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path: root/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp
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* Use default values when MPEG4 audio config parsing fails.Erik Rydgren2012-11-191-3/+10
| | | | | | | | | | | | | | | MPEG4 audio packets may be multiplexed using the so called LATM (Low Overhead Audio Transport Multiplex) scheme. LATM parsing was recently introduced in Stagefright and it has caused issues in cases when the LATM config element cannot be parsed correctly. The main problem occurrs when the AudioSpecificConfig part of the config element contains more information than what is expected, causing the frameLengthType parameter to get the wrong value. This fix introduces default values of some config parameters that are used in case config parsing fails. Change-Id: I3cb35df76826f95ca0831dc08c2a1e7c6c2c586d
* Prefix MPEG4-generic audio data with ADTS headersAndreas Huber2012-05-171-21/+1
| | | | | | | to work around limitations of the new AAC decoder. Change-Id: I4988c7c39fedb7d04eb1ae2ba2d618aa6cb14e77 related-to-bug: 6488547
* Add new APIs AMessage::(set|find)Buffer to make it safer to passAndreas Huber2012-02-221-1/+1
| | | | | | ABuffer objects through messages. Change-Id: I9f8b4e4c4767d0d70a0105e0c0813b754379b49d
* Rename (IF_)LOGI(_IF) to (IF_)ALOGI(_IF) DO NOT MERGESteve Block2012-01-041-3/+3
| | | | | | | See https://android-git.corp.google.com/g/156801 Bug: 5449033 Change-Id: Ib08fe86d23db91ee153e9f91a99a35c42b9208ea
* Support more MPEG4-LATM audio functionality.Andreas Huber2011-02-221-9/+74
| | | | | | | related-to-bug: 3474610 Change-Id: I6dab40e8b465922c62be9ee7f168718822c6caac Now skipping extra header that the spec claimed shouldn't be present in LATM...
* This particular RTSP server streams MPEG4-LATM audio with extra trailing bytes.Andreas Huber2011-01-261-1/+4
| | | | | | | And now we're just ignoring them. Yay standards. Change-Id: I76529ad8d585f143d6f99621ff671d179caf7b35 related-to-bug: 3353752
* Better support for MP4A-LATM RTP disassembly. This used to fail if ↵Andreas Huber2010-10-271-17/+367
| | | | | | mNumSubFrames > 1 and the sub frames did not align with RTP packet boundaries. Change-Id: I20e3b86f52b7f0f41663ffe8bc1f4db92280e884
* Better support for rtsp (normal play-)time display. Better seek support, ↵Andreas Huber2010-08-271-5/+1
| | | | | | | timeout if no packets arrive for too long. Change-Id: Id491541a6ae501604cda815f8e961a3bfe26db7d related-to-bug: 2556656
* Support for Gtalk video, includes AMR/H.263 assembler and packetization ↵Andreas Huber2010-08-041-0/+6
| | | | | | 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
* Various changes to improve rtsp networking, reduce packet loss and adapt to ↵Andreas Huber2010-07-221-0/+5
| | | | | | ALooper API changes. Change-Id: I110e19d5ce33e597add3ffbd3e3ff3815862396d
* Initial checkin of preliminary rtsp support for stagefright.Andreas Huber2010-06-071-0/+166
Change-Id: I0722aa888098c0c1361c97a4c1b123d910afc207