| 820c4893fdec784321826fd903da34fe3d609b93 |
23-Sep-2014 |
Wei Jia <wjia@google.com> |
MyHandler: set ip address to an invalid one when getsockname() returns error.
Bug: 17556472
Change-Id: I0387c78727d9a18abddcfdb4b480f4b1412bbc9f
yHandler.h
|
| f4eadb67ba9130b583b8f2f192276b53fa3d50bc |
16-Sep-2014 |
Wei Jia <wjia@google.com> |
ASessionDescription: allow open-ended NTP range.
Bug: 17435211
Change-Id: I450d512abdc4368f5180d9859f3b4e207e3b5591
SessionDescription.cpp
|
| 80804f4e953d6c5f6ed0c3c8e004c4cce280f5c1 |
20-Aug-2014 |
Chong Zhang <chz@google.com> |
print warning if offset != buffer size
Bug: 17110981
Change-Id: Iacceca203372f4c06ff5ef7ce98edd5554727b64
MPEG4ElementaryAssembler.cpp
|
| dc9aa7e2cb903bb4ebfce558671a97088477bb6e |
20-Aug-2014 |
Chong Zhang <chz@google.com> |
Don't crash for bitstream errors in AMPEG4ElementaryAssembler
Bug: 17110981
Change-Id: I0d0960fa12f2ad179231494be29af307de217b2a
MPEG4ElementaryAssembler.cpp
|
| b9e55c4f17a91f070f78fb9fd72c08e461526e9e |
11-Jun-2014 |
Christopher Ferris <cferris@google.com> |
am ca44dc79: am 8d6d8f54: Merge "Add libcrypto for users of libstagefright."
* commit 'ca44dc79b5a163030ab0963f80aa771871de092d':
Add libcrypto for users of libstagefright.
|
| 7dc5bfcf42cfb59025f615f494e29ff9e55990cc |
11-Jun-2014 |
Christopher Ferris <cferris@google.com> |
Add libcrypto for users of libstagefright.
libstagefright_rtsp uses some MD5 functions that used to be in bionic,
but it was removed recently. As an initial fix, I statically linked in
libcrypto_static to the libstagefright_rtsp library. However, I think
it's better to modify the single user of this library to link against
the shared libcrypto library.
Change-Id: Iaf2e1aeea32fd8af038f6e77bf58ea7df50d807a
ndroid.mk
|
| 11cbb06b35cbcb488c7f39b71886ce379e57f867 |
11-Jun-2014 |
Christopher Ferris <cferris@google.com> |
resolved conflicts for merge of 281b884c to master
Change-Id: If8924939bdf54d3a9e6a4876a05d0672c27cf8ef
|
| 67ae86eea1aeb574ca19ec6b37d6e4dd7170e4c4 |
10-Jun-2014 |
Christopher Ferris <cferris@google.com> |
Link libcrypto for MD5_* functions.
Change-Id: I5dce8f041b9faf035161b82d5e46bd46166bd05c
ndroid.mk
|
| db43b34c3428e480f8c4c66e7e88f4001f37f91e |
04-Apr-2014 |
Mark Salyzyn <salyzyn@google.com> |
media: 64 bit compile issues
- change internal sized types to use stdint.h
- printf & scanf formats
- size_t or unsigned int for iterators
Change-Id: Id993a70d8bf54c667c5d652b34179a2c727ed446
DPLoader.cpp
|
| f6d0c1fd6d9e697bb3a891fae14c7e9d4b685de6 |
15-Apr-2014 |
Colin Cross <ccross@google.com> |
libstagefright: fix 64-bit warnings
%lld -> %" PRId64 " for int64_t
%d -> %zu for size_t
Also fixes some casts from void* to integer types, and some comparisons
between signed and unsigned.
(cherry picked from commit b4a7a2df4c28c3f32b5d877b54831d2cc5d78f81)
Change-Id: I76ba94d0b67776fd7abdc83b43d47c61d6c32f4c
AVCAssembler.cpp
MPEG4ElementaryAssembler.cpp
RTPConnection.cpp
RTPWriter.cpp
|
| d411b4ca2945cd8974a3a78199fce94646950128 |
11-Apr-2014 |
Andreas Huber <andih@google.com> |
warnings be gone.
