| dbe09da6ac4d1e9e78e5c8f33fbc6d32822ba7ed |
16-Feb-2011 |
Andreas Huber <andih@google.com> |
DO NOT MERGE: Respond to RTSP server->client requests.
Even if it's just to tell them that we don't support any (this is optional).
Change-Id: Iee50b4020f28a47dfbe5d56f1732fe044b3b3655
related-to-bug: 3353752
RTSPConnection.cpp
RTSPConnection.h
|
| ea47cb4edea4426b0da7807db10548ddae7104f2 |
15-Feb-2011 |
Andreas Huber <andih@google.com> |
DO NOT MERGE: Derive the Transport "source" attribute from the RTSP endpoint address if necessary
and continue even if we were unable to poke a hole into the firewall.
Change-Id: I5757a2521b8d81a42d03cca379179ce2c9ee46e7
related-to-bug: 3457201
yHandler.h
|
| d1ba051a465518fa4325c364ed77025fc1a2a794 |
15-Feb-2011 |
Andreas Huber <andih@google.com> |
DO NOT MERGE: 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: I502a04a7e1d690fd461b7ecf0b56c6a6c2ac1325
related-to-bug: 3452103
MPEG4ElementaryAssembler.cpp
PacketSource.cpp
RTPConnection.cpp
RTPSource.cpp
SessionDescription.cpp
yHandler.h
|
| b6c2e2b46c574a90450438cccfb6cb97b7edc20f |
28-Jan-2011 |
Andreas Huber <andih@google.com> |
DO NOT MERGE: More robust parsing of NPT time ranges in RTSP.
Change-Id: If5a00f1e29dbc12956e1fb000dac859706d19791
related-to-bug: 3217210
SessionDescription.cpp
SessionDescription.h
yHandler.h
|
| 2dce338e01678620db0734fc3d84bcb3f2512d62 |
27-Jan-2011 |
Andreas Huber <andih@google.com> |
DO NOT MERGE: This particular RTSP server streams MPEG4-LATM audio with extra trailing bytes.
And now we're just ignoring them. Yay standards.
Change-Id: Ia8c0b9161e606152fb681f0dda3ba901954dc749
related-to-bug: 3353752
MPEG4AudioAssembler.cpp
|
| 7c6153606cc963191362494c8cb5669749e84326 |
12-Jan-2011 |
Andreas Huber <andih@google.com> |
Fail to parse duration instead of asserting, if the server response cannot be parsed.
Change-Id: I95c61ed83800db82e99c0023b942fb8ae05ed3cf
related-to-bug: 3338518
SessionDescription.cpp
|
| 549f12ad04c491a2f25f599794868a4e21e9f1eb |
11-Jan-2011 |
Andreas Huber <andih@google.com> |
DO NOT MERGE: Fix parsing of ntp= PLAY response.
related-to-bug: 3340186
Squashed commit of the following:
commit b61c36b7228aec9f5360883b1e1c1e0530488974
Author: Andreas Huber <andih@google.com>
Date: Wed Oct 27 13:59:59 2010 -0700
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.
commit b10f322c07e5bebcaf032e8624cb4a5d733dfc15
Author: Andreas Huber <andih@google.com>
Date: Mon Oct 25 09:40:52 2010 -0700
We don't have access to the md5 implementation on the simulator, let's disable digest authentication in rtsp for simulator targets.
commit 0aa83cf9e4637adf9501708fcdf7d0d6d4dc4fe1
Author: Andreas Huber <andih@google.com>
Date: Wed Oct 20 15:00:34 2010 -0700
Support for BASIC and DIGEST authentication schemes in RTSP. Support for malformed packet descriptions that end lines in LF only, instead of CRLF.
related-to-bug: 3084183
Change-Id: I6e512cb73cc8d5624a83f7154aa5699f7fef7534
MPEG4AudioAssembler.cpp
MPEG4AudioAssembler.h
RTPSource.cpp
RTSPConnection.cpp
RTSPConnection.h
SessionDescription.cpp
ndroid.mk
yHandler.h
|
| a4f391c9bf581af19d6dec4603c194126995b6bf |
15-Oct-2010 |
Andreas Huber <andih@google.com> |
Include the framework copy of the OpenMAX headers instead of referencing external/opencore.
Change-Id: I762f59acf5e1f770e4d7c2d89af362bfffebefa6
related-to-bug: 3101573
ndroid.mk
|
| cc5fb1d5e5c1971cabfc2cba89de63ba65678882 |
13-Oct-2010 |
Andreas Huber <andih@google.com> |
Some webcams output rtp streams but never send any rtcp data in violation of
the specs. Attempt to use fake timestamps to be able to play these...
