8d237a5ce1e3c1dbc1d538f47e68cff2cc52d799 |
|
14-Jul-2015 |
Robert Shih <robertshih@google.com> |
RTSP: clear data/eos status before returning from seek The original RTSP seek implementation involves pausing and restarting a session. This change clears data/eos status after an rtsp session is paused for a seek, and delays the seek to return after data/eos status are cleared. Bug: 22207372 Change-Id: I1bdf65653f90436f7ee5d7fe85eeadc1598a0d56
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
48910f120c59bfcbbe298fdd4a72c6e72e8945e9 |
|
15-May-2015 |
Wei Jia <wjia@google.com> |
RTSPSource: Do not update time when there are no tracks, i.e., when aborted. Bug: 17474566 Change-Id: I0dbd7a6a54edaf5b4fe5bd324d38f791a346b2fd
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
5efccd8da813133060c089c647b07434116406cb |
|
05-May-2015 |
Chong Zhang <chz@google.com> |
RTSP: append track URL to base URL bug: 17310253 Change-Id: I6ce8c4740a3509d82323ccc05f82cb842368caee
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
6d339f1f764bbd32e3381dae7bfa7c6c575bb493 |
|
18-Apr-2015 |
Lajos Molnar <lajos@google.com> |
libmediaplayerservice: fix warnings, make warnings errors, use clang Change-Id: I1b2f6b65c5abbc366068a60b8909104f31b94228
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
1d15ab58bf8239069ef343de6cb21aabf3ef7d78 |
|
05-Mar-2015 |
Lajos Molnar <lajos@google.com> |
media: switch to new AMessage handling Bug: 19607784 Change-Id: I94cddcb81f671422ad4982a23dc4acfe57a9f1aa
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
a1e8944a21e5833b7aadc451776f11797f5f9273 |
|
04-Feb-2015 |
Elliott Hughes <enh@google.com> |
Move AString's StringPrintf out of the way. We should come back and replace AString with std::string and switch to the "real" StringPrintf family, but this fixes the ODR violation that was preventing us from booting. Bug: 19265750 Change-Id: I798eb9ca5dd634e44625af5e583439ef9f0cdc35
/frameworks/av/media/libstagefright/rtsp/MyHandler.h
|
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
|
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
|
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.
|
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
|
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.
|
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
|
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
|
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
|