(cherry picked from commit 84333e0475bc911adc16417f4ca327c975cf6c36)
Modified by Mark Salyzyn <salyzyn@google.com> to keep merge conflicts
or errors downstream to a minimum.
Change-Id: Ic3b272f9cbf3155001aabd2f79728f1bc31de613
MPEG2TSAssembler.cpp
RTPWriter.cpp
RawAudioAssembler.cpp
|
| a1df816c0677185534babba6ffc29970b048e52e |
04-Apr-2014 |
Lajos Molnar <lajos@google.com> |
stagefright: log uri protocols, and opt-in to log full uri
Added property media.stagefright.log-uri. Set it to true or 1 to
log uris by AwesomePlayer.
Added utility function to get uri debug string based on incognito
and log opt-in status.
Change-Id: I5ccc23079ddfb120dd9703a3ed651a162ed5acec
Related-Bug: 6994761
RTSPConnection.cpp
DPLoader.cpp
|
| b4a7a2df4c28c3f32b5d877b54831d2cc5d78f81 |
20-Mar-2014 |
Colin Cross <ccross@android.com> |
libstagefright: fix 64-bit warnings
%lld -> %" PRId64 " for int64_t
%d -> %zu for size_t
Also fixes some casts from void* to integer types, and some comparisons
between signed and unsigned.
Change-Id: I9c52f76240e39399da252c66459042a6fc626a90
AVCAssembler.cpp
MPEG4ElementaryAssembler.cpp
RTPConnection.cpp
RTPWriter.cpp
DPLoader.cpp
|
| beb57a5a08207af80180b93dd80d611a85997c43 |
14-Mar-2014 |
Andreas Huber <andih@google.com> |
am f1ac623f: am 4a67fc49: Merge "Implemented support for RTSP 301 Redirect"
* commit 'f1ac623fcc6bbda2faff9752cd611182a897afe1':
Implemented support for RTSP 301 Redirect
|
| 4a67fc49d926c75fa6a96160ba5627fb0e209db6 |
14-Mar-2014 |
Andreas Huber <andih@google.com> |
Merge "Implemented support for RTSP 301 Redirect"
|
| fca092d953e04c7169242200f0ddb914a9f54ea4 |
12-Mar-2014 |
Marco Nelissen <marcone@google.com> |
am f4431278: am 19afb386: Merge "Remove streaming URI from default logs"
* commit 'f4431278a9613f55ecd944ab2e3eb615b372f269':
Remove streaming URI from default logs
|
| a8b8488f703bb6bda039d7d98f87e4f9d845664d |
06-Sep-2012 |
David Williams <david.williams@sonymobile.com> |
Remove streaming URI from default logs
Streaming URI should not be visible in default logcat logs
Change-Id: I104cc56b5335f8c5621013e4c5be8028f0379833
RTSPConnection.cpp
yHandler.h
DPLoader.cpp
|
| 84333e0475bc911adc16417f4ca327c975cf6c36 |
08-Feb-2014 |
Andreas Huber <andih@google.com> |
warnings be gone.
Change-Id: Ie3bae3f037730e316d7fca12e7a3527973f752ef
MPEG2TSAssembler.cpp
RTPWriter.cpp
RawAudioAssembler.cpp
ndroid.mk
yHandler.h
|
| 81e68448f3361eaf8618930471fdc3c21bdf5cbc |
05-Feb-2014 |
Andreas Huber <andih@google.com> |
Remove no longer needed http proxy handling code, it's obsolete now
since we started to use java's HTTPConnection instead of the native
implementation. Also remove other remnants of the previous http implementation,
such as accounting for the http user's uid.
