df64d15042bbd5e0e4933ac49bf3c177dd94752c |
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04-Jan-2012 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGI(_IF) to (IF_)ALOGI(_IF) DO NOT MERGE See https://android-git.corp.google.com/g/156801 Bug: 5449033 Change-Id: Ib08fe86d23db91ee153e9f91a99a35c42b9208ea
/frameworks/av/media/libstagefright/rtsp/ASessionDescription.cpp
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3856b090cd04ba5dd4a59a12430ed724d5995909 |
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20-Oct-2011 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGV(_IF) to (IF_)ALOGV(_IF) DO NOT MERGE See https://android-git.corp.google.com/g/#/c/143865 Bug: 5449033 Change-Id: I0122812ed6ff6f5b59fe4a43ab8bff0577adde0a
/frameworks/av/media/libstagefright/rtsp/ASessionDescription.cpp
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ac5767a96df9fae46a37ffba62611472135a0f6d |
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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/ASessionDescription.cpp
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a6925e6149faf4a936a5b557a769d117454413d8 |
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01-Jun-2011 |
Andreas Huber <andih@google.com> |
Parse RTP-Info even for live streams. Change-Id: Ib2c39ce8d5366f5ea350e71b7a54f5f7c2b510b9
/frameworks/av/media/libstagefright/rtsp/ASessionDescription.cpp
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dc468c5f9d72ce54de0070493e9a23efb8907e06 |
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15-Feb-2011 |
Andreas Huber <andih@google.com> |
Work around several issues with non-compliant RTSP servers. In this particular case these RTSP servers were implemented as local services, retransmitting live streams via a local RTSP server instance. They picked wrong rtp/rtcp port pairs (odd rtp port), blank lines in the session description, wrong case of the format description, relative base URLs... Change-Id: I63fa90ca2398f19e8b52c147248bd2c5c2372426 related-to-bug: 3452103
/frameworks/av/media/libstagefright/rtsp/ASessionDescription.cpp
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783e5cd85d4bd40b1a04dfdfed256c5dcb2525cc |
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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/ASessionDescription.cpp
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934ca8cb1bcffcf1781a576ca625d2f901e5f0a9 |
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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
/frameworks/av/media/libstagefright/rtsp/ASessionDescription.cpp
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fc9ac988e08a8b4c42e58999300265989f26f24c |
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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
/frameworks/av/media/libstagefright/rtsp/ASessionDescription.cpp
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4579b7d49f6dd4f37e6043e59debfd72d69b8e7b |
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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/ASessionDescription.cpp
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3f94dacbd43b48bb629a79e45e738ead37c5debd |
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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.
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6e4c5c499999c04c2477b987f9e64f3ff2bf1a06 |
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21-Sep-2010 |
Andreas Huber <andih@google.com> |
Remove stagefright foundation's incompatible logging interface and update callsites. Change-Id: I45fba7d60530ea0f233ac3695a97306b6dc1795c
/frameworks/av/media/libstagefright/rtsp/ASessionDescription.cpp
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91d113e8daa9d71c4ea8afd595a3921e03787cbf |
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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.
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ef7af7fec702db2fde72b16dedf9064585e6db77 |
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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/ASessionDescription.cpp
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982a93173bc84f005172152d823cbb59dfcbeb12 |
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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 |
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04-Aug-2010 |
Andreas Huber <andih@google.com> |
Support for Gtalk video, includes AMR/H.263 assembler and packetization support, extensions to MediaRecorder to stream via RTP over a pair of UDP sockets as well as various fixes to the RTP implementation. Change-Id: I95b8dd487061add9bade15749e563b01cd99d9a6
/frameworks/av/media/libstagefright/rtsp/ASessionDescription.cpp
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4e2ffa400b82559cab2c5717c8dcdff393d334a9 |
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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/ASessionDescription.cpp
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cf7b9c7aae758ac0b99833915053c63c2ac46e09 |
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08-Jun-2010 |
Andreas Huber <andih@google.com> |
Initial checkin of preliminary rtsp support for stagefright. Change-Id: I0722aa888098c0c1361c97a4c1b123d910afc207
/frameworks/av/media/libstagefright/rtsp/ASessionDescription.cpp
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