| 8647bbe4420ca487467318404127f52c567e346b |
|
17-May-2012 |
Andreas Huber <andih@google.com> |
Prefix MPEG4-generic audio data with ADTS headers
to work around limitations of the new AAC decoder.
Change-Id: I4988c7c39fedb7d04eb1ae2ba2d618aa6cb14e77
related-to-bug: 6488547
/frameworks/av/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp
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| 2d8bedd05437b6fccdbc6bf70f673ffd86744d59 |
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21-Feb-2012 |
Andreas Huber <andih@google.com> |
Add new APIs AMessage::(set|find)Buffer to make it safer to pass
ABuffer objects through messages.
Change-Id: I9f8b4e4c4767d0d70a0105e0c0813b754379b49d
/frameworks/av/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp
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| df64d15042bbd5e0e4933ac49bf3c177dd94752c |
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04-Jan-2012 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGI(_IF) to (IF_)ALOGI(_IF) DO NOT MERGE
See https://android-git.corp.google.com/g/156801
Bug: 5449033
Change-Id: Ib08fe86d23db91ee153e9f91a99a35c42b9208ea
/frameworks/av/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp
|
| 55e26193c885b7d5acdae9978848e6587987790f |
|
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/av/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp
|
| 9202cca86e9017cc5ce30970c92a91ab32a0835e |
|
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/av/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp
|
| fc9ac988e08a8b4c42e58999300265989f26f24c |
|
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/av/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp
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| 8d342970108926c4ea355c90d26a2a353ec0fd47 |
|
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/av/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp
|
| 39ddf8e0f18766f7ba1e3246b774aa6ebd93eea8 |
|
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/av/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp
|
| 348a8eab84f4bba76c04ca83b2f5418467aa1a48 |
|
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/av/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp
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| cf7b9c7aae758ac0b99833915053c63c2ac46e09 |
|
08-Jun-2010 |
Andreas Huber <andih@google.com> |
Initial checkin of preliminary rtsp support for stagefright.
Change-Id: I0722aa888098c0c1361c97a4c1b123d910afc207
/frameworks/av/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp
|