Change-Id: I60bfd31381ea40d2220db587ec5c433093b60034
DPLoader.cpp
|
| 1b86fe063badb5f28c467ade39be0f4008688947 |
29-Jan-2014 |
Andreas Huber <andih@google.com> |
FINAL ATTEMPT: HTTP services are now provided from JAVA and made available to media code
Change-Id: I9f74a86e70422187c9cf0ca1318a29019700192d
PacketSource.cpp
RTSPConnection.cpp
ndroid.mk
DPLoader.cpp
|
| 9843e8c9446aec0c25168ff4561bdbb12948f1c7 |
25-Sep-2013 |
Chong Zhang <chz@google.com> |
am 58dd0786: Merge "Send kWhatConnected in onTimeUpdate() before first access unit" into klp-dev
* commit '58dd07863571951408b67fa0a7f17cb23606fb1c':
Send kWhatConnected in onTimeUpdate() before first access unit
|
| ffd5687c9ece8e28779793a20f06f99c7199ce44 |
24-Sep-2013 |
Chong Zhang <chz@google.com> |
Send kWhatConnected in onTimeUpdate() before first access unit
Bug: 10642588
Change-Id: If2b4fbbf250d5307e304f31c7aa4ac480e279484
yHandler.h
|
| cb18b6987bb3c928b2ec69e344923b427ed39627 |
28-Aug-2013 |
Andreas Huber <andih@google.com> |
am af66fae1: am fb949d5d: Merge "Fix crash in MyHandler when sockets are not set."
* commit 'af66fae15f8c386ad884e5fa83db4eaef4c4f2ee':
Fix crash in MyHandler when sockets are not set.
|
| fb949d5dc8a764e31fbd65bee87f59fcfeb6d848 |
28-Aug-2013 |
Andreas Huber <andih@google.com> |
Merge "Fix crash in MyHandler when sockets are not set."
|
| 9bdc9c4ee0b44ca407cdca4499df9b3134bc5884 |
09-Aug-2013 |
Andreas Huber <andih@google.com> |
am bcd86896: am d0f5664a: Merge "Handle undefined NAL type for h264 streaming"
* commit 'bcd86896e486e303d285e13477e0623b2a920e78':
Handle undefined NAL type for h264 streaming
|
| d0f5664abb5a8d94ae13f63a5f3491b47383ee2f |
08-Aug-2013 |
Andreas Huber <andih@google.com> |
Merge "Handle undefined NAL type for h264 streaming"
|
| 9610adc395d18e474e6e35c0bc8b9c3220e6e525 |
31-Jul-2013 |
Andreas Huber <andih@google.com> |
am b57fb786: am d0ef1ccd: Merge "rtsp handle response line ended with \'\n\'"
* commit 'b57fb786a32d4ea78cd8bbf24a65593353d87a88':
rtsp handle response line ended with '\n'
|
| 3e3af91f70b20623fa5f3845f26260235c0b212d |
14-May-2013 |
Yajun Zeng <beanz@marvell.com> |
rtsp handle response line ended with '\n'
Change-Id: I5bfafd3fa2c95083e833da2846556282eada2b02
Signed-off-by: Yajun Zeng <beanz@marvell.com>
RTSPConnection.cpp
|
| a355bb4f5ce39a77d05f62263d4be888e903c4cd |
16-Nov-2012 |
Patrik2 Carlsson <patrik2.carlsson@sonyericsson.com> |
Handle undefined NAL type for h264 streaming
Packages of undefined NAL type (0) was observed but lead to deleting
the subsequent package due to the current assembler implementation.
Identifying and ignoring this package without returning an error
handles undefined packages without side-effects.
Change-Id: I02e15b8682bee3154b3c4acf82639a28417f0c85
AVCAssembler.cpp
|
| 59d3f809024ae5b5a7ea35dcfdd056f1c7ca42b2 |
23-Jul-2013 |
Chad Brubaker <cbrubaker@google.com> |
Fix typo in socket name
Change-Id: I29171368f1b69333ef7eae53ada2fab94e3e28b9
yHandler.h
|
| 5908f88a7e45380a9b0d71a3b1ea535d76c420b3 |
16-Jul-2013 |
Chad Brubaker <cbrubaker@google.com> |
Add routing sockets for the requesting user
Mediaserver sockets are now routed as if the connection was in the
requesting app in per user routing.