Change-Id: Ia7a926616fb764e972955df4acdb59d85cdd93df
related-to-bug: 3087310
RTPConnection.cpp
RTPConnection.h
yHandler.h
|
| 0dc6403f3c660f6e6f1840276e3240365889103d |
11-Oct-2010 |
Andreas Huber <andih@google.com> |
Just in case we're behind a NAT router/firewall, attempt to poke holes into it for future incoming RTP/RTCP packets to pass through.
Change-Id: Idd47968d4027f357222f19f15eecfd70fdec3282
related-to-bug: 3073813
yHandler.h
|
| 0c46b69f612da61ed39b32823d2d6baf2e8215e9 |
09-Oct-2010 |
Andreas Huber <andih@google.com> |
RTSP seeking is now asynchronous, MediaPlayer is not notified that the seek is complete until it actually is. Ignore seek requests on live streams.
Change-Id: Ie61230cd60dd6c682baf72529100369ad6291189
related-to-bug: 3073955
RTSPController.cpp
yHandler.h
|
| 38285db197ba11ee396873713e504fdc3e836725 |
08-Oct-2010 |
Andreas Huber <andih@google.com> |
Refactor some more h.264 utility code out into avc_utils. Work around a hardware decoder issue by making sure the first access unit submitted to a decoder at startup or after seek is an IDR.
Change-Id: I61936601e55df7e4c23a8c13087579a4f85bd6e6
PacketSource.cpp
PacketSource.h
|
| e51e80990e72dee6372e3300fbbcdac3a115b60a |
08-Oct-2010 |
Andreas Huber <andih@google.com> |
Disable the access unit timeout temporarily while a seek operation is in progress.
Change-Id: I116cb76342aae4168f34ebae49ecb2301702a0ea
related-to-bug: 3073955
yHandler.h
|
| 6e3fa444c5b3970666707bb2b6d25e2615dafe80 |
21-Sep-2010 |
Andreas Huber <andih@google.com> |
Remove stagefright foundation's incompatible logging interface and update callsites.
Change-Id: I45fba7d60530ea0f233ac3695a97306b6dc1795c
AMRAssembler.cpp
AVCAssembler.cpp
MPEG4ElementaryAssembler.cpp
PacketSource.cpp
RTPConnection.cpp
RTPSession.cpp
RTPSource.cpp
RTPSource.h
RTPWriter.cpp
RTSPConnection.cpp
SessionDescription.cpp
yHandler.h
DPPusher.cpp
tp_test.cpp
|
| f3d2bdf73c36be549f1ddff4238e97b3629c480d |
15-Sep-2010 |
Andreas Huber <andih@google.com> |
Various fixes to improve resilience of the rtsp stack against spurious errors instead of asserting.
Change-Id: Idbec5996ed0675c70e911b9c0514961fea099fb4
PacketSource.cpp
RTSPConnection.cpp
yHandler.h
|
| 4d8f66bce32fbc8700b4ae5b2f6673a9cf1d20ad |
02-Sep-2010 |
Andreas Huber <andih@google.com> |
Better support for buffered streaming of rtsp content, if buffer drops below a certain threshold we will temporarily pause playback until we have sufficient data.
Change-Id: Ice8564e902e48c89c9c00f6651c5504b3c41fcad
related-to-bug: 2556656
PacketSource.cpp
PacketSource.h
RTSPController.cpp
|
| 27b9c8ec168f0b26a663960c6ee6fb973265d195 |
01-Sep-2010 |
Andreas Huber <andih@google.com> |
Keep gtalk video chat specific code consistent with rtsp changes.
