9da9b138b60c2d21afc3e66487c60346ccf75ebd |
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16-Jul-2016 |
Robert Shih <robertshih@google.com> |
Merge \\"rtsp: remove CHECK to ignore unsupported tracks\\" into nyc-mr1-dev am: b800dfd823 am: 77e39d9c06 Change-Id: I32dc5c278999a64e831641466662104e8f1b6f36
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30474b19a4f9785494d34c15ee2deabe36970814 |
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15-Jul-2016 |
Robert Shih <robertshih@google.com> |
rtsp: remove CHECK to ignore unsupported tracks RFC 3551 specifies a number of static A/V profiles for media payload. These don't use the 'rtpmap' or 'fmtp' properties in the session description, and thus need to be handled differently. For now, we just ignore these types as they are very rarely used. The previous paragraph was taken from https://android-review.googlesource.com/#/c/240630/, author David Williams <david.williams@sonymobile.com>. Bug: 29493841 Change-Id: I12581447f18344ac503fe427e3c940a85d44644e
/frameworks/av/media/libstagefright/rtsp/ASessionDescription.cpp
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e20f434b16c61d03e5313d0166104e1c1cb1faac |
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14-Jul-2016 |
Ray Essick <essick@google.com> |
Merge \\\"Fix corruption via buffer overflow in mediaserver\\\" into klp-dev am: b5299033ce am: 357e0f132e am: efa1272404 Change-Id: Ia5996291c99df58d5986195424ce98ea354e6a8b
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b2396a52aa1620a68bba4d776b136b03707f7556 |
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06-Jul-2016 |
Ray Essick <essick@google.com> |
Fix corruption via buffer overflow in mediaserver change unbound sprintf() to snprintf() so network-provided values can't overflow the buffers. Applicable to all K/L/M/N branches. Bug: 25747670 Change-Id: Id6a5120c2d08a6fbbd47deffb680ecf82015f4f6
/frameworks/av/media/libstagefright/rtsp/ASessionDescription.cpp
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2c9cb62d0160fc4353d89fd9d648725794e7995e |
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29-Feb-2016 |
George Burgess IV <gbiv@google.com> |
Cleanup uses of sprintf so we can deprecate it. Also removes a seemingly useless branch. Change-Id: Icd61d4facd64985f4c73e9876569e1ae1a74ee83
/frameworks/av/media/libstagefright/rtsp/ASessionDescription.cpp
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f4eadb67ba9130b583b8f2f192276b53fa3d50bc |
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16-Sep-2014 |
Wei Jia <wjia@google.com> |
ASessionDescription: allow open-ended NTP range. Bug: 17435211 Change-Id: I450d512abdc4368f5180d9859f3b4e207e3b5591
/frameworks/av/media/libstagefright/rtsp/ASessionDescription.cpp
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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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