| 820c4893fdec784321826fd903da34fe3d609b93 |
|
23-Sep-2014 |
Wei Jia <wjia@google.com> |
MyHandler: set ip address to an invalid one when getsockname() returns error.
Bug: 17556472
Change-Id: I0387c78727d9a18abddcfdb4b480f4b1412bbc9f
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| beb57a5a08207af80180b93dd80d611a85997c43 |
|
14-Mar-2014 |
Andreas Huber <andih@google.com> |
am f1ac623f: am 4a67fc49: Merge "Implemented support for RTSP 301 Redirect"
* commit 'f1ac623fcc6bbda2faff9752cd611182a897afe1':
Implemented support for RTSP 301 Redirect
|
| 4a67fc49d926c75fa6a96160ba5627fb0e209db6 |
|
14-Mar-2014 |
Andreas Huber <andih@google.com> |
Merge "Implemented support for RTSP 301 Redirect"
|
| fca092d953e04c7169242200f0ddb914a9f54ea4 |
|
12-Mar-2014 |
Marco Nelissen <marcone@google.com> |
am f4431278: am 19afb386: Merge "Remove streaming URI from default logs"
* commit 'f4431278a9613f55ecd944ab2e3eb615b372f269':
Remove streaming URI from default logs
|
| a8b8488f703bb6bda039d7d98f87e4f9d845664d |
|
06-Sep-2012 |
David Williams <david.williams@sonymobile.com> |
Remove streaming URI from default logs
Streaming URI should not be visible in default logcat logs
Change-Id: I104cc56b5335f8c5621013e4c5be8028f0379833
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| 84333e0475bc911adc16417f4ca327c975cf6c36 |
|
08-Feb-2014 |
Andreas Huber <andih@google.com> |
warnings be gone.
Change-Id: Ie3bae3f037730e316d7fca12e7a3527973f752ef
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| 9843e8c9446aec0c25168ff4561bdbb12948f1c7 |
|
25-Sep-2013 |
Chong Zhang <chz@google.com> |
am 58dd0786: Merge "Send kWhatConnected in onTimeUpdate() before first access unit" into klp-dev
* commit '58dd07863571951408b67fa0a7f17cb23606fb1c':
Send kWhatConnected in onTimeUpdate() before first access unit
|
| ffd5687c9ece8e28779793a20f06f99c7199ce44 |
|
24-Sep-2013 |
Chong Zhang <chz@google.com> |
Send kWhatConnected in onTimeUpdate() before first access unit
Bug: 10642588
Change-Id: If2b4fbbf250d5307e304f31c7aa4ac480e279484
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| cb18b6987bb3c928b2ec69e344923b427ed39627 |
|
28-Aug-2013 |
Andreas Huber <andih@google.com> |
am af66fae1: am fb949d5d: Merge "Fix crash in MyHandler when sockets are not set."
* commit 'af66fae15f8c386ad884e5fa83db4eaef4c4f2ee':
Fix crash in MyHandler when sockets are not set.
|
| fb949d5dc8a764e31fbd65bee87f59fcfeb6d848 |
|
28-Aug-2013 |
Andreas Huber <andih@google.com> |
Merge "Fix crash in MyHandler when sockets are not set."
|
| 59d3f809024ae5b5a7ea35dcfdd056f1c7ca42b2 |
|
23-Jul-2013 |
Chad Brubaker <cbrubaker@google.com> |
Fix typo in socket name
Change-Id: I29171368f1b69333ef7eae53ada2fab94e3e28b9
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| 5908f88a7e45380a9b0d71a3b1ea535d76c420b3 |
|
16-Jul-2013 |
Chad Brubaker <cbrubaker@google.com> |
Add routing sockets for the requesting user
Mediaserver sockets are now routed as if the connection was in the
requesting app in per user routing.
Change-Id: I60f4649c3c4145a65264b54c1aa2c6c7741efaba
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| c582fde93ded7219107157333a9e46d780adcf9c |
|
08-Jul-2013 |
Jean-Baptiste Queru <jbq@google.com> |
resolved conflicts for merge of c158971f to stage-aosp-master
Change-Id: I3d77b86f7e616af62a826fc37126706ad8ff6158
|
| bbbf9c4552402ab18b255f4058e9e6e506f3f106 |
|
24-Apr-2013 |
Yajun Zeng <beanz@marvell.com> |
Store rtsp accessunit until PLAY response parsed
If RTP accessunit comes earlier than play response,
the normal play time mapping posted in func onAccessUnitComplete is wrong.
