+      avimux->auds_hdr.rate = avimux->auds.av_bps = 0;
       goto done;
     }
   }
@@ -969,8 +977,14 @@ gst_avimux_stop_file (GstAviMux *avimux)
   /* set rate and everything having to do with that */
   avimux->avi_hdr.us_frame = avimux->vids_hdr.scale = 1000000/avimux->framerate;
   avimux->avi_hdr.max_bps = 0;
-  if (avimux->audio_pad_connected)
+  if (avimux->audio_pad_connected) {
+    /* calculate bps if needed */
+    if (!avimux->auds.av_bps) {
+      g_warning("Bps calculation needed!");
+      avimux->auds_hdr.rate = avimux->auds.av_bps * avimux->auds_hdr.scale;
+    }
     avimux->avi_hdr.max_bps += avimux->auds.av_bps;
+  }
   if (avimux->video_pad_connected)
     avimux->avi_hdr.max_bps += ((avimux->vids.bit_cnt+7)/8) *
                                avimux->framerate *