Change-Id: I5f3f46c2150e16b26674432e427f79c04a69cd8e
RTPSession.cpp
|
| 48ac68e1b117b6b55f06daced7d9d5d550853306 |
31-Aug-2010 |
Andreas Huber <andih@google.com> |
Merge "Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr." into gingerbread
|
| 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
MPEG4ElementaryAssembler.cpp
MPEG4ElementaryAssembler.h
PacketSource.cpp
RTPSource.cpp
|
| 3a48d4d7269a37308eee4affd021adfcab7629a1 |
31-Aug-2010 |
Andreas Huber <andih@google.com> |
Better detection of connection problems - timeout if no rtcp packets arrive within a certain time, not a final frame (which may take longer)
Change-Id: I3c1ae79bb9342770e959ebdcdc6b748549b76330
related-to-bug: 2556656
RTPConnection.cpp
yHandler.h
|
| 68ae91cbd20939e48ad15c15405048e7ff9fe2f8 |
31-Aug-2010 |
Andreas Huber <andih@google.com> |
Merge "Recent changes to the rtsp code require every buffer fed to the packet source to have a timestamp, we're ignoring timestamps for gtalk videochat but we still have to have a placeholder." into gingerbread
|
| 0ddf8c09f9610bf0a810c7852681738741802cb9 |
31-Aug-2010 |
Andreas Huber <andih@google.com> |
Recent changes to the rtsp code require every buffer fed to the packet source to have a timestamp, we're ignoring timestamps for gtalk videochat but we still have to have a placeholder.
Change-Id: Ib8615ce5a89a9a846ee2f9f96cdfb23462f72c7a
RTPSession.cpp
|
| f88ca7a0335c36732a5550c58c073e549c3cb0dd |
31-Aug-2010 |
Andreas Huber <andih@google.com> |
Instead of closing the connection altogether if no UDP packets arrive after a certain time, try changing transports (to interleaved TCP). Also properly close the sockets on disconnection.
Change-Id: Ie8d6a3865a0477e28d4b76bb9038e468451287b1
related-to-bug: 2556656
RTPConnection.cpp
yHandler.h
|
| f6639c46e83a1ccab7b293192c208091d17c61be |
30-Aug-2010 |
Andreas Huber <andih@google.com> |
Finetune some rtsp timeout constants.
Change-Id: Ice731c5097c2a2dee8a7f0cd45b547cd34f532c6
yHandler.h
|
| c4e0b70a21fadb47d70955c71fc31ce1473da925 |
28-Aug-2010 |
Andreas Huber <andih@google.com> |
ALoopers can now be named (useful to distinguish threads).
Change-Id: Ieabaddb2e3a9e3a7a5bc36e55cd0721b60dbd50e
RTPWriter.cpp
yHandler.h
|
| eeb97d91b97f1fc0b26815f098515e9c06d219b8 |
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
AMRAssembler.cpp
AVCAssembler.cpp
H263Assembler.cpp
MPEG4AudioAssembler.cpp
MPEG4ElementaryAssembler.cpp
PacketSource.cpp
PacketSource.h
RTPAssembler.cpp
RTPAssembler.h
RTSPController.cpp
yHandler.h
|
| d6a40047412d0269f79e6c992115642f0c65ea96 |
27-Aug-2010 |
Andreas Huber <andih@google.com> |
We accidentally always aborted after 10 secs, even if the connection was fine.
Change-Id: I3f2ae2f46ae62b84b1e253658d7182c04ee3dfae
yHandler.h
|
| 0416da73a0addfc7b3eddfea4a6a0a0215e1dd0b |
26-Aug-2010 |
Andreas Huber <andih@google.com> |
Support for RTP packets arriving interleaved with RTSP responses.
Change-Id: Ib32fba257da32a199134cf8943117cf3eaa07a25
RTPConnection.cpp
RTPConnection.h
RTPSession.cpp
RTSPConnection.cpp
RTSPConnection.h
yHandler.h
|
| e0dd7d396051942ccce0429d7a1fe968d63ac3f7 |
24-Aug-2010 |
Andreas Huber <andih@google.com> |
A first shot at proper support for seeking of rtsp streams.
Change-Id: I9604f2d09feedc0074c0e715be58e719d4483760
related-to-bug: 2556656
PacketSource.cpp
PacketSource.h
RTSPController.cpp
yHandler.h
|
| 8370be11debc574b4a9fee62009009d999e29fa3 |
23-Aug-2010 |
Andreas Huber <andih@google.com> |
Better handling of rtsp connection and disconnection.
Change-Id: Ib126af6c14c5a212a51a5ee3c4a0a7d1860ad167
RTSPController.cpp
yHandler.h
|
| cbd038fe207f183bc7e0a610973473f7c2e9d118 |
19-Aug-2010 |
James Dong <jdong@google.com> |
Merge "Make MediaWriter stop and pause return errors if necessary" into gingerbread
|
| d036662470ceb6b20b0591b7d4123f2db911536d |
19-Aug-2010 |
James Dong <jdong@google.com> |
Make MediaWriter stop and pause return errors if necessary
o Make the API consistent with SF framework, which the MediaSource
provides a return status for stop
o Also, helps to convey errors that occurred right when a
premature stop() is called, leading to a potentially
mal-formed output file.
