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path: root/media/libstagefright/rtsp/ARTPSession.cpp
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* Add new APIs AMessage::(set|find)Buffer to make it safer to passAndreas Huber2012-02-221-4/+2
| | | | | | ABuffer objects through messages. Change-Id: I9f8b4e4c4767d0d70a0105e0c0813b754379b49d
* Rename (IF_)LOGE(_IF) to (IF_)ALOGE(_IF) DO NOT MERGESteve Block2012-01-081-4/+4
| | | | | | | See https://android-git.corp.google.com/g/#/c/157220 Bug: 5449033 Change-Id: Ic9c19d30693bd56755f55906127cd6bd7126096c
* Rename (IF_)LOGI(_IF) to (IF_)ALOGI(_IF) DO NOT MERGESteve Block2012-01-041-3/+3
| | | | | | | See https://android-git.corp.google.com/g/156801 Bug: 5449033 Change-Id: Ib08fe86d23db91ee153e9f91a99a35c42b9208ea
* Change timestamp handling in RTSP, remove unused, experimental, gtalk supportAndreas Huber2011-02-101-3/+1
| | | | | | | | related-to-bug: 3216447 NTP timestamp handling is now done at a higher layer than before. Change-Id: I9fb23f1335110ec59e534f9aa0fe6f6a6406dd52
* Remove stagefright foundation's incompatible logging interface and update ↵Andreas Huber2010-09-211-11/+13
| | | | | | callsites. Change-Id: I45fba7d60530ea0f233ac3695a97306b6dc1795c
* Keep gtalk video chat specific code consistent with rtsp changes.Andreas Huber2010-09-011-0/+8
| | | | Change-Id: I5f3f46c2150e16b26674432e427f79c04a69cd8e
* Recent changes to the rtsp code require every buffer fed to the packet ↵Andreas Huber2010-08-301-0/+1
| | | | | | source to have a timestamp, we're ignoring timestamps for gtalk videochat but we still have to have a placeholder. Change-Id: Ib8615ce5a89a9a846ee2f9f96cdfb23462f72c7a
* Support for RTP packets arriving interleaved with RTSP responses.Andreas Huber2010-08-261-1/+2
| | | | Change-Id: Ib32fba257da32a199134cf8943117cf3eaa07a25
* We're now going to ignore timestamps completely in gtalk video conferencing, ↵Andreas Huber2010-08-101-1/+4
| | | | | | 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
* Support for Gtalk video, includes AMR/H.263 assembler and packetization ↵Andreas Huber2010-08-041-0/+231
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