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gerrit-public.fairphone.software / platform / frameworks / base / ae7752798a98fc81ff5e6ae69dde2137692106be / media / libstagefright / rtsp / APacketSource.h
  1. 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
  2. 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
  3. 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
  4. e0dd7d3 A first shot at proper support for seeking of rtsp streams. by Andreas Huber · 14 years ago
  5. 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
  6. 7a747b8 Initial checkin of preliminary rtsp support for stagefright. by Andreas Huber · 14 years ago