Gitiles
Code Review
Sign In
gerrit-public.fairphone.software
/
platform
/
frameworks
/
base
/
ae7752798a98fc81ff5e6ae69dde2137692106be
/
media
/
libstagefright
/
rtsp
/
APacketSource.cpp
b672373
Support for PCMA and PCMU raw audio data in RTP/RTSP.
by Andreas Huber
· 13 years ago
0407269
Work around several issues with non-compliant RTSP servers.
by Andreas Huber
· 13 years ago
77034e6
Implement parsing of vbv buffering info in RTSP.
by Andreas Huber
· 14 years ago
38285db
Refactor some more h.264 utility code out into avc_utils. Work around a hardware decoder issue by making sure the first access unit submitted to a decoder at startup or after seek is an IDR.
by Andreas Huber
· 14 years ago
6e3fa44
Remove stagefright foundation's incompatible logging interface and update callsites.
by Andreas Huber
· 14 years ago
f3d2bdf
Various fixes to improve resilience of the rtsp stack against spurious errors instead of asserting.
by Andreas Huber
· 14 years ago
4d8f66b
Better support for buffered streaming of rtsp content, if buffer drops below a certain threshold we will temporarily pause playback until we have sufficient data.
by Andreas Huber
· 14 years ago
e536f80
Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr.
by Andreas Huber
· 14 years ago
eeb97d9
Better support for rtsp (normal play-)time display. Better seek support, timeout if no packets arrive for too long.
by Andreas Huber
· 14 years ago
e0dd7d3
A first shot at proper support for seeking of rtsp streams.
by Andreas Huber
· 14 years ago
a979ad6
Support for MP4V-ES packetization format according to RFC3016.
by Andreas Huber
· 14 years ago
eef3c33
In the absence of width/height information in the sdp, extract the dimensions from the avc codec specific data.
by Andreas Huber
· 14 years ago
af063a6
Better support for rtsp streamed through VLC. Temporarily make the socket blocking to read all of the session description.
by Andreas Huber
· 14 years ago
3f55576
APacketSource is too verbose.
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
426b650
Specification of codec specific data as part of the session description is now optional.
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
7a747b8
Initial checkin of preliminary rtsp support for stagefright.
by Andreas Huber
· 14 years ago