| 1d15ab58bf8239069ef343de6cb21aabf3ef7d78 |
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05-Mar-2015 |
Lajos Molnar <lajos@google.com> |
media: switch to new AMessage handling
Bug: 19607784
Change-Id: I94cddcb81f671422ad4982a23dc4acfe57a9f1aa
/frameworks/av/media/libstagefright/rtsp/ARTPConnection.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/ARTPConnection.cpp
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| f6d0c1fd6d9e697bb3a891fae14c7e9d4b685de6 |
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15-Apr-2014 |
Colin Cross <ccross@google.com> |
libstagefright: fix 64-bit warnings
%lld -> %" PRId64 " for int64_t
%d -> %zu for size_t
Also fixes some casts from void* to integer types, and some comparisons
between signed and unsigned.
(cherry picked from commit b4a7a2df4c28c3f32b5d877b54831d2cc5d78f81)
Change-Id: I76ba94d0b67776fd7abdc83b43d47c61d6c32f4c
/frameworks/av/media/libstagefright/rtsp/ARTPConnection.cpp
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| 6cb3f224d7e2280f8834d361bba1a72682aaaad1 |
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24-Apr-2013 |
Yajun Zeng <beanz@marvell.com> |
Fix overflow of rand in ARTPConnection
without this fix, (rand()*1000)/RAND_MAX is mainly 0.
Change-Id: I48ae940a7b6974b197d81732774c9dcea107bcf1
Signed-off-by: Yajun Zeng <beanz@marvell.com>
/frameworks/av/media/libstagefright/rtsp/ARTPConnection.cpp
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| 3677437296fd1547d762b1b227a3de83dbc960d6 |
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27-Jul-2012 |
Tareq A. Siraj <tareq.a.siraj@intel.com> |
Fixed member access into incomplete type build error
Included the ARTPAssembler.h file to fix the 'member access into
incomplete type "android::ARTPAssembler"' error reported by clang.
Change-Id: I10cb1e38bf360858bb7ebdeae82ba1e64431f87d
Author: Tareq A. Siraj <tareq.a.siraj@intel.com>
Reviewed-by: Edwin Vane<edwin.vane@intel.com>
/frameworks/av/media/libstagefright/rtsp/ARTPConnection.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/ARTPConnection.cpp
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| 5ff1dd576bb93c45b44088a51544a18fc43ebf58 |
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06-Jan-2012 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGW(_IF) to (IF_)ALOGW(_IF) DO NOT MERGE
See https://android-git.corp.google.com/g/157065
Bug: 5449033
Change-Id: I00a4b904f9449e6f93b7fd35eac28640d7929e69
/frameworks/av/media/libstagefright/rtsp/ARTPConnection.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/ARTPConnection.cpp
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| 8c308ffd781132c8417cebc3bf77c2e56a464e0b |
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09-Nov-2011 |
Andreas Huber <andih@google.com> |
Instead of asserting, remove active streams if their sockets return failure
Change-Id: Ic5cc786f718cf921876b181927cf1b03e8373ff1
related-to-bug: 5593654
/frameworks/av/media/libstagefright/rtsp/ARTPConnection.cpp
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| 3856b090cd04ba5dd4a59a12430ed724d5995909 |
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20-Oct-2011 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGV(_IF) to (IF_)ALOGV(_IF) DO NOT MERGE
See https://android-git.corp.google.com/g/#/c/143865
Bug: 5449033
Change-Id: I0122812ed6ff6f5b59fe4a43ab8bff0577adde0a
/frameworks/av/media/libstagefright/rtsp/ARTPConnection.cpp
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| dc468c5f9d72ce54de0070493e9a23efb8907e06 |
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15-Feb-2011 |
Andreas Huber <andih@google.com> |
Work around several issues with non-compliant RTSP servers.
In this particular case these RTSP servers were implemented as local services,
retransmitting live streams via a local RTSP server instance.
They picked wrong rtp/rtcp port pairs (odd rtp port), blank lines in the session
description, wrong case of the format description, relative base URLs...
