From upstream: fix AAC decoding in GStreamer 0.10. Fixes, e.g., internet

radio playing in Clementine.

See http://cgit.freedesktop.org/gstreamer/gst-ffmpeg/commit/?h=1.0&id=a3b0ae22d76522d0a79f5d946872c0260dd1e3b2
and https://bugzilla.gnome.org/show_bug.cgi?id=679639 for details.

ok ajacoutot@

multimedia/gstreamer-0.10/plugins-ffmpeg/Makefile

@@ -1,11 +1,11 @@
-# $OpenBSD: Makefile,v 1.32 2013/03/25 12:31:58 sthen Exp $
+# $OpenBSD: Makefile,v 1.33 2013/03/25 16:09:41 zhuk Exp $
 
 COMMENT=		ffmpeg element for GStreamer
 
 V=			0.10.13
 DISTNAME=		gst-ffmpeg-${V}
 PKGNAME=		gstreamer-ffmpeg-${V}
-REVISION=		5
+REVISION=		6
 
 # sync with graphics/ffmpeg
 PERMIT_PACKAGE_CDROM=	patents

multimedia/gstreamer-0.10/plugins-ffmpeg/patches/patch-ext_ffmpeg_gstffmpegdec_c

@@ -1,4 +1,4 @@
-$OpenBSD: patch-ext_ffmpeg_gstffmpegdec_c,v 1.4 2012/03/03 10:33:42 ajacoutot Exp $
+$OpenBSD: patch-ext_ffmpeg_gstffmpegdec_c,v 1.5 2013/03/25 16:09:41 zhuk Exp $
 
 From upstream:
 - Report latency if B-frames are present.
@@ -6,6 +6,12 @@ From upstream:
 - Only set get_buffer() function for video.
 - Only slice-threading.
 
+Also:
+From a3b0ae22d76522d0a79f5d946872c0260dd1e3b2 Mon Sep 17 00:00:00 2001
+From: Wim Taymans <wim.taymans@collabora.co.uk>
+Date: Tue, 10 Jul 2012 14:10:14 +0000
+Subject: avdec: ignore AAC errors instead of erroring out
+
 --- ext/ffmpeg/gstffmpegdec.c.orig	Wed Nov  2 09:04:05 2011
 +++ ext/ffmpeg/gstffmpegdec.c	Fri Mar  2 17:23:48 2012
 @@ -477,36 +477,44 @@ static gboolean
@@ -120,7 +126,23 @@ From upstream:
          GST_DEBUG_OBJECT (ffmpegdec,
              "Buffer interlacing does not match pad, updating");
          buffer = gst_buffer_make_metadata_writable (buffer);
-@@ -3043,14 +3058,6 @@ gst_ffmpegdec_register (GstPlugin * plugin)
+@@ -2205,15 +2205,6 @@ gst_ffmpegdec_audio_frame (GstFFMpegDec * ffmpegdec,
+     *outbuf = NULL;
+   }
+ 
+-  /* If we don't error out after the first failed read with the AAC decoder,
+-   * we must *not* carry on pushing data, else we'll cause segfaults... */
+-  if (len == -1 && (in_plugin->id == CODEC_ID_AAC
+-          || in_plugin->id == CODEC_ID_AAC_LATM)) {
+-    GST_ELEMENT_ERROR (ffmpegdec, STREAM, DECODE, (NULL),
+-        ("Decoding of AAC stream by FFMPEG failed."));
+-    *ret = GST_FLOW_ERROR;
+-  }
+-
+ beach:
+   GST_DEBUG_OBJECT (ffmpegdec, "return flow %d, out %p, len %d",
+       *ret, *outbuf, len);
+@@ -3043,14 +3049,6 @@ gst_ffmpegdec_register (GstPlugin * plugin)
          rank = GST_RANK_SECONDARY;
          break;
        case CODEC_ID_MP3: