and at the end ffmpeg is definitely able to handle the aac, as ffplay
plays the aac stream also
so you are right that it is most likely the gstreamer ffmpeg plugin not
ffmpeg itself which has problems
I stop here now
ffplay http://mp3channels.webradio.antenne.de/antenne.aac
ffplay version 2.1.1 Copyright (c) 2003-2013 the FFmpeg developers
built on Nov 27 2013 18:31:36 with gcc 4.8 (SUSE Linux)
configuration: --shlibdir=/usr/lib64 --prefix=/usr
--mandir=/usr/share/man --libdir=/usr/lib64 --enable-shared
--disable-static --enable-debug --disable-stripping
--extra-cflags='-fmessage-length=0 -grecord-gcc-switches
-fstack-protector -O2 -Wall -D_FORTIFY_SOURCE=2 -funwind-tables
-fasynchronous-unwind-tables -g -fPIC -I/usr/include/gsm' --enable-gpl
--enable-x11grab --enable-version3 --enable-pthreads --enable-avfilter
--enable-libpulse --enable-libvpx --enable-libopus --enable-libass
--enable-libmp3lame --enable-libvorbis --enable-libtheora
--enable-libspeex --enable-libxvid --enable-libx264
--enable-libschroedinger --enable-libgsm --enable-libopencore-amrnb
--enable-libopencore-amrwb --enable-postproc --enable-libdc1394
--enable-librtmp --enable-libfreetype --enable-avresample
--enable-libtwolame --enable-libvo-aacenc --enable-gnutls
libavutil 52. 48.101 / 52. 48.101
libavcodec 55. 39.101 / 55. 39.101
libavformat 55. 19.104 / 55. 19.104
libavdevice 55. 5.100 / 55. 5.100
libavfilter 3. 90.100 / 3. 90.100
libavresample 1. 1. 0 / 1. 1. 0
libswscale 2. 5.101 / 2. 5.101
libswresample 0. 17.104 / 0. 17.104
libpostproc 52. 3.100 / 52. 3.100
Input #0, aac, from 'http://mp3channels.webradio.antenne.de/antenne.aac':
Duration: N/A, bitrate: 70 kb/s
Stream #0:0: Audio: aac, 44100 Hz, stereo, fltp, 70 kb/s
6.66 M-A: 0.000 fd= 0 aq= 2KB vq= 0KB sq= 0B f=0/0