frameworks_av.git - frameworks/av

	Commit message (Collapse)	Author	Age	Files	Lines
*	Add support for mpeg2 transport streams to the RTSP implementation.	Andreas Huber	2012-08-31	1	-0/+3
\| \| \| \|	Change-Id: I409d7133a53a71e62523b1acc2b03302fcf824a5
*	Rename (IF_)LOGW(_IF) to (IF_)ALOGW(_IF) DO NOT MERGE	Steve Block	2012-01-06	1	-3/+3
\| \| \| \| \| \| \|	See https://android-git.corp.google.com/g/157065 Bug: 5449033 Change-Id: I00a4b904f9449e6f93b7fd35eac28640d7929e69
*	Rename (IF_)LOGV(_IF) to (IF_)ALOGV(_IF) DO NOT MERGE	Steve Block	2011-10-26	1	-1/+1
\| \| \| \| \| \| \|	See https://android-git.corp.google.com/g/#/c/143865 Bug: 5449033 Change-Id: I0122812ed6ff6f5b59fe4a43ab8bff0577adde0a
*	Support for PCMA and PCMU raw audio data in RTP/RTSP.	Andreas Huber	2011-02-22	1	-0/+3
\| \| \| \| \|	Change-Id: Icb87bdfa7cf572c572e2a86c46fa072d9fad18f6 related-to-bug: 3084183
*	Work around several issues with non-compliant RTSP servers.	Andreas Huber	2011-02-15	1	-1/+1
\| \| \| \| \| \| \| \| \| \| \|	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
*	Change timestamp handling in RTSP, remove unused, experimental, gtalk support	Andreas Huber	2011-02-10	1	-43/+8
\| \| \| \| \| \| \| \|	related-to-bug: 3216447 NTP timestamp handling is now done at a higher layer than before. Change-Id: I9fb23f1335110ec59e534f9aa0fe6f6a6406dd52
*	Better support for MP4A-LATM RTP disassembly. This used to fail if ↵	Andreas Huber	2010-10-27	1	-1/+1
\| \| \| \| \| \|	mNumSubFrames > 1 and the sub frames did not align with RTP packet boundaries. Change-Id: I20e3b86f52b7f0f41663ffe8bc1f4db92280e884
*	Remove stagefright foundation's incompatible logging interface and update ↵	Andreas Huber	2010-09-21	1	-94/+9
\| \| \| \| \| \|	callsites. Change-Id: I45fba7d60530ea0f233ac3695a97306b6dc1795c
*	Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr.	Andreas Huber	2010-08-31	1	-2/+3
\| \| \| \| \|	Change-Id: Ied92ea8c2448a2cb1a732c72c21c69da1913dbc8 related-to-bug: 2556656
*	Support for MP4V-ES packetization format according to RFC3016.	Andreas Huber	2010-08-19	1	-0/+4
\| \| \| \|	Change-Id: I5e182936c52f9eb80cdcf6132ead03705ee32d61
*	We're now going to ignore timestamps completely in gtalk video conferencing, ↵	Andreas Huber	2010-08-10	1	-8/+1
\| \| \| \| \| \|	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
*	Better support for fake timestamps in RTP, H.263 video now also requests FIR.	Andreas Huber	2010-08-05	1	-0/+1
\| \| \| \|	Change-Id: I2385461887197fe4062d329086e0204f6d6620fc
*	Support for Gtalk video, includes AMR/H.263 assembler and packetization ↵	Andreas Huber	2010-08-04	1	-6/+149
\| \| \| \| \| \|	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
*	Initial checkin of preliminary rtsp support for stagefright.	Andreas Huber	2010-06-07	1	-0/+250
	Change-Id: I0722aa888098c0c1361c97a4c1b123d910afc207