Change-Id: I60f4649c3c4145a65264b54c1aa2c6c7741efaba
RTSPConnection.cpp
yHandler.h
|
| 9046684244e6adaf4db46f1a5e5b1fea221cd781 |
08-Jul-2013 |
Jean-Baptiste Queru <jbq@google.com> |
am 1468dd9c: am c582fde9: resolved conflicts for merge of c158971f to stage-aosp-master
* commit '1468dd9cefe11d5938a5497688f99701b6b14706':
Store rtsp accessunit until PLAY response parsed
|
| c582fde93ded7219107157333a9e46d780adcf9c |
08-Jul-2013 |
Jean-Baptiste Queru <jbq@google.com> |
resolved conflicts for merge of c158971f to stage-aosp-master
Change-Id: I3d77b86f7e616af62a826fc37126706ad8ff6158
|
| bbbf9c4552402ab18b255f4058e9e6e506f3f106 |
24-Apr-2013 |
Yajun Zeng <beanz@marvell.com> |
Store rtsp accessunit until PLAY response parsed
If RTP accessunit comes earlier than play response,
the normal play time mapping posted in func onAccessUnitComplete is wrong.
This leads wrong timestamp of the first few frames.
This issue is found in the 3 CtsVerifier RTSP streaming cases.
Change-Id: I640eea375b1f3f4730238f9d561c3b40ec682395
Signed-off-by: Yajun Zeng <beanz@marvell.com>
yHandler.h
|
| 89407b01795ebc56033b09e3a48defaa290bb3c5 |
24-Apr-2013 |
Andreas Huber <andih@google.com> |
am 0fb06b85: am 0dbff625: Merge "Fix overflow of rand in ARTPConnection"
* commit '0fb06b85e9f40cc695542a101113255693c91321':
Fix overflow of rand in ARTPConnection
|
| a3840fdfe6fdb8dd07d78d3f3202003649e952e9 |
24-Apr-2013 |
Andreas Huber <andih@google.com> |
am 0dbff625: Merge "Fix overflow of rand in ARTPConnection"
* commit '0dbff625c3128962b48f3476ceacb3ac80a3f421':
Fix overflow of rand in ARTPConnection
|
| 0fb06b85e9f40cc695542a101113255693c91321 |
24-Apr-2013 |
Andreas Huber <andih@google.com> |
am 0dbff625: Merge "Fix overflow of rand in ARTPConnection"
* commit '0dbff625c3128962b48f3476ceacb3ac80a3f421':
Fix overflow of rand in ARTPConnection
|
| be21e039d7d993872ac85a0279ea657e40f674fd |
24-Apr-2013 |
Yajun Zeng <beanz@marvell.com> |
Fix overflow of rand in ARTPConnection
without this fix, (rand()*1000)/RAND_MAX is mainly 0.
Change-Id: I48ae940a7b6974b197d81732774c9dcea107bcf1
Signed-off-by: Yajun Zeng <beanz@marvell.com>
RTPConnection.cpp
|
| 6cb3f224d7e2280f8834d361bba1a72682aaaad1 |
24-Apr-2013 |
Yajun Zeng <beanz@marvell.com> |
Fix overflow of rand in ARTPConnection
without this fix, (rand()*1000)/RAND_MAX is mainly 0.
Change-Id: I48ae940a7b6974b197d81732774c9dcea107bcf1
Signed-off-by: Yajun Zeng <beanz@marvell.com>
RTPConnection.cpp
|
| c2140bb6c7e91e77bb6cdae4b1e4db83e1d786fa |
26-Mar-2013 |
Andreas Huber <andih@google.com> |
am 1e7d497c: am cd77d4a1: Identify network servers and clients with a OS version related string
* commit '1e7d497c91e429b70fff592e6ae78aa81a4cea16':
Identify network servers and clients with a OS version related string
|
| 190cdbab6ba24519d6b5e8bec6c2c74e6650e284 |
26-Mar-2013 |
Andreas Huber <andih@google.com> |
Identify network servers and clients with a OS version related string
and put the logic to create that string in one location instead of many...