This leads wrong timestamp of the first few frames.
This issue is found in the 3 CtsVerifier RTSP streaming cases.
Change-Id: I640eea375b1f3f4730238f9d561c3b40ec682395
Signed-off-by: Yajun Zeng <beanz@marvell.com>
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| 190cdbab6ba24519d6b5e8bec6c2c74e6650e284 |
|
26-Mar-2013 |
Andreas Huber <andih@google.com> |
Identify network servers and clients with a OS version related string
and put the logic to create that string in one location instead of many...
Change-Id: I1f729f2e7376cd3b45eea0e48f7bd10084b41b39
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| 4f4c2655dc3f6fcef766db6e793b1642ad0fd605 |
|
15-Mar-2013 |
Andreas Huber <andih@google.com> |
am 59ac7b30: am 66abe3ae: Merge "Fix for crash if no content in DESCRIBE response"
* commit '59ac7b3056db57e5a8e851b7946a181c5fc34852':
Fix for crash if no content in DESCRIBE response
|
| 5f1897538bab324f53efc6bec65487516041f2e9 |
|
07-Jan-2013 |
Xuefei Chen <xuefei.chen@sonymobile.com> |
Fix for crash if no content in DESCRIBE response
If DESCRIBE response is received with status 200 but no content,
MyHandler will still set content data for session description
parsing. This will cause NULL Pointer crash.
This fix checks whether DESCRIBE response has content before
parsing session description.
Change-Id: I114ae6fd54ce804e61718f62618ca9008425a433
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| d32b7b479fad359d7fe779a9c5b4c090cdc14b56 |
|
07-Jan-2013 |
Xuefei Chen <xuefei.chen@sonymobile.com> |
Fix for crash if no content in DESCRIBE response
If DESCRIBE response is received with status 200 but no content,
MyHandler will still set content data for session description
parsing. This will cause NULL Pointer crash.
This fix checks whether DESCRIBE response has content before
parsing session description.
Change-Id: I114ae6fd54ce804e61718f62618ca9008425a433
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| 0955986e6c1c27ba752e293246086ea79c49d39c |
|
23-Jan-2013 |
Roger Jönsson <roger1.jonsson@sonymobile.com> |
Avoid rebuffering after RTSP pause
If pausing an RTSP stream, an RTSP Pause request is sent and then
if the stream is immediately resumed again, an RTSP Play request
will be sent to the server.
But the new data after the pause will not be buffered until
Sender Reports have arrived again on both channels.
Meanwhile the player will resume playback and continue consuming
the already existing buffer.
This means that there is a risk that the buffer is emptied while
waiting for sender reports.
This commit simply adds a delay before the RTSP pause request is
sent, allowing some additional RTSP buffering that might be needed
when the stream is resumed again.
Also, if the stream is resumed again before the RTSP pause request
is sent, there is no need for any RTSP pause request, hence it is
omitted.
Change-Id: I928c8bfb5e99a6a146dcda4e51e528973ecbe065
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| 1a37ee3c877165c812734b405f922f6e0d747052 |
|
23-Jan-2013 |
joakim johansson <joakim.c.johansson@sonyericsson.com> |
EOS fixes for RTSP streams
The fix takes care of several near end of stream use cases:
seek, pause and fake timestamps.
Change-Id: I5f5fa881b1f619dfd5e1afd2af957082345c59eb
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| b6ec588faa7728ff3b518bf809ff75e8dd14f08c |
|
23-Jan-2013 |
Måns Zigher <mans.zigher@sonyericsson.com> |
RTSP: Parse session level control attribute from SDP
If a=control: is present at session-level in the SDP response,
RFC2326:C.1.1 defines the URL to be used for aggregate commands.
This includes PLAY and PAUSE but not TEARDOWN.
Change-Id: Iaa1dc2271d00df39dc83477a99fda6fbeb73c5b4
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| 46d13e3606b87d71379287672b54b50d0d9aa5cc |
|
21-Jan-2013 |
Roger Jönsson <roger1.jonsson@sonymobile.com> |
Enable pause/resume for RTSP streaming
When a stream is paused, RTSP Pause is also sent to the server.
Otherwise the buffering might continue until the memory runs out.
When the stream is resumed, RTSP Play will be sent in order to
resume the buffering.
Change-Id: I5dc1761140827c532451638c3fd3f34271e5b9ab
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| cfc3083927df14bf82403b20a45ae303a01c39f5 |
|
21-Jan-2013 |
Roger Jönsson <roger1.jonsson@sonymobile.com> |
RTSP buffering improvements
Added buffering start and end notifications for RTSP.
MEDIA_INFO_BUFFERING_START is sent when buffering is started
and MEDIA_INFO_BUFFERING_END is sent when the buffer has
filled up.
This patch also adds RTSP end of stream handling.
EOS is signalled when BYE is received OR when
detecting end of stream even if no actual EOS is received.
Change-Id: I5cccb6845060ae6afd66d9f735b89da81476cd13
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| 7f475c34ffc8e35345f2cceee2ef56a50bb5fea6 |
|
05-Feb-2013 |
Andreas Huber <andih@google.com> |
RTSP now properly publishes its "seekable" flags after connection
has successfully completed and only then signals that preparation is
complete.
Change-Id: I1a60f718e673fe1462c69369c40eafbed6a14326
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| ec29a2bfb5364a5968b77559fd13821b827d173a |
|
17-Jan-2013 |
Roger Jönsson <roger1.jonsson@sonymobile.com> |
Detect live streams
The information is used to decide on visibility of pause button and
to handle the duration clock correctly.
Change-Id: I286ac992fd171c7fc313e429326d38b6fc80e3fb
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| 81dd60e0340ddcf7f1d5fb80b6c30906fabf201a |
|
20-Feb-2012 |
Oscar Rydhé <oscar.rydhe@sonyericsson.com> |
Added HTTP support for SDP files.
Added support for playing SDP files from http links. Previously,
SDP files only worked when started from rtsp links
(rtsp://a.b.c/abc.sdp), but they are just as common in http links.
patch provided by "Oscar Rydhé <oscar.rydhe@sonyericsson.com>"
Change-Id: Ic73af3a9a002009dbe8b04c267a4621bf7fe2f46
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| b6f7642496f955da04d1eb9e33df0dab653c9c4e |
|
20-Sep-2011 |
Henrik Backlund <henrik.backlund@sonyericsson.com> |
Fix crash in MyHandler when sockets are not set.
-When going quickly in and out of the video view during an rtsp
streaming session, a race condition occurs and MyHandler tries to
connect to a socket that has been reset. To avoid this,
checks are added.
- If there are errors during setupTrack 1, it is no use
setting up track 2. It will cause new errors.
- No assert for socket connect since there is a normal
status check already.
Change-Id: Ie06221d6c0d78ce0449f76c782ed5120fa646bfd
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| 4bb026ba585d5b37795bd9765459f0607b7aa60a |
|
24-Feb-2011 |
David Williams <david.williams@sonyericsson.com> |
Implemented support for RTSP 301 Redirect
RTSP 301 (Permament Redirect) support has been implemented.
Change-Id: If82ffc87f4e7dcbdf98e0a662a35cc086750fc1b
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| 738198a16cfd7b125d15b0bab0708ba7fbf7e64a |
|
26-Sep-2011 |
Patric Frederiksen <patric.frederiksen@sonyericsson.com> |
Crash in android::MyHandler::parsePlayResponse
This fix handles problems with several asynchronous calls
within streaming. This case is when the phone has sent a
request to the server and while the response is being sent
back by the server the request is aborted by the user.
The fix is an if case that checks if we have aborted while
waiting for a response from the server. If we have aborted
we should ignore the late response instead of continuing.
