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path: root/media/libstagefright/rtsp/ARTPConnection.cpp
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* libstagefright: fix 64-bit warningsColin Cross2014-04-141-1/+1
| | | | | | | | | | | %lld -> %" PRId64 " for int64_t %d -> %zu for size_t Also fixes some casts from void* to integer types, and some comparisons between signed and unsigned. (cherry picked from commit b4a7a2df4c28c3f32b5d877b54831d2cc5d78f81) Change-Id: I76ba94d0b67776fd7abdc83b43d47c61d6c32f4c
* Fix overflow of rand in ARTPConnectionYajun Zeng2013-04-241-1/+2
| | | | | | | without this fix, (rand()*1000)/RAND_MAX is mainly 0. Change-Id: I48ae940a7b6974b197d81732774c9dcea107bcf1 Signed-off-by: Yajun Zeng <beanz@marvell.com>
* Fixed member access into incomplete type build errorTareq A. Siraj2012-08-221-0/+1
| | | | | | | | | Included the ARTPAssembler.h file to fix the 'member access into incomplete type "android::ARTPAssembler"' error reported by clang. Change-Id: I10cb1e38bf360858bb7ebdeae82ba1e64431f87d Author: Tareq A. Siraj <tareq.a.siraj@intel.com> Reviewed-by: Edwin Vane<edwin.vane@intel.com>
* Add new APIs AMessage::(set|find)Buffer to make it safer to passAndreas Huber2012-02-221-5/+3
| | | | | | ABuffer objects through messages. Change-Id: I9f8b4e4c4767d0d70a0105e0c0813b754379b49d
* Rename (IF_)LOGW(_IF) to (IF_)ALOGW(_IF) DO NOT MERGESteve Block2012-01-061-3/+3
| | | | | | | See https://android-git.corp.google.com/g/157065 Bug: 5449033 Change-Id: I00a4b904f9449e6f93b7fd35eac28640d7929e69
* Rename (IF_)LOGI(_IF) to (IF_)ALOGI(_IF) DO NOT MERGESteve Block2012-01-041-2/+2
| | | | | | | See https://android-git.corp.google.com/g/156801 Bug: 5449033 Change-Id: Ib08fe86d23db91ee153e9f91a99a35c42b9208ea
* Instead of asserting, remove active streams if their sockets return failureAndreas Huber2011-11-091-15/+47
| | | | | Change-Id: Ic5cc786f718cf921876b181927cf1b03e8373ff1 related-to-bug: 5593654
* Rename (IF_)LOGV(_IF) to (IF_)ALOGV(_IF) DO NOT MERGESteve Block2011-10-261-2/+2
| | | | | | | See https://android-git.corp.google.com/g/#/c/143865 Bug: 5449033 Change-Id: I0122812ed6ff6f5b59fe4a43ab8bff0577adde0a
* Work around several issues with non-compliant RTSP servers.Andreas Huber2011-02-151-1/+3
| | | | | | | | | | | In this particular case these RTSP servers were implemented as local services, retransmitting live streams via a local RTSP server instance. They picked wrong rtp/rtcp port pairs (odd rtp port), blank lines in the session description, wrong case of the format description, relative base URLs... Change-Id: I63fa90ca2398f19e8b52c147248bd2c5c2372426 related-to-bug: 3452103
* Change timestamp handling in RTSP, remove unused, experimental, gtalk supportAndreas Huber2011-02-101-37/+1
| | | | | | | | related-to-bug: 3216447 NTP timestamp handling is now done at a higher layer than before. Change-Id: I9fb23f1335110ec59e534f9aa0fe6f6a6406dd52
* Some webcams output rtp streams but never send any rtcp data in violation ofAndreas Huber2010-10-131-1/+37
| | | | | | | the specs. Attempt to use fake timestamps to be able to play these... Change-Id: Ia7a926616fb764e972955df4acdb59d85cdd93df related-to-bug: 3087310
* Remove stagefright foundation's incompatible logging interface and update ↵Andreas Huber2010-09-211-9/+12
| | | | | | callsites. Change-Id: I45fba7d60530ea0f233ac3695a97306b6dc1795c
* Better detection of connection problems - timeout if no rtcp packets arrive ↵Andreas Huber2010-08-311-2/+6
| | | | | | | within a certain time, not a final frame (which may take longer) Change-Id: I3c1ae79bb9342770e959ebdcdc6b748549b76330 related-to-bug: 2556656
* Instead of closing the connection altogether if no UDP packets arrive after ↵Andreas Huber2010-08-301-0/+8
| | | | | | | a certain time, try changing transports (to interleaved TCP). Also properly close the sockets on disconnection. Change-Id: Ie8d6a3865a0477e28d4b76bb9038e468451287b1 related-to-bug: 2556656
* Support for RTP packets arriving interleaved with RTSP responses.Andreas Huber2010-08-261-2/+66
| | | | Change-Id: Ib32fba257da32a199134cf8943117cf3eaa07a25
* Support for MP4V-ES packetization format according to RFC3016.Andreas Huber2010-08-191-0/+2
| | | | Change-Id: I5e182936c52f9eb80cdcf6132ead03705ee32d61
* We're now going to ignore timestamps completely in gtalk video conferencing, ↵Andreas Huber2010-08-101-12/+12
| | | | | | playing video as soon as it comes in. We also make up fake timestamps in the rtp code, ignoring rtcp SR information to enable early startup. Change-Id: Idc3df74b42000f7a6aa3eae090718dc9d9c4186f
* Better support for fake timestamps in RTP, H.263 video now also requests FIR.Andreas Huber2010-08-051-6/+8
| | | | Change-Id: I2385461887197fe4062d329086e0204f6d6620fc
* Support for Gtalk video, includes AMR/H.263 assembler and packetization ↵Andreas Huber2010-08-041-61/+121
| | | | | | support, extensions to MediaRecorder to stream via RTP over a pair of UDP sockets as well as various fixes to the RTP implementation. Change-Id: I95b8dd487061add9bade15749e563b01cd99d9a6
* Initial checkin of preliminary rtsp support for stagefright.Andreas Huber2010-06-071-0/+499
Change-Id: I0722aa888098c0c1361c97a4c1b123d910afc207
generated by cgit v1.1 at 2024-05-23 17:23:45 +0000