History log of /frameworks/av/media/libstagefright/rtsp/AAMRAssembler.cpp
Revision Date Author Comments (<<< Hide modified files) (Show modified files >>>)
a1ca351f98e2e9c3d03654fb9794a7bf7d8f9617 21-Dec-2010 Roger1 Jonsson <roger1.jonsson@sonyericsson.com> Fix bad checks that causes crash when streaming H.263 content.

Remove checks that causes crash for rtsp streamed h.263 content
with certain values in the RTP payload header:
Remove zero check for the five reserved bits in the payload header.
According to RFC 4629 these bits MUST be ignored by receivers.
Remove zero-check for the VRC (Video Redundancy Coding) bit,
skip packet instead.
Remove zero-check for the PLEN bits (extra picture header),
skip packet instead.
Remove zero-check for the PEBIT bits (extra picture header),
skip packet instead.
Remove corresponding zero check for the four resreved bits in the
AMR payload header. According to RFC 4867 these bits MUST be
ignored by receivers.

Change-Id: I7fc21d69a19d23da24f9267623c338d415ef1387
2d8bedd05437b6fccdbc6bf70f673ffd86744d59 21-Feb-2012 Andreas Huber <andih@google.com> Add new APIs AMessage::(set|find)Buffer to make it safer to pass

ABuffer objects through messages.

Change-Id: I9f8b4e4c4767d0d70a0105e0c0813b754379b49d
4aae77cbe1bf4369910314a55c2bc2349af10d3c 10-Dec-2011 Andreas Huber <andih@google.com> Support for parsing of "folded" RTSP header values and some tweaks to the AMR assembler

contributed by Samsung (untested).

Change-Id: I182561fe0a1a564126bdbb317e96aa52bf525726
3856b090cd04ba5dd4a59a12430ed724d5995909 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
6e4c5c499999c04c2477b987f9e64f3ff2bf1a06 21-Sep-2010 Andreas Huber <andih@google.com> Remove stagefright foundation's incompatible logging interface and update callsites.

Change-Id: I45fba7d60530ea0f233ac3695a97306b6dc1795c
8d342970108926c4ea355c90d26a2a353ec0fd47 27-Aug-2010 Andreas Huber <andih@google.com> Better support for rtsp (normal play-)time display. Better seek support, timeout if no packets arrive for too long.

Change-Id: Id491541a6ae501604cda815f8e961a3bfe26db7d
related-to-bug: 2556656
39ddf8e0f18766f7ba1e3246b774aa6ebd93eea8 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