Change-Id: I1264bb992f6abcaee1f10a89479e08b54a95e3c2
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| e1a31d16dda3460a34e5dfd65c4e96e422dbdbfc |
|
26-Sep-2011 |
Patric Frederiksen <patric.frederiksen@sonyericsson.com> |
Crash in android::MyHandler::parsePlayResponse
This fix handles problems with several asynchronous calls
within streaming. This case is when the phone has sent a
request to the server and while the response is being sent
back by the server the request is aborted by the user.
The fix is an if case that checks if we have aborted while
waiting for a response from the server. If we have aborted
we should ignore the late response instead of continuing.
Change-Id: I1264bb992f6abcaee1f10a89479e08b54a95e3c2
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| 2d8bedd05437b6fccdbc6bf70f673ffd86744d59 |
|
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/MyHandler.h
|
| 7e73e44c2d2208a7079e562f7b0b9b73ef6a29f1 |
|
20-Jan-2012 |
Andreas Huber <andih@google.com> |
Starhub RTSP apparently does not establish time on all tracks
i.e. the "SR" RTCP packet is sent for only one of the two tracks.
fake timestamps if that's the case, previously we'd only fake timestamps
if we didn't receive _any_ "SR" packets.
Change-Id: Id63d4940d453ba6c04c62e02ab9a0ad843936bc1
related-to-bug: 5669027
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| 29357bc2c0dd7c43ad3bd0c8e3efa4e6fd9bfd47 |
|
06-Jan-2012 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGE(_IF) to (IF_)ALOGE(_IF) DO NOT MERGE
See https://android-git.corp.google.com/g/#/c/157220
Bug: 5449033
Change-Id: Ic9c19d30693bd56755f55906127cd6bd7126096c
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| 5ff1dd576bb93c45b44088a51544a18fc43ebf58 |
|
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/MyHandler.h
|
| 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/MyHandler.h
|
| aa5ba9a27f4c483ee116b7b296a681f4f8e23e62 |
|
10-Dec-2011 |
Andreas Huber <andih@google.com> |
am aa82c39b: am 0ba9380a: Merge "Fix Bitreader "putBits" implementation, make sure we emulate timestamps" into ics-mr1
* commit 'aa82c39bdb4ad9c1fdcb09f3bea11be5197d3ce6':
Fix Bitreader "putBits" implementation, make sure we emulate timestamps
|
| 1906e5c7492b9cbc88601365536a69e9a490c963 |
|
08-Dec-2011 |
Andreas Huber <andih@google.com> |
Fix Bitreader "putBits" implementation, make sure we emulate timestamps
if we don't receive npt time mapping from the rtsp server (i.e. live stream)
Change-Id: I5147d665bd90c9a303ad6ffdafbf770f930f917c
related-to-bug: 5660357
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| 78ff828e28c22715f5b6c320d967744cb4f51fd4 |
|
11-Nov-2011 |
Andreas Huber <andih@google.com> |
am 8a065423: am 23217182: Merge "Didn\'t mean to check this in..." into ics-mr1
* commit '8a0654231ff36d938bc3451190cf67231195f1d0':
Didn't mean to check this in...
|
| 516fb1dad0c434fd89624c418543d35436a5374c |
|
11-Nov-2011 |
Andreas Huber <andih@google.com> |
am 40461ee7: am cd556b82: Merge "Instead of asserting, signal a runtime error if the session doesn\'t contain" into ics-mr1
* commit '40461ee70161d8568663332f72be2353b04c34e7':
Instead of asserting, signal a runtime error if the session doesn't contain
|
| 91f230461288a2a5091182ef9e17079aabf8ebaa |
|
11-Nov-2011 |
Andreas Huber <andih@google.com> |
Didn't mean to check this in...
Change-Id: Ie5a1902ff2613cd349ca5724f63a3fe3306640c7
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| f0c86a83c687074be79397e082e3775ca56641b1 |
|
10-Nov-2011 |
Andreas Huber <andih@google.com> |
Instead of asserting, signal a runtime error if the session doesn't contain
any playable tracks at all.
Change-Id: Ibbbe2fdcd53b7e020da80c84c8229856107a87e6
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| 7cad0b48243f86c516181d09185dc83223ae51d7 |
|
10-Nov-2011 |
Andreas Huber <andih@google.com> |
am 9e2949c6: am 2375d163: Merge "Send RTSP control connection keep-alive requests" into ics-mr1
* commit '9e2949c6ab4e791b5c20d5e85c3eff62f206a99b':
Send RTSP control connection keep-alive requests
|
| 908dbdee96856693decc04fa143c2ba525495d43 |
|
09-Nov-2011 |
Andreas Huber <andih@google.com> |
Send RTSP control connection keep-alive requests
default to 60 secs unless overridden by server's session-id response.
Change-Id: I7c3aff5b787dbb57cc0dccf9db3c75e5cf7e778c
related-to-bug: 5562303
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| 3856b090cd04ba5dd4a59a12430ed724d5995909 |
|
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/MyHandler.h
|
| 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/MyHandler.h
|
| a23456b306f35b9ecf973bf5818ca39295e9e029 |
|
08-Jul-2011 |
Ashish Sharma <ashishsharma@google.com> |
Network traffic accounting for chromium stack support in mediaserver.
- Atribute network activity to uid calling the mediaplayer
- Enables logging of chromium network stack in logcat
Change-Id: I2d28c8392248a056b3cee305dd4d4475ebba4337
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| dab718bba3945332dc75e268e1e7f0fe2eb91c4a |
|
14-Jul-2011 |
Andreas Huber <andih@google.com> |
Remove legacy http support from stagefright, chromium is the new hotness.
Change-Id: I6725d42d38b91e6a1cbca43174870f445aeb3d99
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| 9b80c2bdb205bc143104f54d0743b6eedd67b14e |
|
01-Jul-2011 |
Andreas Huber <andih@google.com> |
Charge network traffic to the uid of the process using the MediaPlayer.
Change-Id: I2bcb54b8232afd3fc7ee16289f37c7a7b3f23067
related-to-bug: 4517282
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| ac5767a96df9fae46a37ffba62611472135a0f6d |
|
30-Jun-2011 |
Andreas Huber <andih@google.com> |
Revert "Parse RTP-Info even for live streams."
This reverts commit d873413ff9f742f259c29d7d0b58265db6b24529.
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| a6925e6149faf4a936a5b557a769d117454413d8 |
|
01-Jun-2011 |
Andreas Huber <andih@google.com> |
Parse RTP-Info even for live streams.
Change-Id: Ib2c39ce8d5366f5ea350e71b7a54f5f7c2b510b9
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| de9a20c274983d4f7a688acb68d5dfc6a432eb10 |
|
15-Feb-2011 |
Andreas Huber <andih@google.com> |
Derive the Transport "source" attribute from the RTSP endpoint address if necessary
and continue even if we were unable to poke a hole into the firewall.
related-to-bug: 3457201
Change-Id: I0a523f38e6959bf00b8b140a70bb65fcc414c9c1
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| 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/MyHandler.h
|
| 100a4408968b90e314526185d572c72ea4cc784a |
|
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/MyHandler.h
|
| 783e5cd85d4bd40b1a04dfdfed256c5dcb2525cc |
|
28-Jan-2011 |
Andreas Huber <andih@google.com> |
More robust parsing of NPT time ranges in RTSP.
Change-Id: I3674501d2fd66aaface805c0a8678c74671a6dd3
related-to-bug: 3217210
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| 4579b7d49f6dd4f37e6043e59debfd72d69b8e7b |
|
21-Oct-2010 |
Andreas Huber <andih@google.com> |
Support for BASIC and DIGEST authentication schemes in RTSP. Support for malformed packet descriptions that end lines in LF only, instead of CRLF.
Change-Id: I57eaefdc4b300a8f56bbe5cf3a34c424e8efe63a
related-to-bug: 3084183
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| a44501ea0896c2508bd6b807185d9049be6752f3 |
|
15-Oct-2010 |
Andreas Huber <andih@google.com> |
am 8e4f3c76: am 646e0d5a: Merge "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..." into gingerbread
Merge commit '8e4f3c76dd7f5596fa2fe10bdf008d4c02353160'
* commit '8e4f3c76dd7f5596fa2fe10bdf008d4c02353160':
Some webcams output rtp streams but never send any rtcp data in violation of
|
| f61551f4fc79e7da879802e3974afa9b03ffb5d0 |
|
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/MyHandler.h
|
| 43a2b3b5fd4e15ffed4235f348d5eba168e8432c |
|
12-Oct-2010 |
Andreas Huber <andih@google.com> |
am 5b0d0630: am 1010da2e: Merge "Just in case we\'re behind a NAT router/firewall, attempt to poke holes into it for future incoming RTP/RTCP packets to pass through." into gingerbread
Merge commit '5b0d063010b364102ffb7a533e2b76ecfd9636d5'
* commit '5b0d063010b364102ffb7a533e2b76ecfd9636d5':
Just in case we're behind a NAT router/firewall, attempt to poke holes into it for future incoming RTP/RTCP packets to pass through.
|
| 2bc940b4f961e588459c83862b2c6bea314a4027 |
|
11-Oct-2010 |
Andreas Huber <andih@google.com> |
Just in case we're behind a NAT router/firewall, attempt to poke holes into it for future incoming RTP/RTCP packets to pass through.
Change-Id: Idd47968d4027f357222f19f15eecfd70fdec3282
related-to-bug: 3073813
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| 250e051e564e3b6f5a88314379d5e145a2b5615f |
|
11-Oct-2010 |
Andreas Huber <andih@google.com> |
am cac43e8a: am beffefa2: Merge "RTSP seeking is now asynchronous, MediaPlayer is not notified that the seek is complete until it actually is. Ignore seek requests on live streams." into gingerbread
Merge commit 'cac43e8a2ce59c1151d5a2028330b2a769591d22'
* commit 'cac43e8a2ce59c1151d5a2028330b2a769591d22':
RTSP seeking is now asynchronous, MediaPlayer is not notified that the seek is complete until it actually is. Ignore seek requests on live streams.
|
| 1c8ef86f2c25272488c171f1469f996ebf335edc |
|
11-Oct-2010 |
Andreas Huber <andih@google.com> |
am 14ea1048: am c5912acc: Merge "Disable the access unit timeout temporarily while a seek operation is in progress." into gingerbread
Merge commit '14ea1048e7e8a4b40836b5601bc86b91663525cb'
* commit '14ea1048e7e8a4b40836b5601bc86b91663525cb':
Disable the access unit timeout temporarily while a seek operation is in progress.
|
| 0dcd837af4169bdb6fb2a0c384722dc4f57433c6 |
|
09-Oct-2010 |
Andreas Huber <andih@google.com> |
RTSP seeking is now asynchronous, MediaPlayer is not notified that the seek is complete until it actually is. Ignore seek requests on live streams.
Change-Id: Ie61230cd60dd6c682baf72529100369ad6291189
related-to-bug: 3073955
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| a9d9dd2425c32f6868c35f49a3e8f29aafba931a |
|
08-Oct-2010 |
Andreas Huber <andih@google.com> |
Disable the access unit timeout temporarily while a seek operation is in progress.
Change-Id: I116cb76342aae4168f34ebae49ecb2301702a0ea
related-to-bug: 3073955
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| 3f94dacbd43b48bb629a79e45e738ead37c5debd |
|
22-Sep-2010 |
Andreas Huber <andih@google.com> |
am af909581: am 67738486: Merge "Remove stagefright foundation\'s incompatible logging interface and update callsites." into gingerbread
Merge commit 'af90958184fc5cfa1a4190e28bcfc4fdd4a5bcd6'
* commit 'af90958184fc5cfa1a4190e28bcfc4fdd4a5bcd6':
Remove stagefright foundation's incompatible logging interface and update callsites.
|
| 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/MyHandler.h
|
| ac5f724d00c8ac2040f01485873b6373f8994354 |
|
16-Sep-2010 |
Andreas Huber <andih@google.com> |
am 7ff94577: am 9909b948: Merge "Various fixes to improve resilience of the rtsp stack against spurious errors instead of asserting." into gingerbread
Merge commit '7ff945775210c60e6f113fb00903449cbb05c68a'
* commit '7ff945775210c60e6f113fb00903449cbb05c68a':
Various fixes to improve resilience of the rtsp stack against spurious errors instead of asserting.
|
| 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/MyHandler.h
|
| 16c4e8c778d8518af4c0cbefadc5d5b1272c1762 |
|
31-Aug-2010 |
Andreas Huber <andih@google.com> |
am e1a3cddd: am 99fa510e: Merge "Better detection of connection problems - timeout if no rtcp packets arrive within a certain time, not a final frame (which may take longer)" into gingerbread
Merge commit 'e1a3cddd94749a42457a8f32cf21f663f07e4edf'
* commit 'e1a3cddd94749a42457a8f32cf21f663f07e4edf':
Better detection of connection problems - timeout if no rtcp packets arrive within a certain time, not a final frame (which may take longer)
|
| e7d3e90d8761f52a6acfdcd926f0392aca8ebb52 |
|
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/MyHandler.h
|
| b62029edb6e0f97759ffb6d8f587267bee2dc31b |
|
31-Aug-2010 |
Andreas Huber <andih@google.com> |
am 987556bc: am abb8398e: Merge "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." into gingerbread
Merge commit '987556bc9bc1a61415b6e65bd600b8daf5b24d30'
* commit '987556bc9bc1a61415b6e65bd600b8daf5b24d30':
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.
|
| 7aef03379179c109c2547c33c410bfc93c8db576 |
|
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/MyHandler.h
|
| 9d876aca5ede85e6d9ccb82f11fae2834955c6f9 |
|
30-Aug-2010 |
Andreas Huber <andih@google.com> |
am 7ed9104c: am f6639c46: Finetune some rtsp timeout constants.
Merge commit '7ed9104c3acb172a480ebd7fd456fe69efd1ec3d'
* commit '7ed9104c3acb172a480ebd7fd456fe69efd1ec3d':
Finetune some rtsp timeout constants.
|
| c5c4286bebffa4c2a9539c8e09207c3130351531 |
|
30-Aug-2010 |
Andreas Huber <andih@google.com> |
am 6df6d606: am df992ac9: Merge "ALoopers can now be named (useful to distinguish threads)." into gingerbread
Merge commit '6df6d60681be9d524ce7fc07f2511008de424d27'
* commit '6df6d60681be9d524ce7fc07f2511008de424d27':
ALoopers can now be named (useful to distinguish threads).
|
| e56121bc4cb29c91d736eab181b1f51c4f125e78 |
|
30-Aug-2010 |
Andreas Huber <andih@google.com> |
Finetune some rtsp timeout constants.
Change-Id: Ice731c5097c2a2dee8a7f0cd45b547cd34f532c6
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| 9fbd6ae6b6d9f3eb791a3385df6fed3524531bd4 |
|
28-Aug-2010 |
Andreas Huber <andih@google.com> |
am 05c1cada: am 577615c9: Merge "Better support for rtsp (normal play-)time display. Better seek support, timeout if no packets arrive for too long." into gingerbread
Merge commit '05c1cadaeaf272a70acc889bfccd607648058470'
* commit '05c1cadaeaf272a70acc889bfccd607648058470':
Better support for rtsp (normal play-)time display. Better seek support, timeout if no packets arrive for too long.
|
| 437ab8c4b66a6c9dc47faa257df90089ebef10a9 |
|
28-Aug-2010 |
Andreas Huber <andih@google.com> |
am e25e0361: am e250c220: Merge "We accidentally always aborted after 10 secs, even if the connection was fine." into gingerbread
Merge commit 'e25e03612e1a2988ed83f24d2658cf0898fd1944'
* commit 'e25e03612e1a2988ed83f24d2658cf0898fd1944':
We accidentally always aborted after 10 secs, even if the connection was fine.
|
| a814c1fdc2acf0ed2ee3b175110f6039be7c4873 |
|
28-Aug-2010 |
Andreas Huber <andih@google.com> |
ALoopers can now be named (useful to distinguish threads).
Change-Id: Ieabaddb2e3a9e3a7a5bc36e55cd0721b60dbd50e
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| 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/MyHandler.h
|
| cc6adf524c1bb3bfaa5be464b50b8bcca899761c |
|
27-Aug-2010 |
Andreas Huber <andih@google.com> |
We accidentally always aborted after 10 secs, even if the connection was fine.
Change-Id: I3f2ae2f46ae62b84b1e253658d7182c04ee3dfae
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| 7cb54d6f0e6c89f45e3db0bd9246f35836d67b8f |
|
27-Aug-2010 |
Andreas Huber <andih@google.com> |
am 74ae6973: am 17a765a1: Merge "Support for RTP packets arriving interleaved with RTSP responses." into gingerbread
Merge commit '74ae6973f8d5b7bc7bc4a7dcac5ddce90f382cd4'
* commit '74ae6973f8d5b7bc7bc4a7dcac5ddce90f382cd4':
Support for RTP packets arriving interleaved with RTSP responses.
|
| 0792ce7e0924ebb0dbe7b7cfcd79d12cbdb03ed2 |
|
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/MyHandler.h
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| 5ac7b5def64625fdc9cfaf1bbdd013f5ada241f3 |
|
25-Aug-2010 |
Andreas Huber <andih@google.com> |
am 67ca90b3: am 6b6ae996: Merge "A first shot at proper support for seeking of rtsp streams." into gingerbread
Merge commit '67ca90b339feb8bb6889ca289a9dbc82c447b0d2'
* commit '67ca90b339feb8bb6889ca289a9dbc82c447b0d2':
A first shot at proper support for seeking of rtsp streams.
|
| 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/MyHandler.h
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| d9734dc5f25730944ec4e62bb028092e1841e4a3 |
|
24-Aug-2010 |
Andreas Huber <andih@google.com> |
am 31e71131: am 3e22ef1e: Merge "Better handling of rtsp connection and disconnection." into gingerbread
Merge commit '31e71131049c943a388134e796087e109248efcc'
* commit '31e71131049c943a388134e796087e109248efcc':
Better handling of rtsp connection and disconnection.
|
| 1b543242102ef3c28145c6ad50ee8e8ce2fb26d3 |
|
23-Aug-2010 |
Andreas Huber <andih@google.com> |
Better handling of rtsp connection and disconnection.
Change-Id: Ib126af6c14c5a212a51a5ee3c4a0a7d1860ad167
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
| 91d113e8daa9d71c4ea8afd595a3921e03787cbf |
|
21-Aug-2010 |
Andreas Huber <andih@google.com> |
am 6bcffcd2: am 8c192fe9: Merge "Better support for rtsp streamed through VLC. Temporarily make the socket blocking to read all of the session description." into gingerbread
Merge commit '6bcffcd2dc410db780c152c70a01b22da6ca58be'
* commit '6bcffcd2dc410db780c152c70a01b22da6ca58be':
Better support for rtsp streamed through VLC. Temporarily make the socket blocking to read all of the session description.
|
| 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/MyHandler.h
|
| 982a93173bc84f005172152d823cbb59dfcbeb12 |
|
05-Aug-2010 |
Andreas Huber <andih@google.com> |
am 1f513d88: am c17f35dd: Merge "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." into gingerbread
Merge commit '1f513d8821670a33d6361ea521b6756163a3f9bf'
* commit '1f513d8821670a33d6361ea521b6756163a3f9bf':
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.
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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/MyHandler.h
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| f661058d77d1484e5911d1962f8e1e8466240687 |
|
22-Jul-2010 |
Andreas Huber <andih@google.com> |
am b72d3180: am 81046c8c: Merge "Various changes to improve rtsp networking, reduce packet loss and adapt to ALooper API changes." into gingerbread
Merge commit 'b72d3180dc8d41d6269664bea808b04410bbe40f'
* commit 'b72d3180dc8d41d6269664bea808b04410bbe40f':
Various changes to improve rtsp networking, reduce packet loss and adapt to ALooper API changes.
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| 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/MyHandler.h
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| 4e2ffa400b82559cab2c5717c8dcdff393d334a9 |
|
15-Jul-2010 |
Mike Lockwood <lockwood@android.com> |
Fixes for simulator build on lucid
strchr and strrchr now return const char* instead of char*
Change-Id: I5ca831b8951af7e6306eb9d9d6f78ed2ec13d649
Signed-off-by: Mike Lockwood <lockwood@android.com>
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
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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/MyHandler.h
|