| df9e6aaf716279baf0e27b99acf10005924245ed |
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29-Dec-2015 |
Robert Shih <robertshih@google.com> |
ARTPSource: avoid over/underflow in seq # recovery
Bug: 25801317
Change-Id: Id9a5ad2d6c27f64b502c78f06174b29edb486134
/frameworks/av/media/libstagefright/rtsp/ARTPSource.cpp
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| cfaeeec0900014d97e15829e0fa52f865ee4c786 |
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31-Aug-2012 |
Andreas Huber <andih@google.com> |
Add support for mpeg2 transport streams to the RTSP implementation.
Change-Id: I409d7133a53a71e62523b1acc2b03302fcf824a5
/frameworks/av/media/libstagefright/rtsp/ARTPSource.cpp
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| 5ff1dd576bb93c45b44088a51544a18fc43ebf58 |
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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/ARTPSource.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/ARTPSource.cpp
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| fcea8f7a7d178e5426aa06586cff54726e18d1f6 |
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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
/frameworks/av/media/libstagefright/rtsp/ARTPSource.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/ARTPSource.cpp
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| 100a4408968b90e314526185d572c72ea4cc784a |
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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/ARTPSource.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/ARTPSource.cpp
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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/ARTPSource.cpp
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| 4dba3e90f211eb5f5af19b10c5d3fc8c967b0086 |
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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
/frameworks/av/media/libstagefright/rtsp/ARTPSource.cpp
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| 62cb04d23642a2ea7c005f050494c8ef3c370dd3 |
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19-Aug-2010 |
Andreas Huber <andih@google.com> |
Support for MP4V-ES packetization format according to RFC3016.
Change-Id: I5e182936c52f9eb80cdcf6132ead03705ee32d61
/frameworks/av/media/libstagefright/rtsp/ARTPSource.cpp
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| f8ca90452ff3e252f20de38f1c3eee524c808c3e |
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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
/frameworks/av/media/libstagefright/rtsp/ARTPSource.cpp
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| ff53123821a3ec2e71fdb1a971ea2cbae3119826 |
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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
/frameworks/av/media/libstagefright/rtsp/ARTPSource.cpp
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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/ARTPSource.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/ARTPSource.cpp
|