| b3f9759c8c9437c45b9a34519ce2ea38a8314d4e |
|
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/APacketSource.cpp
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| 1b86fe063badb5f28c467ade39be0f4008688947 |
|
29-Jan-2014 |
Andreas Huber <andih@google.com> |
FINAL ATTEMPT: HTTP services are now provided from JAVA and made available to media code
Change-Id: I9f74a86e70422187c9cf0ca1318a29019700192d
/frameworks/av/media/libstagefright/rtsp/APacketSource.cpp
|
| cfaeeec0900014d97e15829e0fa52f865ee4c786 |
|
31-Aug-2012 |
Andreas Huber <andih@google.com> |
Add support for mpeg2 transport streams to the RTSP implementation.
Change-Id: I409d7133a53a71e62523b1acc2b03302fcf824a5
/frameworks/av/media/libstagefright/rtsp/APacketSource.cpp
|
| 8033393a74a6872ad8d702b10da34d98dde0bf41 |
|
20-Aug-2012 |
Patrik2 Carlsson <patrik2.carlsson@sonymobile.com> |
h264 streaming: make profile-level-id optional
profile-level-id is made optional according to rfc3984:
"If no profile-level-id is present, the Baseline Profile without
additional constraints at Level 1 MUST be implied."
Change-Id: If868468a48917ceccb963b8ac15767583da29723
/frameworks/av/media/libstagefright/rtsp/APacketSource.cpp
|
| 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/APacketSource.cpp
|
| df64d15042bbd5e0e4933ac49bf3c177dd94752c |
|
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/APacketSource.cpp
|
| 2bfdd428c56c7524d1a11979f200a1762866032d |
|
12-Oct-2011 |
Andreas Huber <andih@google.com> |
NuPlayer is now taking on the task of streaming over RTSP.
Change-Id: Ie204db8810807f1e7981959e34dc0149e5d9563a
/frameworks/av/media/libstagefright/rtsp/APacketSource.cpp
|
| f89d780df70b7fbb8465bce4913c46cca019721f |
|
05-Aug-2011 |
Andreas Huber <andih@google.com> |
Eliminate superfluous memcpys by wrapping an ABuffer in a MediaBuffer
Change-Id: I1313f117cd7cdfaf7d6ec25413a0b4b8ea495037
related-to-bug: 5122973
/frameworks/av/media/libstagefright/rtsp/APacketSource.cpp
|
| 386d609dc513e838c7e7c4c46c604493ccd560be |
|
19-May-2011 |
Andreas Huber <andih@google.com> |
Support mpeg1,2 audio and mpeg1,2,4 video content extraction from .ts streams.
Change-Id: I9d2ee63495f161e30daba7c3aab16cb9d8ced6a5
/frameworks/av/media/libstagefright/rtsp/APacketSource.cpp
|
| fcea8f7a7d178e5426aa06586cff54726e18d1f6 |
|
23-Feb-2011 |
Andreas Huber <andih@google.com> |
Support for PCMA and PCMU raw audio data in RTP/RTSP.
Change-Id: Icb87bdfa7cf572c572e2a86c46fa072d9fad18f6
related-to-bug: 3084183
/frameworks/av/media/libstagefright/rtsp/APacketSource.cpp
|
| dc468c5f9d72ce54de0070493e9a23efb8907e06 |
|
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/APacketSource.cpp
|
| 21a6f9ffee8b3c014abfe165b8f7fd2224f49e1f |
|
18-Jan-2011 |
Andreas Huber <andih@google.com> |
Implement parsing of vbv buffering info in RTSP.
Change-Id: I7d871cafda2c4c65670a40ad9ab4f24317f8568a
related-to-bug: 3351915
/frameworks/av/media/libstagefright/rtsp/APacketSource.cpp
|
| c68a48c474f609df3eeb7d9738675d6ac8835e0a |
|
08-Oct-2010 |
Andreas Huber <andih@google.com> |
Refactor some more h.264 utility code out into avc_utils. Work around a hardware decoder issue by making sure the first access unit submitted to a decoder at startup or after seek is an IDR.
