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path: root/media/libstagefright/rtsp/AAMRAssembler.cpp
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* Fix bad checks that causes crash when streaming H.263 content.Roger1 Jonsson2012-12-031-1/+0
| | | | | | | | | | | | | | | | | | Remove checks that causes crash for rtsp streamed h.263 content with certain values in the RTP payload header: Remove zero check for the five reserved bits in the payload header. According to RFC 4629 these bits MUST be ignored by receivers. Remove zero-check for the VRC (Video Redundancy Coding) bit, skip packet instead. Remove zero-check for the PLEN bits (extra picture header), skip packet instead. Remove zero-check for the PEBIT bits (extra picture header), skip packet instead. Remove corresponding zero check for the four resreved bits in the AMR payload header. According to RFC 4867 these bits MUST be ignored by receivers. Change-Id: I7fc21d69a19d23da24f9267623c338d415ef1387
* Add new APIs AMessage::(set|find)Buffer to make it safer to passAndreas Huber2012-02-221-1/+1
| | | | | | ABuffer objects through messages. Change-Id: I9f8b4e4c4767d0d70a0105e0c0813b754379b49d
* Support for parsing of "folded" RTSP header values and some tweaks to the ↵Andreas Huber2011-12-091-6/+10
| | | | | | | | AMR assembler contributed by Samsung (untested). Change-Id: I182561fe0a1a564126bdbb317e96aa52bf525726
* Rename (IF_)LOGV(_IF) to (IF_)ALOGV(_IF) DO NOT MERGESteve Block2011-10-261-5/+5
| | | | | | | See https://android-git.corp.google.com/g/#/c/143865 Bug: 5449033 Change-Id: I0122812ed6ff6f5b59fe4a43ab8bff0577adde0a
* Remove stagefright foundation's incompatible logging interface and update ↵Andreas Huber2010-09-211-7/+9
| | | | | | callsites. Change-Id: I45fba7d60530ea0f233ac3695a97306b6dc1795c
* Better support for rtsp (normal play-)time display. Better seek support, ↵Andreas Huber2010-08-271-5/+1
| | | | | | | timeout if no packets arrive for too long. Change-Id: Id491541a6ae501604cda815f8e961a3bfe26db7d related-to-bug: 2556656
* Support for Gtalk video, includes AMR/H.263 assembler and packetization ↵Andreas Huber2010-08-041-0/+232
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