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gerrit-public.fairphone.software
/
platform
/
frameworks
/
base
/
0df0ed5814a83c4cf0dd3d6d6e177f5ed9003863
/
media
/
libstagefright
/
rtsp
/
ARTPSession.cpp
6215d3f
Rename (IF_)LOGI(_IF) to (IF_)ALOGI(_IF) DO NOT MERGE
by Steve Block
· 13 years ago
b2934b1
Change timestamp handling in RTSP, remove unused, experimental, gtalk support
by Andreas Huber
· 13 years ago
6e3fa44
Remove stagefright foundation's incompatible logging interface and update callsites.
by Andreas Huber
· 14 years ago
27b9c8e
Keep gtalk video chat specific code consistent with rtsp changes.
by Andreas Huber
· 14 years ago
0ddf8c0
Recent changes to the rtsp code require every buffer fed to the packet source to have a timestamp, we're ignoring timestamps for gtalk videochat but we still have to have a placeholder.
by Andreas Huber
· 14 years ago
0416da7
Support for RTP packets arriving interleaved with RTSP responses.
by Andreas Huber
· 14 years ago
f88f844
We're now going to ignore timestamps completely in gtalk video conferencing, 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.
by Andreas Huber
· 14 years ago
57648e4
Support for Gtalk video, includes AMR/H.263 assembler and packetization support, extensions to MediaRecorder to stream via RTP over a pair of UDP sockets as well as various fixes to the RTP implementation.
by Andreas Huber
· 14 years ago