Change-Id: I52a932345f38570fdf8ea04d67d73dd94ccd30ef
RTPWriter.cpp
RTPWriter.h
|
| a979ad6739d573b3823b0fe7321f554ef5544753 |
19-Aug-2010 |
Andreas Huber <andih@google.com> |
Support for MP4V-ES packetization format according to RFC3016.
Change-Id: I5e182936c52f9eb80cdcf6132ead03705ee32d61
MPEG4ElementaryAssembler.cpp
MPEG4ElementaryAssembler.h
PacketSource.cpp
RTPConnection.cpp
RTPSource.cpp
ndroid.mk
|
| eef3c33e5604ae6304364b7aa6337616e2d4c61e |
19-Aug-2010 |
Andreas Huber <andih@google.com> |
In the absence of width/height information in the sdp, extract the dimensions from the avc codec specific data.
Change-Id: I98c4194593c7e6e24f6fc339c862245111800293
PacketSource.cpp
|
| af063a67b291c4622321a35af6966b8568d5a564 |
18-Aug-2010 |
Andreas Huber <andih@google.com> |
Better support for rtsp streamed through VLC. Temporarily make the socket blocking to read all of the session description.
Change-Id: Ibe71f5941485660510e24d714da3865b9c6f89a2
PacketSource.cpp
RTSPConnection.cpp
SessionDescription.cpp
SessionDescription.h
yHandler.h
|
| 00237b79a031e95073f7f9ee8f7c022e149a4f3b |
12-Aug-2010 |
Andreas Huber <andih@google.com> |
Fix the h.263 assembler to properly subset a buffer's range if it already has a range applied.
Change-Id: I7cc468a3095537347d86803579001458b62fcadb
H263Assembler.cpp
RTPWriter.cpp
|
| 3f55576e049b7244103f10d03c626c70a195db2d |
12-Aug-2010 |
Andreas Huber <andih@google.com> |
APacketSource is too verbose.
Change-Id: I48ca7b070d89e43405d05e5f41e650db587e12b4
PacketSource.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
PacketSource.cpp
RTPConnection.cpp
RTPConnection.h
RTPSession.cpp
RTPSource.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
RTPConnection.cpp
RTPSource.cpp
RTPSource.h
|
| 426b650e1bf18b6fd0df67af323318a9611053f5 |
04-Aug-2010 |
Andreas Huber <andih@google.com> |
Specification of codec specific data as part of the session description is now optional.
Change-Id: Ie1953909e1d241381add3cc82a7a1f7d7d1540f2
PacketSource.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
AMRAssembler.cpp
AMRAssembler.h
AVCAssembler.cpp
AVCAssembler.h
H263Assembler.cpp
H263Assembler.h
MPEG4AudioAssembler.cpp
MPEG4AudioAssembler.h
PacketSource.cpp
PacketSource.h
RTPAssembler.h
RTPConnection.cpp
RTPConnection.h
RTPSession.cpp
RTPSession.h
RTPSource.cpp
RTPSource.h
RTPWriter.cpp
RTPWriter.h
SessionDescription.cpp
ndroid.mk
yHandler.h
DPPusher.cpp
DPPusher.h
tp_test.cpp
|
| 4e4173b0af52bdf2b5730a5837476e400c5b2040 |
22-Jul-2010 |
Andreas Huber <andih@google.com> |
Various changes to improve rtsp networking, reduce packet loss and adapt to ALooper API changes.
Change-Id: I110e19d5ce33e597add3ffbd3e3ff3815862396d
MPEG4AudioAssembler.cpp
RTSPController.cpp
yHandler.h
|
| 7a747b8e0dadf909ea4ac0b67fd88fc14b4eb3f8 |
08-Jun-2010 |
Andreas Huber <andih@google.com> |
Initial checkin of preliminary rtsp support for stagefright.
Change-Id: I0722aa888098c0c1361c97a4c1b123d910afc207
AVCAssembler.cpp
AVCAssembler.h
MPEG4AudioAssembler.cpp
MPEG4AudioAssembler.h
PacketSource.cpp
PacketSource.h
RTPAssembler.cpp
RTPAssembler.h
RTPConnection.cpp
RTPConnection.h
RTPSource.cpp
RTPSource.h
RTSPConnection.cpp
RTSPConnection.h
RTSPController.cpp
SessionDescription.cpp
SessionDescription.h
ndroid.mk
yHandler.h
yTransmitter.h
ideoSource.h
|