Merge remote-tracking branch 'qatar/master'

* qatar/master: (27 commits)
  avconv: free packet in write_frame() when discarding due to frame number limit
  FATE: use +/- flag option syntax for vp8 emu-edge tests
  lavf: make av_interleave_packet_per_dts() private.
  lavf: deprecate av_read_packet().
  oggdec: output correct timestamps for Vorbis
  avconv: pass input stream timestamps to audio encoders
  lavc: shrink encoded audio packet size after encoding.
  xa: set correct bit rate
  xa: do not set bit_rate, block_align, or bits_per_coded_sample
  xa: fix end-of-file handling
  xa: fix timestamp calculation
  bink: fix typo in FFALIGN() argument
  bink: align plane width to 8 when calculating bundle sizes
  doc: pass -Idoc texi2html and texi2pod
  doc: texi2pod: add -I flag
  movenc: Add a min_frag_duration option
  rtsp: Set the default delay to 0.1 s for the RTSP/SDP/RTP demuxers
  libavformat: Set the default for the max_delay option to -1
  Generate manpages for AV{Format,Codec}Context AVOptions.
  doc/avconv: remove entries for AVOptions.
  ...

Conflicts:
	doc/Makefile
	doc/ffmpeg.texi
	doc/muxers.texi
	ffmpeg.c
	libavcodec/Makefile
	libavcodec/options.c
	libavcodec/vp8.c
	libavformat/options.c
	tests/fate/demux.mak
	tests/ref/fate/truemotion1-15
	tests/ref/fate/truemotion1-24

Merged-by: Michael Niedermayer <michaelni@gmx.at>

libavformat/rtpenc.c

@@ -126,7 +126,7 @@ static int rtp_write_header(AVFormatContext *s1)
     s->max_payload_size = s1->packet_size - 12;
 
     s->max_frames_per_packet = 0;
-    if (s1->max_delay) {
+    if (s1->max_delay > 0) {
         if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
             int frame_size = av_get_audio_frame_duration(st->codec, 0);
             if (!frame_size)