History log of /frameworks/base/media/libstagefright/rtsp/ARTPSource.cpp
Revision Date Author Comments (<<< Hide modified files) (Show modified files >>>)
b6723735cf1fef04f8af0b4a5a8cb19a96c3c26d 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/base/media/libstagefright/rtsp/ARTPSource.cpp
0407269ae35e62a6aa2f6e40964970db1bd4b14a 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/base/media/libstagefright/rtsp/ARTPSource.cpp
b2934b16eac8d8a866c37a7d1d7e03635f475b08 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/base/media/libstagefright/rtsp/ARTPSource.cpp
b0d25a00fe28d3153d4c56b24d8e2792230d68be 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/base/media/libstagefright/rtsp/ARTPSource.cpp
6e3fa444c5b3970666707bb2b6d25e2615dafe80 21-Sep-2010 Andreas Huber <andih@google.com> Remove stagefright foundation's incompatible logging interface and update callsites.

Change-Id: I45fba7d60530ea0f233ac3695a97306b6dc1795c
/frameworks/base/media/libstagefright/rtsp/ARTPSource.cpp
e536f800c695bcd2ef861b9b9877b2108ed21613 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/base/media/libstagefright/rtsp/ARTPSource.cpp
a979ad6739d573b3823b0fe7321f554ef5544753 19-Aug-2010 Andreas Huber <andih@google.com> Support for MP4V-ES packetization format according to RFC3016.

Change-Id: I5e182936c52f9eb80cdcf6132ead03705ee32d61
/frameworks/base/media/libstagefright/rtsp/ARTPSource.cpp
f88f84414ae7baead03497f1d650ad8ea2f87688 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/base/media/libstagefright/rtsp/ARTPSource.cpp
3eaa3006a8230bd607375bedd79b2e328b0fc6b7 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/base/media/libstagefright/rtsp/ARTPSource.cpp
57648e4eec7dd2593af467877bc7cce4aa654759 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/base/media/libstagefright/rtsp/ARTPSource.cpp
7a747b8e0dadf909ea4ac0b67fd88fc14b4eb3f8 08-Jun-2010 Andreas Huber <andih@google.com> Initial checkin of preliminary rtsp support for stagefright.

Change-Id: I0722aa888098c0c1361c97a4c1b123d910afc207
/frameworks/base/media/libstagefright/rtsp/ARTPSource.cpp