| 8aa5af703ba29d7bd7b5efaefd76aac2d568f11f |
|
19-Apr-2011 |
Sunil Shah <sunil.shah@sonyericsson.com> |
Handle malformed audio packets received during RTSP stream switching
During RTSP stream switching (for example channel switching in a
Mobile TV application) we occasionally receive packets that
don't contain valid data, so we cannot remove LATM framing (as per
the MPEG4 Audio Assembler). This fix allows the frame remover
to exit gracefully (instead of crashing), when such frames are
encountered, and as a consequence, Mobile TV apps can change
channels properly.
Change-Id: Ie4c3d2766c87b43f31624192de96bc47180ca514
/frameworks/av/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp
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| 99187615b2cb42e39842083c2998a97e8277a5d5 |
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20-Aug-2015 |
Abhishek Arya <aarya@google.com> |
am d3b6f9c1: am c5b9a48f: am a27fe8d7: am 2fd79fa3: am cb2acbfe: am 635d38a8: Merge "Check RTSP payload length" into klp-dev
* commit 'd3b6f9c17ed10df01d682b0fac6b13fca396e5fb':
Check RTSP payload length
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| 4d46f6f18f5160b8992ec1e66ef1844212fc7d48 |
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20-Aug-2015 |
Marco Nelissen <marcone@google.com> |
Check RTSP payload length
Bug: 23346388
Change-Id: Ifd918cefc90527c2f52177c3ce0da7a13259ad08
/frameworks/av/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp
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| cd8d9c6cfe5ca53f17be1ea8edac6b324e203f52 |
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09-Dec-2014 |
Chih-Hung Hsieh <chh@google.com> |
Fix print format mismatches.
Clang complains about mismatch of argument type and print format.
Change-Id: Ib07da09d8b1b62b3018033f9eaf7aa01bf7f7f9c
/frameworks/av/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp
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| b3f9759c8c9437c45b9a34519ce2ea38a8314d4e |
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24-Nov-2014 |
Andreas Gampe <agampe@google.com> |
Stagefright: Fix unused variables, functions, values
For build-system CFLAGS clean-up, remove unused functions and
variables.
Change-Id: Ic3dee56b589ea9a693efa1d72ba394036efff168
/frameworks/av/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp
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| a45a600d69a5d8ab99eeb7e0dfa58c3cb99a2e61 |
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19-Sep-2011 |
Erik Rydgren <erik.rydgren@sonyericsson.com> |
Use default values when MPEG4 audio config parsing fails.
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
/frameworks/av/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp
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| 8af5fe5a2431522a7d30bc546dcd31c0c64db70c |
|
19-Sep-2011 |
Erik Rydgren <erik.rydgren@sonyericsson.com> |
Use default values when MPEG4 audio config parsing fails.
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
/frameworks/av/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp
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| 8647bbe4420ca487467318404127f52c567e346b |
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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
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| 55e26193c885b7d5acdae9978848e6587987790f |
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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
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| 9202cca86e9017cc5ce30970c92a91ab32a0835e |
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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
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| 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 |
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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
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| 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
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| 348a8eab84f4bba76c04ca83b2f5418467aa1a48 |
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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
|