Change-Id: I1f729f2e7376cd3b45eea0e48f7bd10084b41b39
RTSPConnection.cpp
RTSPConnection.h
yHandler.h
|
| cd77d4a1d38b7609a03f6826a1ff5fa7c98aa34f |
26-Mar-2013 |
Andreas Huber <andih@google.com> |
Identify network servers and clients with a OS version related string
and put the logic to create that string in one location instead of many...
Change-Id: I1f729f2e7376cd3b45eea0e48f7bd10084b41b39
RTSPConnection.cpp
RTSPConnection.h
yHandler.h
|
| 0e6858d6aea12fc585a8c7d217c1271878655081 |
07-Mar-2013 |
Dan Morrill <morrildl@google.com> |
Turn off debug tags in stagefright modules.
LOCAL_MODULE_TAGS := debug causes the module to be included in every userdebug
build, regardless of whether it's specified as a dep by the device config.
This CL switches them all to optional (i.e. default behavior) so that we can
do (userdebug) device builds without pulling these in.
Change-Id: I4b7b65afea61865dd38b3af55550fb8f10edf66d
ndroid.mk
|
| 4f4c2655dc3f6fcef766db6e793b1642ad0fd605 |
15-Mar-2013 |
Andreas Huber <andih@google.com> |
am 59ac7b30: am 66abe3ae: Merge "Fix for crash if no content in DESCRIBE response"
* commit '59ac7b3056db57e5a8e851b7946a181c5fc34852':
Fix for crash if no content in DESCRIBE response
|
| ee6ad3bd4bfc8e71b3b8c96eb4ea56a592e13e65 |
15-Mar-2013 |
Andreas Huber <andih@google.com> |
am 59ac7b30: am 66abe3ae: Merge "Fix for crash if no content in DESCRIBE response"
* commit '59ac7b3056db57e5a8e851b7946a181c5fc34852':
Fix for crash if no content in DESCRIBE response
|
| 5f1897538bab324f53efc6bec65487516041f2e9 |
07-Jan-2013 |
Xuefei Chen <xuefei.chen@sonymobile.com> |
Fix for crash if no content in DESCRIBE response
If DESCRIBE response is received with status 200 but no content,
MyHandler will still set content data for session description
parsing. This will cause NULL Pointer crash.
This fix checks whether DESCRIBE response has content before
parsing session description.
Change-Id: I114ae6fd54ce804e61718f62618ca9008425a433
yHandler.h
|
| d32b7b479fad359d7fe779a9c5b4c090cdc14b56 |
07-Jan-2013 |
Xuefei Chen <xuefei.chen@sonymobile.com> |
Fix for crash if no content in DESCRIBE response
If DESCRIBE response is received with status 200 but no content,
MyHandler will still set content data for session description
parsing. This will cause NULL Pointer crash.
This fix checks whether DESCRIBE response has content before
parsing session description.
Change-Id: I114ae6fd54ce804e61718f62618ca9008425a433
yHandler.h
|
| 0955986e6c1c27ba752e293246086ea79c49d39c |
23-Jan-2013 |
Roger Jönsson <roger1.jonsson@sonymobile.com> |
Avoid rebuffering after RTSP pause
If pausing an RTSP stream, an RTSP Pause request is sent and then
if the stream is immediately resumed again, an RTSP Play request
will be sent to the server.
But the new data after the pause will not be buffered until
Sender Reports have arrived again on both channels.
Meanwhile the player will resume playback and continue consuming
the already existing buffer.
This means that there is a risk that the buffer is emptied while
waiting for sender reports.
This commit simply adds a delay before the RTSP pause request is
sent, allowing some additional RTSP buffering that might be needed
when the stream is resumed again.
Also, if the stream is resumed again before the RTSP pause request
is sent, there is no need for any RTSP pause request, hence it is
omitted.
Change-Id: I928c8bfb5e99a6a146dcda4e51e528973ecbe065
yHandler.h
|
| a0dd006834f4a424b67773ab6724e961a61de923 |
23-Jan-2013 |
Roger Jönsson <roger1.jonsson@sonymobile.com> |
Avoid rebuffering after RTSP pause
If pausing an RTSP stream, an RTSP Pause request is sent and then
if the stream is immediately resumed again, an RTSP Play request
will be sent to the server.
