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30 Commits

Author SHA1 Message Date
bertrand 52dd75e903 Allows the LIVE555 library to forces the client's port to be used 
when reading from an RTP/RTSP source.

Patch from Benjamin Zores <ben@geexbox.org> and Patrick Labatut <plabatut@gmail.com>


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@17651 b3059339-0415-0410-9bf9-f77b7e298cf2
 2006-02-19 13:27:27 +00:00
rsf b383ffa4b7 "LIVE.COM Streaming Media" is now called "LIVE555 Streaming Media". 
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@16573 b3059339-0415-0410-9bf9-f77b7e298cf2
 2005-09-23 22:35:04 +00:00
rtognimp cd68e1618b Demuxer modularization 
Demuxer selection by name with -demuxer command (bakward compatible)


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@16176 b3059339-0415-0410-9bf9-f77b7e298cf2
 2005-08-05 19:57:47 +00:00
nicodvb 83b3c822be ported all network streams to the new API 
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@15586 b3059339-0415-0410-9bf9-f77b7e298cf2
 2005-05-29 12:54:00 +00:00
rsf a26dcc7b48 Updated to conform to a small change in the LIVE.COM API. 
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@12867 b3059339-0415-0410-9bf9-f77b7e298cf2
 2004-07-20 02:12:08 +00:00
diego 19cf857451 MinGW compilation fix from a patch by Joey Parrish, approved by Sascha 
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@12532 b3059339-0415-0410-9bf9-f77b7e298cf2
 2004-06-03 13:30:55 +00:00
rsf 4d0be09a5d Added "audio_id", "video_id", "dvdsub_id" to the call to "demux_open()". 
(Thanks to Nico Sabbi for suggesting this.)


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@12522 b3059339-0415-0410-9bf9-f77b7e298cf2
 2004-06-02 06:48:25 +00:00
rsf 7611ee601e Fixed a bug that was accidentally introduced by the addition of MPEG Transport 
Stream support.  We now handle errors such as 'stream not found' correctly once
again.


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@12044 b3059339-0415-0410-9bf9-f77b7e298cf2
 2004-03-19 10:15:41 +00:00
rsf 162310efe9 We now allow for the possibility of the RTCP audio/video synchronization being 
incorrect.  (I encounted a stream for which this was the case.)  Now, if
audio and video are out-of-sync by >60 seconds, we assume that the RTCP
sync is incorrect, and we don't discard any packets.


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@12008 b3059339-0415-0410-9bf9-f77b7e298cf2
 2004-03-02 08:52:59 +00:00
rsf 4594011e67 Added support for multiplexed audio+video RTP streams. 
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@11985 b3059339-0415-0410-9bf9-f77b7e298cf2
 2004-02-22 06:20:08 +00:00
rsf 3b8f28041c Changed to conform to recent changes to the "LIVE.COM Streaming Media" code. 
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@11756 b3059339-0415-0410-9bf9-f77b7e298cf2
 2004-01-06 09:30:27 +00:00
rsf d7f68d1212 Changed the criteria for when to drop RTP packets whose timestamp is too far 
behind that of the other (audio or video) stream.  Now, this is done only
if both streams have been synchronized using RTCP.


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@10938 b3059339-0415-0410-9bf9-f77b7e298cf2
 2003-09-24 08:41:57 +00:00
rsf 4867e17bfa Added support for checking whether a RTP demuxer contains combined audio_video data. 
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@10478 b3059339-0415-0410-9bf9-f77b7e298cf2
 2003-07-27 10:15:10 +00:00
rsf a3b6526ac6 Added SIP (IP telephony) client support. (This was already supported in the 
LIVE.COM libraries, so updating the MPlayer code to support it required
only relatively minor changes.)


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@10055 b3059339-0415-0410-9bf9-f77b7e298cf2
 2003-05-03 06:13:11 +00:00
rsf a8ae67f639 Fixed a bug that could sometimes cause the first video packet in a RTP stream 
to be rejected.


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@9912 b3059339-0415-0410-9bf9-f77b7e298cf2
 2003-04-12 09:30:19 +00:00
rsf 208d38b876 Access-controlled RTSP sessions can now be played, if the user uses the 
"-user" and "-passwd" options.


