Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / frameworks / base / 0df0ed5814a83c4cf0dd3d6d6e177f5ed9003863 / media / libstagefright / rtsp / AMPEG4AudioAssembler.cpp
  1. 6215d3f Rename (IF_)LOGI(_IF) to (IF_)ALOGI(_IF) DO NOT MERGE by Steve Block · 13 years ago
  2. 8c7c6dc Support more MPEG4-LATM audio functionality. by Andreas Huber · 13 years ago
  3. 0f535af This particular RTSP server streams MPEG4-LATM audio with extra trailing bytes. by Andreas Huber · 14 years ago
  4. b0d25a0 Better support for MP4A-LATM RTP disassembly. This used to fail if mNumSubFrames > 1 and the sub frames did not align with RTP packet boundaries. by Andreas Huber · 14 years ago
  5. 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
  6. 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
  7. 4e4173b Various changes to improve rtsp networking, reduce packet loss and adapt to ALooper API changes. by Andreas Huber · 14 years ago
  8. 7a747b8 Initial checkin of preliminary rtsp support for stagefright. by Andreas Huber · 14 years ago