Change-Id: I61936601e55df7e4c23a8c13087579a4f85bd6e6
/frameworks/av/media/libstagefright/rtsp/APacketSource.cpp
|
| 6e4c5c499999c04c2477b987f9e64f3ff2bf1a06 |
|
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/APacketSource.cpp
|
| 6f85dba3768089679ff5e35ad2f1841918d0adb2 |
|
15-Sep-2010 |
Andreas Huber <andih@google.com> |
Various fixes to improve resilience of the rtsp stack against spurious errors instead of asserting.
Change-Id: Idbec5996ed0675c70e911b9c0514961fea099fb4
/frameworks/av/media/libstagefright/rtsp/APacketSource.cpp
|
| c9e894872c298b25fe9d74e68aa1e7287a541ac3 |
|
02-Sep-2010 |
Andreas Huber <andih@google.com> |
Better support for buffered streaming of rtsp content, if buffer drops below a certain threshold we will temporarily pause playback until we have sufficient data.
Change-Id: Ice8564e902e48c89c9c00f6651c5504b3c41fcad
related-to-bug: 2556656
/frameworks/av/media/libstagefright/rtsp/APacketSource.cpp
|
| 4dba3e90f211eb5f5af19b10c5d3fc8c967b0086 |
|
31-Aug-2010 |
Andreas Huber <andih@google.com> |
Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr.
Change-Id: Ied92ea8c2448a2cb1a732c72c21c69da1913dbc8
related-to-bug: 2556656
/frameworks/av/media/libstagefright/rtsp/APacketSource.cpp
|
| 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/APacketSource.cpp
|
| cce326fe43411855aca2f719e505b051bc4b61b3 |
|
24-Aug-2010 |
Andreas Huber <andih@google.com> |
A first shot at proper support for seeking of rtsp streams.
Change-Id: I9604f2d09feedc0074c0e715be58e719d4483760
related-to-bug: 2556656
/frameworks/av/media/libstagefright/rtsp/APacketSource.cpp
|
| 62cb04d23642a2ea7c005f050494c8ef3c370dd3 |
|
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/APacketSource.cpp
|
| 85f12e9b9062402d6110df3f7099707912040edb |
|
19-Aug-2010 |
Andreas Huber <andih@google.com> |
In the absence of width/height information in the sdp, extract the dimensions from the avc codec specific data.
Change-Id: I98c4194593c7e6e24f6fc339c862245111800293
/frameworks/av/media/libstagefright/rtsp/APacketSource.cpp
|
| ef7af7fec702db2fde72b16dedf9064585e6db77 |
|
18-Aug-2010 |
Andreas Huber <andih@google.com> |
Better support for rtsp streamed through VLC. Temporarily make the socket blocking to read all of the session description.
Change-Id: Ibe71f5941485660510e24d714da3865b9c6f89a2
/frameworks/av/media/libstagefright/rtsp/APacketSource.cpp
|
| 6dc387a8c3f031f9f17d1138295368946563f7a5 |
|
12-Aug-2010 |
Andreas Huber <andih@google.com> |
APacketSource is too verbose.
Change-Id: I48ca7b070d89e43405d05e5f41e650db587e12b4
/frameworks/av/media/libstagefright/rtsp/APacketSource.cpp
|
| f8ca90452ff3e252f20de38f1c3eee524c808c3e |
|
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/APacketSource.cpp
|
| 33a8457868eb00b94b37b53321a80d9307202a9d |
|
04-Aug-2010 |
Andreas Huber <andih@google.com> |
Specification of codec specific data as part of the session description is now optional.
Change-Id: Ie1953909e1d241381add3cc82a7a1f7d7d1540f2
/frameworks/av/media/libstagefright/rtsp/APacketSource.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/APacketSource.cpp
|
| 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/APacketSource.cpp
|