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gerrit-public.fairphone.software / platform / frameworks / base / ae7752798a98fc81ff5e6ae69dde2137692106be / media / libstagefright / rtsp / ARTPSource.cpp
  1. b672373 Support for PCMA and PCMU raw audio data in RTP/RTSP. by Andreas Huber · 13 years ago
  2. 0407269 Work around several issues with non-compliant RTSP servers. by Andreas Huber · 13 years ago
  3. b2934b1 Change timestamp handling in RTSP, remove unused, experimental, gtalk support 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. 6e3fa44 Remove stagefright foundation's incompatible logging interface and update callsites. by Andreas Huber · 14 years ago
  6. e536f80 Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr. by Andreas Huber · 14 years ago
  7. a979ad6 Support for MP4V-ES packetization format according to RFC3016. by Andreas Huber · 14 years ago
  8. 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
  9. 3eaa300 Better support for fake timestamps in RTP, H.263 video now also requests FIR. by Andreas Huber · 14 years ago
  10. 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
  11. 7a747b8 Initial checkin of preliminary rtsp support for stagefright. by Andreas Huber · 14 years ago