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@9905 b3059339-0415-0410-9bf9-f77b7e298cf2
 2003-04-11 02:35:01 +00:00
rsf bc5277b1bd Added some optional debugging printfs (disabled by default) 
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@9787 b3059339-0415-0410-9bf9-f77b7e298cf2
 2003-04-02 01:38:07 +00:00
rsf 555b3f61fe Improved RTP packet buffering, by relying on the underlying OS's UDP 
socket buffering.  Improve A/V sync by dropping packets when one stream
gets too far behind the other.  Now tries to figure out the video frame
rate automatically (if "-fps" is not used).  Added support for MPEG-4
Elementary Stream video and MPEG-4 Generic audio RTP streams.


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@9566 b3059339-0415-0410-9bf9-f77b7e298cf2
 2003-03-11 19:08:31 +00:00
bertrand 12322d2517 Repairing breakage to RTP streaming. Patch by Ross Finlayson <finlayson@live.com> 
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@9458 b3059339-0415-0410-9bf9-f77b7e298cf2
 2003-02-18 22:33:44 +00:00
arpi e19879533d Motion-JPEG RTP streams can now be played. Some MPEG-4 ES video RTP 
streams can also be played.
patch by Ross Finlayson <finlayson@live.com>


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@9371 b3059339-0415-0410-9bf9-f77b7e298cf2
 2003-02-09 17:06:38 +00:00
bertrand 06d22fab96 Restruct by Ross Finlayson <finlayson@live.com> 
The code now supports 'QuickTime generic' RTP streams (the "X-QT" MIME type),
which - thanks to the QuickTime codecs - makes it possible to play more QuickTime
RTP streams.


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@9251 b3059339-0415-0410-9bf9-f77b7e298cf2
 2003-02-03 10:27:50 +00:00
bertrand 1991d31057 Support for MPEG-4 (AAC) audio RTSP/RTP 
Patch by Ross Finlayson <finlayson@live.com>


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@8988 b3059339-0415-0410-9bf9-f77b7e298cf2
 2003-01-18 14:22:30 +00:00
bertrand 504e9aa82a From live.com 2002.11.30: 
Renamed "TaskScheduler::blockMyself()" to "doEventLoop()", to better
describe what this member function actually does.

Patch from Andreas Hess <jaska@gmx.net>


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@8384 b3059339-0415-0410-9bf9-f77b7e298cf2
 2002-12-06 09:41:13 +00:00
arpi 1413b42ad5 use standard gsm fourcc 'agsm' instead of msgsm id 0x31 
patch by Ross


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@7751 b3059339-0415-0410-9bf9-f77b7e298cf2
 2002-10-16 15:15:43 +00:00
arpi d28c74a027 -fps autodetection 
patch by Ross Finlayson <finlayson@live.com>


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@7731 b3059339-0415-0410-9bf9-f77b7e298cf2
 2002-10-13 21:57:54 +00:00
bertrand f767a62a42 Added support for RTSP stream over TCP. 
Patch from Ross Finlayson <finlayson@live.com>


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@7665 b3059339-0415-0410-9bf9-f77b7e298cf2
 2002-10-08 05:46:23 +00:00
arpi db99f581cf passthrough timestamps to demuxer 
patch by Ross Finlayson <finlayson@live.com>


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@7535 b3059339-0415-0410-9bf9-f77b7e298cf2
 2002-09-28 18:51:44 +00:00
arpi d4197da915 - Tell the RTSP client code to use the string "mplayer" in RTSP 
"User-Agent:" fields.  NOTE: This requires an up-to-date version of the
LIVE.COM Streaming Media libraries.
- Fix a bug that could cause mplayer to crash on exit if a "rtsp://" URL was bad.
patch by Ross Finlayson <finlayson@live.com>


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@7144 b3059339-0415-0410-9bf9-f77b7e298cf2
 2002-08-29 20:19:33 +00:00
arpi d2673b6e3b - Create and set up a "BITMAPINFOHEADER" and "WAVEFORMATEX" structure for 
video and audio (respectively) RTP streams.  (This allows RTP streams that
  use non MPEG codecs to work.)
patch by Ross Finlayson <finlayson@live.com>


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@7009 b3059339-0415-0410-9bf9-f77b7e298cf2
 2002-08-14 21:56:31 +00:00
arpi fa788640e2 applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com> 
see <http://www.live.com/mplayer/> for details.


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@6911 b3059339-0415-0410-9bf9-f77b7e298cf2
 2002-08-05 00:39:07 +00:00