History log of /frameworks/base/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp
Revision Date Author Comments (<<< Hide modified files) (Show modified files >>>)
8c7c6dc413e12b2394aae06bd2ccc3db7a29c710 22-Feb-2011 Andreas Huber <andih@google.com> Support more MPEG4-LATM audio functionality.

related-to-bug: 3474610

Change-Id: I6dab40e8b465922c62be9ee7f168718822c6caac
Now skipping extra header that the spec claimed shouldn't be present in LATM...
/frameworks/base/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp
0f535afd7e177b9a133f3ef4d014042797b225ff 27-Jan-2011 Andreas Huber <andih@google.com> This particular RTSP server streams MPEG4-LATM audio with extra trailing bytes.

And now we're just ignoring them. Yay standards.

Change-Id: I76529ad8d585f143d6f99621ff671d179caf7b35
related-to-bug: 3353752
/frameworks/base/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp
b0d25a00fe28d3153d4c56b24d8e2792230d68be 27-Oct-2010 Andreas Huber <andih@google.com> 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.

Change-Id: I20e3b86f52b7f0f41663ffe8bc1f4db92280e884
/frameworks/base/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp
eeb97d91b97f1fc0b26815f098515e9c06d219b8 27-Aug-2010 Andreas Huber <andih@google.com> Better support for rtsp (normal play-)time display. Better seek support, timeout if no packets arrive for too long.

Change-Id: Id491541a6ae501604cda815f8e961a3bfe26db7d
related-to-bug: 2556656
/frameworks/base/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp
57648e4eec7dd2593af467877bc7cce4aa654759 04-Aug-2010 Andreas Huber <andih@google.com> 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.

Change-Id: I95b8dd487061add9bade15749e563b01cd99d9a6
/frameworks/base/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp
4e4173b0af52bdf2b5730a5837476e400c5b2040 22-Jul-2010 Andreas Huber <andih@google.com> Various changes to improve rtsp networking, reduce packet loss and adapt to ALooper API changes.

Change-Id: I110e19d5ce33e597add3ffbd3e3ff3815862396d
/frameworks/base/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp
7a747b8e0dadf909ea4ac0b67fd88fc14b4eb3f8 08-Jun-2010 Andreas Huber <andih@google.com> Initial checkin of preliminary rtsp support for stagefright.

Change-Id: I0722aa888098c0c1361c97a4c1b123d910afc207
/frameworks/base/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp