| af5dd7753e62353411cf0daf3b513c38818e9662 |
02-Oct-2012 |
Andreas Huber <andih@google.com> |
ALooper::GetNowUs() now relies on systemTime instead of gettimeofday.
Change-Id: Ia31eb0940b02581327a8bf51af6df135f9ab6de3
related-to-bug: 7266324
RTPAssembler.cpp
|
| cfaeeec0900014d97e15829e0fa52f865ee4c786 |
31-Aug-2012 |
Andreas Huber <andih@google.com> |
Add support for mpeg2 transport streams to the RTSP implementation.
Change-Id: I409d7133a53a71e62523b1acc2b03302fcf824a5
MPEG2TSAssembler.cpp
MPEG2TSAssembler.h
PacketSource.cpp
RTPSource.cpp
ndroid.mk
|
| 3677437296fd1547d762b1b227a3de83dbc960d6 |
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>
RTPConnection.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
PacketSource.cpp
|
| 3d51d5cb53cc630709a0ba78d0e60501a675f2d5 |
13-Jun-2012 |
James Dong <jdong@google.com> |
Add NOTICE and MODULE_LICENSE_APACH2 to libs build under /frameworks/av/
Change-Id: I0a3af3e2abdedebd5934f3d941d01c32cfc75e26
related-to-bug: 6647465
ODULE_LICENSE_APACHE2
OTICE
|
| 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
MPEG4AudioAssembler.cpp
MPEG4ElementaryAssembler.cpp
MPEG4ElementaryAssembler.h
PacketSource.cpp
RTPAssembler.cpp
RTPAssembler.h
|
| f95439afa8eb2484969d4a928b0fdd6a4d3a38d7 |
11-Apr-2012 |
Andreas Huber <andih@google.com> |
Changes to add support for H263-1999/2000 formats for streaming
contributed by sureshc@nvidia.com (and subsequently simplified)
Change-Id: Ia1c2ac9233f5414ce3e4a70e42e68c1c5c35eb9d
H263Assembler.cpp
|
| 559bf2836f5da25b75bfb229fec0d20d540ee426 |
28-Mar-2012 |
James Dong <jdong@google.com> |
AV Android make files changes
o plus a few file relocation: ActivityManager.cpp/h, SoundPool.h, etc
o remove some runtime dependencies to libandroid, libandroid_runtime, etc
Change-Id: I047a47c5fb361dd5cf85cd98798c39f629a75d10
ndroid.mk
|
| 3ee26944b082def647fe5bb2b75116ffb0267059 |
24-Mar-2012 |
James Dong <jdong@google.com> |
Remove JNI in LOCAL_C_INCLUDE from non-JNI related Android.mk files.
o related-to-bug: 6214141
Change-Id: Ic88d1732b3e014af47532a0809e01f6086e8464d
ndroid.mk
|
| 6c6b4d0d2b98a7ceee8b697daaf611f8df3254fb |
12-Mar-2012 |
James Dong <jdong@google.com> |
Switched to use the header files in /frameworks/native
and deleted the duplicate header files in /frameworks/base
o related-to-bug: 6044887
Change-Id: I17e0692d9a9b5c8796ded36677c833ca8ab36795
ndroid.mk
|
| 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
AMRAssembler.cpp
AVCAssembler.cpp
H263Assembler.cpp
MPEG4AudioAssembler.cpp
MPEG4ElementaryAssembler.cpp
RTPConnection.cpp
RTPSession.cpp
RTSPConnection.cpp
RawAudioAssembler.cpp
yHandler.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
yHandler.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
RTPSession.cpp
RTSPConnection.cpp
yHandler.h
DPPusher.cpp
tp_test.cpp
|
| 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
RTPConnection.cpp
RTPSource.cpp
RTSPConnection.cpp
yHandler.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
MPEG4AudioAssembler.cpp
PacketSource.cpp
RTPConnection.cpp
RTPSession.cpp
RTPWriter.cpp
RTSPConnection.cpp
SessionDescription.cpp
yHandler.h
DPPusher.cpp
|
| 6af1e76b61d04ed524b570f92091680a851207df |
12-Dec-2011 |
Andreas Huber <andih@google.com> |
Merge "Support for parsing of "folded" RTSP header values and some tweaks to the AMR assembler"
|
| 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
|
| 4aae77cbe1bf4369910314a55c2bc2349af10d3c |
10-Dec-2011 |
Andreas Huber <andih@google.com> |
Support for parsing of "folded" RTSP header values and some tweaks to the AMR assembler
contributed by Samsung (untested).
Change-Id: I182561fe0a1a564126bdbb317e96aa52bf525726
AMRAssembler.cpp
RTSPConnection.cpp
|
| 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
yHandler.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
|
| a36d8caf15d56a75906e9cc75b5e04463c1317a6 |
11-Nov-2011 |
Andreas Huber <andih@google.com> |
am 9c981cd3: am d9f25bc9: Merge "Disconnect on socket error on the RTSP control connection." into ics-mr1
* commit '9c981cd3d53238f10842368c1cd82d625b701a47':
Disconnect on socket error on the RTSP control connection.
|
| 91f230461288a2a5091182ef9e17079aabf8ebaa |
11-Nov-2011 |
Andreas Huber <andih@google.com> |
Didn't mean to check this in...
Change-Id: Ie5a1902ff2613cd349ca5724f63a3fe3306640c7
yHandler.h
|
| 73b1fd56d99b356b0effe8cf96ecf7446beb207f |
11-Nov-2011 |
Andreas Huber <andih@google.com> |
Merge "Instead of asserting, signal a runtime error if the session doesn't contain" into ics-mr1
|
| 4ab3045755d33ab24bd312cfbc888f300c5e01f9 |
11-Nov-2011 |
Andreas Huber <andih@google.com> |
Merge "DO NOT MERGE: Instead of asserting, remove active streams if their sockets" into ics-mr1
|
| 0fbe0577cfeda28bd016110e670708cce0752044 |
10-Nov-2011 |
Andreas Huber <andih@google.com> |
Disconnect on socket error on the RTSP control connection.
Change-Id: Ib52a69f9b0830b481c6f5c9b1991d1f4cb36ec7b
RTSPConnection.cpp
RTSPConnection.h
|
| 19de627354d465c4e9ccd1fcdcffd132861330b2 |
09-Nov-2011 |
Andreas Huber <andih@google.com> |
DO NOT MERGE: Instead of asserting, remove active streams if their sockets
return failure
Change-Id: Icb47adfd2fbe0398c473ba66e068186311c9cc79
related-to-bug: 5593654
RTPConnection.cpp
|
| 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
yHandler.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
|
| 8c308ffd781132c8417cebc3bf77c2e56a464e0b |
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
RTPConnection.cpp
|
| 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
yHandler.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
AMRAssembler.cpp
AVCAssembler.cpp
MPEG4ElementaryAssembler.cpp
RTPConnection.cpp
RTPSource.cpp
RTPWriter.cpp
RTSPConnection.cpp
RawAudioAssembler.cpp
SessionDescription.cpp
yHandler.h
|
| 2bfdd428c56c7524d1a11979f200a1762866032d |
12-Oct-2011 |
Andreas Huber <andih@google.com> |
NuPlayer is now taking on the task of streaming over RTSP.
Change-Id: Ie204db8810807f1e7981959e34dc0149e5d9563a
PacketSource.cpp
PacketSource.h
RTSPController.cpp
ndroid.mk
yHandler.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
RTSPConnection.cpp
yHandler.h
|
| 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
PacketSource.cpp
|
| dab718bba3945332dc75e268e1e7f0fe2eb91c4a |
14-Jul-2011 |
Andreas Huber <andih@google.com> |
Remove legacy http support from stagefright, chromium is the new hotness.
Change-Id: I6725d42d38b91e6a1cbca43174870f445aeb3d99
RTSPConnection.cpp
yHandler.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
RTSPConnection.cpp
RTSPConnection.h
RTSPController.cpp
yHandler.h
|
| ac5767a96df9fae46a37ffba62611472135a0f6d |
30-Jun-2011 |
Andreas Huber <andih@google.com> |
Revert "Parse RTP-Info even for live streams."
This reverts commit d873413ff9f742f259c29d7d0b58265db6b24529.
SessionDescription.cpp
yHandler.h
|
| a6925e6149faf4a936a5b557a769d117454413d8 |
01-Jun-2011 |
Andreas Huber <andih@google.com> |
Parse RTP-Info even for live streams.
Change-Id: Ib2c39ce8d5366f5ea350e71b7a54f5f7c2b510b9
SessionDescription.cpp
yHandler.h
|
| 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
PacketSource.cpp
|
| e681b91c27439907f216cb6c88426929bc5194bf |
29-Mar-2011 |
Andreas Huber <andih@google.com> |
Add a user-agent header to our RTSP requests.
Change-Id: I02f8ff6a4a37fa59cc8c5fcfd3afb64ee11ba576
related-to-bug: 4173725
RTSPConnection.cpp
RTSPConnection.h
|
| 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
PacketSource.cpp
RTPSource.cpp
RawAudioAssembler.cpp
RawAudioAssembler.h
ndroid.mk
|
| 55e26193c885b7d5acdae9978848e6587987790f |
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...
MPEG4AudioAssembler.cpp
MPEG4AudioAssembler.h
|
| 5ef152132b477a07fa31b2ddd39f4cf7a29f68b4 |
16-Feb-2011 |
Andreas Huber <andih@google.com> |
Respond to RTSP server->client requests.
Even if it's just to tell them that we don't support any (this is optional).
Change-Id: I557865ac00d0fb65ffa69363eb1eceaabe522a1a
related-to-bug: 3353752
RTSPConnection.cpp
RTSPConnection.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
yHandler.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
MPEG4ElementaryAssembler.cpp
PacketSource.cpp
RTPConnection.cpp
RTPSource.cpp
SessionDescription.cpp
yHandler.h
|
| f1958f9442bc937e1f8c8d9175901500b944b021 |
14-Feb-2011 |
Andreas Huber <andih@google.com> |
Enable cancelling the rtsp connection process early.
Change-Id: Ie2059c54541ad8c675944d71b39c772b0f6f04c8
related-to-bug: 3452699
RTSPController.cpp
|
| 864d06670089f79bc177a51fd53de9db0e21fc99 |
10-Feb-2011 |
Andreas Huber <andih@google.com> |
Fix the build.
Change-Id: I9b777ffb260eb0f3790ae0907e4a443d33fa3f2f
ndroid.mk
|
| 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
RTPAssembler.cpp
RTPConnection.cpp
RTPConnection.h
RTPSession.cpp
RTPSource.cpp
RTPSource.h
ndroid.mk
yHandler.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
SessionDescription.cpp
SessionDescription.h
yHandler.h
|
| 9202cca86e9017cc5ce30970c92a91ab32a0835e |
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
MPEG4AudioAssembler.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
PacketSource.cpp
|
| 934ca8cb1bcffcf1781a576ca625d2f901e5f0a9 |
12-Jan-2011 |
Andreas Huber <andih@google.com> |
Fail to parse duration instead of asserting, if the server response cannot be parsed.
Change-Id: I42324468edca5ccce29486059091da8e64f36326
related-to-bug: 3338518
SessionDescription.cpp
|
| 674ebd0b4e1143e38392a4e3bb38b4679a4577bc |
19-Nov-2010 |
James Dong <jdong@google.com> |
Removed uncessary FILE structure pointer for I/O
o also move the fd owner from caller to callee in the Writers
Change-Id: I510ccfdd0fcc58f1777fea4ed1349fd251852c65
RTPWriter.cpp
|
| 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
MPEG4AudioAssembler.cpp
MPEG4AudioAssembler.h
RTPSource.cpp
SessionDescription.cpp
|
| c21143636f2c6078c8ad6b096f69a9208591342b |
25-Oct-2010 |
Andreas Huber <andih@google.com> |
We don't have access to the md5 implementation on the simulator, let's disable digest authentication in rtsp for simulator targets.
Change-Id: I02a9b4af929601c899f04cee9864d0dd0716de62
RTSPConnection.cpp
|
| 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
RTSPConnection.cpp
RTSPConnection.h
SessionDescription.cpp
ndroid.mk
yHandler.h
|
| 8ac0cb9dc8a46f9b2badabc91cb5f7871e2215a9 |
18-Oct-2010 |
Jean-Baptiste Queru <jbq@google.com> |
Merge fb474872 from gingerbread-plus-aosp
Change-Id: I1bbb845a86a7b7df44ea175df3af22e5f47c44e3
|
| 56cfa2376ae87cba730ea7ce4a9e0ca4f0d07627 |
15-Oct-2010 |
Andreas Huber <andih@google.com> |
Include the framework copy of the OpenMAX headers instead of referencing external/opencore.
Change-Id: I762f59acf5e1f770e4d7c2d89af362bfffebefa6
related-to-bug: 3101573
ndroid.mk
|
| 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
RTPConnection.cpp
RTPConnection.h
yHandler.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
yHandler.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.
|
| e31aa743466972764f9db5a88a713621ff0a29ae |
11-Oct-2010 |
Andreas Huber <andih@google.com> |
am e0c8545a: am 0fd4e216: Merge "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." into gingerbread
Merge commit 'e0c8545a2369881fe09582337a9de3db2db1a951'
* commit 'e0c8545a2369881fe09582337a9de3db2db1a951':
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.
|
| 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
RTSPController.cpp
yHandler.h
|
| 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
PacketSource.cpp
PacketSource.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
yHandler.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
AMRAssembler.cpp
AVCAssembler.cpp
MPEG4ElementaryAssembler.cpp
PacketSource.cpp
RTPConnection.cpp
RTPSession.cpp
RTPSource.cpp
RTPSource.h
RTPWriter.cpp
RTSPConnection.cpp
SessionDescription.cpp
yHandler.h
DPPusher.cpp
tp_test.cpp
|
| 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
PacketSource.cpp
RTSPConnection.cpp
yHandler.h
|
| 6faf0cd82346b23075d1f8b9f70f7af43f2c5f04 |
02-Sep-2010 |
Andreas Huber <andih@google.com> |
am fd0eed00: am a2511da9: Merge "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." into gingerbread
Merge commit 'fd0eed007d99178092ede56ec2c4799046615f70'
* commit 'fd0eed007d99178092ede56ec2c4799046615f70':
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.
|
| 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
PacketSource.cpp
PacketSource.h
RTSPController.cpp
|
| 56f2c6e529bc62d55fc8baa7d1b52326307474d4 |
01-Sep-2010 |
Andreas Huber <andih@google.com> |
am 47f2cf62: am 412fc7cd: Merge "Keep gtalk video chat specific code consistent with rtsp changes." into gingerbread
Merge commit '47f2cf620731226a9311db0f864a4e1404e54b96'
* commit '47f2cf620731226a9311db0f864a4e1404e54b96':
Keep gtalk video chat specific code consistent with rtsp changes.
|
| 389636ce967af15e72817e2133907a2cb2efd1ae |
01-Sep-2010 |
Andreas Huber <andih@google.com> |
Keep gtalk video chat specific code consistent with rtsp changes.
Change-Id: I5f3f46c2150e16b26674432e427f79c04a69cd8e
RTPSession.cpp
|
| 3ef9f98aebb76018d2ee48ae4ac727a05efa63df |
01-Sep-2010 |
Andreas Huber <andih@google.com> |
am 6b52911c: am 48ac68e1: Merge "Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr." into gingerbread
Merge commit '6b52911cc7ba548fd3a240ca61eba510a8581e6f'
* commit '6b52911cc7ba548fd3a240ca61eba510a8581e6f':
Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr.
|
| 06124758ff402512f3c7a5fb2b35d8d09a0d6c2e |
31-Aug-2010 |
Andreas Huber <andih@google.com> |
Merge "Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr." into gingerbread
|
| 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)
|
| 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
MPEG4ElementaryAssembler.cpp
MPEG4ElementaryAssembler.h
PacketSource.cpp
RTPSource.cpp
|
| 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
RTPConnection.cpp
yHandler.h
|
| ca999e0f936fc83f321e31ae13f93348d3f7454c |
31-Aug-2010 |
Andreas Huber <andih@google.com> |
am 03e83d4a: am 68ae91cb: Merge "Recent changes to the rtsp code require every buffer fed to the packet source to have a timestamp, we\'re ignoring timestamps for gtalk videochat but we still have to have a placeholder." into gingerbread
Merge commit '03e83d4ad909f5c07fb2011e03348a413453e909'
* commit '03e83d4ad909f5c07fb2011e03348a413453e909':
Recent changes to the rtsp code require every buffer fed to the packet source to have a timestamp, we're ignoring timestamps for gtalk videochat but we still have to have a placeholder.
|
| 5d5f5dfcc16756fe80a7c46cff0949fce9d54fe9 |
31-Aug-2010 |
Andreas Huber <andih@google.com> |
Merge "Recent changes to the rtsp code require every buffer fed to the packet source to have a timestamp, we're ignoring timestamps for gtalk videochat but we still have to have a placeholder." into gingerbread
|
| b186054757f4743eb9a6d6e81d262b9c7b36bec7 |
31-Aug-2010 |
Andreas Huber <andih@google.com> |
Recent changes to the rtsp code require every buffer fed to the packet source to have a timestamp, we're ignoring timestamps for gtalk videochat but we still have to have a placeholder.
Change-Id: Ib8615ce5a89a9a846ee2f9f96cdfb23462f72c7a
RTPSession.cpp
|
| 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
RTPConnection.cpp
yHandler.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
yHandler.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
RTPWriter.cpp
yHandler.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
AMRAssembler.cpp
AVCAssembler.cpp
H263Assembler.cpp
MPEG4AudioAssembler.cpp
MPEG4ElementaryAssembler.cpp
PacketSource.cpp
PacketSource.h
RTPAssembler.cpp
RTPAssembler.h
RTSPController.cpp
yHandler.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
yHandler.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
RTPConnection.cpp
RTPConnection.h
RTPSession.cpp
RTSPConnection.cpp
RTSPConnection.h
yHandler.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
PacketSource.cpp
PacketSource.h
RTSPController.cpp
yHandler.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
RTSPController.cpp
yHandler.h
|
| 263ebfd8a17266eedc84eb879edb6a6a3395f760 |
21-Aug-2010 |
James Dong <jdong@google.com> |
am c8d2fa70: am cbd038fe: Merge "Make MediaWriter stop and pause return errors if necessary" into gingerbread
Merge commit 'c8d2fa704abebdbf0bd8aac185216dc068950217'
* commit 'c8d2fa704abebdbf0bd8aac185216dc068950217':
Make MediaWriter stop and pause return errors if necessary
|
| 9934d0cf66861d331adcad28dc4713874e607a76 |
21-Aug-2010 |
Andreas Huber <andih@google.com> |
am 873ebfb8: am 223e4f73: Merge "Support for MP4V-ES packetization format according to RFC3016." into gingerbread
Merge commit '873ebfb825cb498d9ff3012d1d31b02e31a79980'
* commit '873ebfb825cb498d9ff3012d1d31b02e31a79980':
Support for MP4V-ES packetization format according to RFC3016.
|
| 9b92412737095ab6a06f01a0c6daaebb79dffb55 |
21-Aug-2010 |
Andreas Huber <andih@google.com> |
am b29ebd39: am f0ad5484: Merge "In the absence of width/height information in the sdp, extract the dimensions from the avc codec specific data." into gingerbread
Merge commit 'b29ebd397e25a7176bcc1c81980f17b0190ebe7f'
* commit 'b29ebd397e25a7176bcc1c81980f17b0190ebe7f':
In the absence of width/height information in the sdp, extract the dimensions from the avc codec specific data.
|
| 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.
|
| e0b77ce97ef84c47ae408e92f2afb7509a5051b6 |
19-Aug-2010 |
James Dong <jdong@google.com> |
Merge "Make MediaWriter stop and pause return errors if necessary" into gingerbread
|
| 37187916a486504acaf83bea30147eb5fbf46ae5 |
19-Aug-2010 |
James Dong <jdong@google.com> |
Make MediaWriter stop and pause return errors if necessary
o Make the API consistent with SF framework, which the MediaSource
provides a return status for stop
o Also, helps to convey errors that occurred right when a
premature stop() is called, leading to a potentially
mal-formed output file.
Change-Id: I52a932345f38570fdf8ea04d67d73dd94ccd30ef
RTPWriter.cpp
RTPWriter.h
|
| 62cb04d23642a2ea7c005f050494c8ef3c370dd3 |
19-Aug-2010 |
Andreas Huber <andih@google.com> |
Support for MP4V-ES packetization format according to RFC3016.
Change-Id: I5e182936c52f9eb80cdcf6132ead03705ee32d61
MPEG4ElementaryAssembler.cpp
MPEG4ElementaryAssembler.h
PacketSource.cpp
RTPConnection.cpp
RTPSource.cpp
ndroid.mk
|
| 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
PacketSource.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
PacketSource.cpp
RTSPConnection.cpp
SessionDescription.cpp
SessionDescription.h
yHandler.h
|
| cc760e477378117ef34fb2833d0b6521925b38ad |
12-Aug-2010 |
Andreas Huber <andih@google.com> |
am 3bf8c342: am ae3a1f45: Merge "Fix the h.263 assembler to properly subset a buffer\'s range if it already has a range applied." into gingerbread
Merge commit '3bf8c3427f4c728bb88e5e266b85c96e3e727203'
* commit '3bf8c3427f4c728bb88e5e266b85c96e3e727203':
Fix the h.263 assembler to properly subset a buffer's range if it already has a range applied.
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| db3a7e67a82b48b9b7e2bfa639fc117f75682a76 |
12-Aug-2010 |
Andreas Huber <andih@google.com> |
am 53895c6a: am 66aa0f3d: Merge "APacketSource is too verbose." into gingerbread
Merge commit '53895c6a0e8ecb4e835aab7eca7480779c224356'
* commit '53895c6a0e8ecb4e835aab7eca7480779c224356':
APacketSource is too verbose.
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| a6238a1e5b603ca2ccf3b2297c9bc8a141cf8559 |
12-Aug-2010 |
Andreas Huber <andih@google.com> |
Fix the h.263 assembler to properly subset a buffer's range if it already has a range applied.
Change-Id: I7cc468a3095537347d86803579001458b62fcadb
H263Assembler.cpp
RTPWriter.cpp
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| 6dc387a8c3f031f9f17d1138295368946563f7a5 |
12-Aug-2010 |
Andreas Huber <andih@google.com> |
APacketSource is too verbose.
Change-Id: I48ca7b070d89e43405d05e5f41e650db587e12b4
PacketSource.cpp
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| 5d8e9cd46d21d8cddebe82831b99927363fa896a |
10-Aug-2010 |
Andreas Huber <andih@google.com> |
am 4dc41bb4: am 18f0174f: Merge "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." into gingerbrea
Merge commit '4dc41bb445860cfcb8c0dfbecdc8f0f5f15f5e28'
* commit '4dc41bb445860cfcb8c0dfbecdc8f0f5f15f5e28':
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.
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| 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
PacketSource.cpp
RTPConnection.cpp
RTPConnection.h
RTPSession.cpp
RTPSource.cpp
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| c16acb7a9467831caf2c7c268a3fe57ec4bc69aa |
05-Aug-2010 |
Andreas Huber <andih@google.com> |
am 870678a9: am 2c37f3d3: Merge "Better support for fake timestamps in RTP, H.263 video now also requests FIR." into gingerbread
Merge commit '870678a954e1e2a96caf76453c20de808253ffd1'
* commit '870678a954e1e2a96caf76453c20de808253ffd1':
Better support for fake timestamps in RTP, H.263 video now also requests FIR.
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| b6b546e72818988865d508e380d4445da71c4503 |
05-Aug-2010 |
Andreas Huber <andih@google.com> |
am c6d1519e: am fb861523: Merge "Specification of codec specific data as part of the session description is now optional." into gingerbread
Merge commit 'c6d1519e549740abd56df7a98b5348bd9095ae46'
* commit 'c6d1519e549740abd56df7a98b5348bd9095ae46':
Specification of codec specific data as part of the session description is now optional.
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| 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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| ff53123821a3ec2e71fdb1a971ea2cbae3119826 |
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
RTPConnection.cpp
RTPSource.cpp
RTPSource.h
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| 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
PacketSource.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
AMRAssembler.cpp
AMRAssembler.h
AVCAssembler.cpp
AVCAssembler.h
H263Assembler.cpp
H263Assembler.h
MPEG4AudioAssembler.cpp
MPEG4AudioAssembler.h
PacketSource.cpp
PacketSource.h
RTPAssembler.h
RTPConnection.cpp
RTPConnection.h
RTPSession.cpp
RTPSession.h
RTPSource.cpp
RTPSource.h
RTPWriter.cpp
RTPWriter.h
SessionDescription.cpp
ndroid.mk
yHandler.h
DPPusher.cpp
DPPusher.h
tp_test.cpp
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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
MPEG4AudioAssembler.cpp
RTSPController.cpp
yHandler.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>
RTSPConnection.cpp
SessionDescription.cpp
yHandler.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
AVCAssembler.cpp
AVCAssembler.h
MPEG4AudioAssembler.cpp
MPEG4AudioAssembler.h
PacketSource.cpp
PacketSource.h
RTPAssembler.cpp
RTPAssembler.h
RTPConnection.cpp
RTPConnection.h
RTPSource.cpp
RTPSource.h
RTSPConnection.cpp
RTSPConnection.h
RTSPController.cpp
SessionDescription.cpp
SessionDescription.h
ndroid.mk
yHandler.h
yTransmitter.h
ideoSource.h
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