Change-Id: I63fa90ca2398f19e8b52c147248bd2c5c2372426
related-to-bug: 3452103
/frameworks/av/media/libstagefright/rtsp/ARTPConnection.cpp
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| 100a4408968b90e314526185d572c72ea4cc784a |
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08-Feb-2011 |
Andreas Huber <andih@google.com> |
Change timestamp handling in RTSP, remove unused, experimental, gtalk support
related-to-bug: 3216447
NTP timestamp handling is now done at a higher layer than before.
Change-Id: I9fb23f1335110ec59e534f9aa0fe6f6a6406dd52
/frameworks/av/media/libstagefright/rtsp/ARTPConnection.cpp
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| f61551f4fc79e7da879802e3974afa9b03ffb5d0 |
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13-Oct-2010 |
Andreas Huber <andih@google.com> |
Some webcams output rtp streams but never send any rtcp data in violation of
the specs. Attempt to use fake timestamps to be able to play these...
Change-Id: Ia7a926616fb764e972955df4acdb59d85cdd93df
related-to-bug: 3087310
/frameworks/av/media/libstagefright/rtsp/ARTPConnection.cpp
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| 6e4c5c499999c04c2477b987f9e64f3ff2bf1a06 |
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21-Sep-2010 |
Andreas Huber <andih@google.com> |
Remove stagefright foundation's incompatible logging interface and update callsites.
Change-Id: I45fba7d60530ea0f233ac3695a97306b6dc1795c
/frameworks/av/media/libstagefright/rtsp/ARTPConnection.cpp
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| e7d3e90d8761f52a6acfdcd926f0392aca8ebb52 |
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31-Aug-2010 |
Andreas Huber <andih@google.com> |
Better detection of connection problems - timeout if no rtcp packets arrive within a certain time, not a final frame (which may take longer)
Change-Id: I3c1ae79bb9342770e959ebdcdc6b748549b76330
related-to-bug: 2556656
/frameworks/av/media/libstagefright/rtsp/ARTPConnection.cpp
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| 7aef03379179c109c2547c33c410bfc93c8db576 |
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31-Aug-2010 |
Andreas Huber <andih@google.com> |
Instead of closing the connection altogether if no UDP packets arrive after a certain time, try changing transports (to interleaved TCP). Also properly close the sockets on disconnection.
Change-Id: Ie8d6a3865a0477e28d4b76bb9038e468451287b1
related-to-bug: 2556656
/frameworks/av/media/libstagefright/rtsp/ARTPConnection.cpp
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| 0792ce7e0924ebb0dbe7b7cfcd79d12cbdb03ed2 |
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26-Aug-2010 |
Andreas Huber <andih@google.com> |
Support for RTP packets arriving interleaved with RTSP responses.
Change-Id: Ib32fba257da32a199134cf8943117cf3eaa07a25
/frameworks/av/media/libstagefright/rtsp/ARTPConnection.cpp
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| 62cb04d23642a2ea7c005f050494c8ef3c370dd3 |
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19-Aug-2010 |
Andreas Huber <andih@google.com> |
Support for MP4V-ES packetization format according to RFC3016.
Change-Id: I5e182936c52f9eb80cdcf6132ead03705ee32d61
/frameworks/av/media/libstagefright/rtsp/ARTPConnection.cpp
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| f8ca90452ff3e252f20de38f1c3eee524c808c3e |
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10-Aug-2010 |
Andreas Huber <andih@google.com> |
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.
Change-Id: Idc3df74b42000f7a6aa3eae090718dc9d9c4186f
/frameworks/av/media/libstagefright/rtsp/ARTPConnection.cpp
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| ff53123821a3ec2e71fdb1a971ea2cbae3119826 |
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05-Aug-2010 |
Andreas Huber <andih@google.com> |
Better support for fake timestamps in RTP, H.263 video now also requests FIR.
Change-Id: I2385461887197fe4062d329086e0204f6d6620fc
/frameworks/av/media/libstagefright/rtsp/ARTPConnection.cpp
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| 39ddf8e0f18766f7ba1e3246b774aa6ebd93eea8 |
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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/ARTPConnection.cpp
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| cf7b9c7aae758ac0b99833915053c63c2ac46e09 |
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08-Jun-2010 |
Andreas Huber <andih@google.com> |
Initial checkin of preliminary rtsp support for stagefright.
Change-Id: I0722aa888098c0c1361c97a4c1b123d910afc207
/frameworks/av/media/libstagefright/rtsp/ARTPConnection.cpp
|