But the new data after the pause will not be buffered until
Sender Reports have arrived again on both channels.
Meanwhile the player will resume playback and continue consuming
the already existing buffer.
This means that there is a risk that the buffer is emptied while
waiting for sender reports.
This commit simply adds a delay before the RTSP pause request is
sent, allowing some additional RTSP buffering that might be needed
when the stream is resumed again.
Also, if the stream is resumed again before the RTSP pause request
is sent, there is no need for any RTSP pause request, hence it is
omitted.
Change-Id: I928c8bfb5e99a6a146dcda4e51e528973ecbe065
yHandler.h
|
| 1a37ee3c877165c812734b405f922f6e0d747052 |
23-Jan-2013 |
joakim johansson <joakim.c.johansson@sonyericsson.com> |
EOS fixes for RTSP streams
The fix takes care of several near end of stream use cases:
seek, pause and fake timestamps.
Change-Id: I5f5fa881b1f619dfd5e1afd2af957082345c59eb
yHandler.h
|
| ba021d15cf7bc964bc813688e33d34845bfd89ea |
23-Jan-2013 |
joakim johansson <joakim.c.johansson@sonyericsson.com> |
EOS fixes for RTSP streams
The fix takes care of several near end of stream use cases:
seek, pause and fake timestamps.
Change-Id: I5f5fa881b1f619dfd5e1afd2af957082345c59eb
yHandler.h
|
| b6ec588faa7728ff3b518bf809ff75e8dd14f08c |
23-Jan-2013 |
Måns Zigher <mans.zigher@sonyericsson.com> |
RTSP: Parse session level control attribute from SDP
If a=control: is present at session-level in the SDP response,
RFC2326:C.1.1 defines the URL to be used for aggregate commands.
This includes PLAY and PAUSE but not TEARDOWN.
Change-Id: Iaa1dc2271d00df39dc83477a99fda6fbeb73c5b4
yHandler.h
|
| 599b9655ddf95cdf6cb99970ce03c632bb2a576b |
23-Jan-2013 |
Måns Zigher <mans.zigher@sonyericsson.com> |
RTSP: Parse session level control attribute from SDP
If a=control: is present at session-level in the SDP response,
RFC2326:C.1.1 defines the URL to be used for aggregate commands.
This includes PLAY and PAUSE but not TEARDOWN.
Change-Id: Iaa1dc2271d00df39dc83477a99fda6fbeb73c5b4
yHandler.h
|
| 46d13e3606b87d71379287672b54b50d0d9aa5cc |
21-Jan-2013 |
Roger Jönsson <roger1.jonsson@sonymobile.com> |
Enable pause/resume for RTSP streaming
When a stream is paused, RTSP Pause is also sent to the server.
Otherwise the buffering might continue until the memory runs out.
When the stream is resumed, RTSP Play will be sent in order to
resume the buffering.
Change-Id: I5dc1761140827c532451638c3fd3f34271e5b9ab
yHandler.h
|
| fba60daf77cc74a13ae3bf4b0e9925dd2ee4470c |
21-Jan-2013 |
Roger Jönsson <roger1.jonsson@sonymobile.com> |
Enable pause/resume for RTSP streaming
When a stream is paused, RTSP Pause is also sent to the server.
Otherwise the buffering might continue until the memory runs out.
When the stream is resumed, RTSP Play will be sent in order to
resume the buffering.
Change-Id: I5dc1761140827c532451638c3fd3f34271e5b9ab
yHandler.h
|
| cfc3083927df14bf82403b20a45ae303a01c39f5 |
21-Jan-2013 |
Roger Jönsson <roger1.jonsson@sonymobile.com> |
RTSP buffering improvements
Added buffering start and end notifications for RTSP.
MEDIA_INFO_BUFFERING_START is sent when buffering is started
and MEDIA_INFO_BUFFERING_END is sent when the buffer has
filled up.
This patch also adds RTSP end of stream handling.
EOS is signalled when BYE is received OR when
detecting end of stream even if no actual EOS is received.
