27b70ced96371d261d2c04b583c6d68d55637301 |
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05-Feb-2016 |
Lajos Molnar <lajos@google.com> |
stagefright/rtsp: Remove dependence on OMXSource Use MediaCodecSource and SimpleDecodingSource instead. Bug: 17108024 Change-Id: Idff221fc7131b1622af97bc3c5aa952afcd3d22b (cherry picked from commit a2b4bcf6562de3f8528fc139ec202bd73fa340c7)
/frameworks/av/media/libstagefright/rtsp/rtp_test.cpp
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29357bc2c0dd7c43ad3bd0c8e3efa4e6fd9bfd47 |
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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/rtp_test.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/rtp_test.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/rtp_test.cpp
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