Change-Id: I5cccb6845060ae6afd66d9f735b89da81476cd13
yHandler.h
|
| b50e83eca302a12f0fced6e7bab1b8617d63deaa |
21-Jan-2013 |
Roger Jönsson <roger1.jonsson@sonymobile.com> |
RTSP buffering improvements
Added buffering start and end notifications for RTSP.
MEDIA_INFO_BUFFERING_START is sent when buffering is started
and MEDIA_INFO_BUFFERING_END is sent when the buffer has
filled up.
This patch also adds RTSP end of stream handling.
EOS is signalled when BYE is received OR when
detecting end of stream even if no actual EOS is received.
Change-Id: I5cccb6845060ae6afd66d9f735b89da81476cd13
yHandler.h
|
| 7f475c34ffc8e35345f2cceee2ef56a50bb5fea6 |
05-Feb-2013 |
Andreas Huber <andih@google.com> |
RTSP now properly publishes its "seekable" flags after connection
has successfully completed and only then signals that preparation is
complete.
Change-Id: I1a60f718e673fe1462c69369c40eafbed6a14326
yHandler.h
|
| ec0c597cabf169ca646bcea5faac1bd81ed4484d |
05-Feb-2013 |
Andreas Huber <andih@google.com> |
RTSP now properly publishes its "seekable" flags after connection
has successfully completed and only then signals that preparation is
complete.
Change-Id: I1a60f718e673fe1462c69369c40eafbed6a14326
yHandler.h
|
| ec29a2bfb5364a5968b77559fd13821b827d173a |
17-Jan-2013 |
Roger Jönsson <roger1.jonsson@sonymobile.com> |
Detect live streams
The information is used to decide on visibility of pause button and
to handle the duration clock correctly.
Change-Id: I286ac992fd171c7fc313e429326d38b6fc80e3fb
yHandler.h
|
| 84ca0414fedea2dfe51607b422f6227e1c4f0d7f |
17-Jan-2013 |
Roger Jönsson <roger1.jonsson@sonymobile.com> |
Detect live streams
The information is used to decide on visibility of pause button and
to handle the duration clock correctly.
Change-Id: I286ac992fd171c7fc313e429326d38b6fc80e3fb
yHandler.h
|
| 81dd60e0340ddcf7f1d5fb80b6c30906fabf201a |
20-Feb-2012 |
Oscar Rydhé <oscar.rydhe@sonyericsson.com> |
Added HTTP support for SDP files.
Added support for playing SDP files from http links. Previously,
SDP files only worked when started from rtsp links
(rtsp://a.b.c/abc.sdp), but they are just as common in http links.
patch provided by "Oscar Rydhé <oscar.rydhe@sonyericsson.com>"
Change-Id: Ic73af3a9a002009dbe8b04c267a4621bf7fe2f46
ndroid.mk
yHandler.h
DPLoader.cpp
|
| 7a33b7740412accf6a1cc912686c8d0acfb2a883 |
20-Feb-2012 |
Oscar Rydhé <oscar.rydhe@sonyericsson.com> |
Added HTTP support for SDP files.
Added support for playing SDP files from http links. Previously,
SDP files only worked when started from rtsp links
(rtsp://a.b.c/abc.sdp), but they are just as common in http links.
patch provided by "Oscar Rydhé <oscar.rydhe@sonyericsson.com>"
Change-Id: Ic73af3a9a002009dbe8b04c267a4621bf7fe2f46
ndroid.mk
yHandler.h
DPLoader.cpp
|
| cc4e6091bd24f84e69d4baf4fce6ceae67676ee5 |
21-Dec-2012 |
James Dong <jdong@google.com> |
Merge "Use default values when MPEG4 audio config parsing fails."
|
| b54cedabdf0261211241e2f3af09c75cffd911ed |
21-Dec-2012 |
James Dong <jdong@google.com> |
Merge "Use default values when MPEG4 audio config parsing fails."
|
| b6f7642496f955da04d1eb9e33df0dab653c9c4e |
20-Sep-2011 |
Henrik Backlund <henrik.backlund@sonyericsson.com> |
Fix crash in MyHandler when sockets are not set.
-When going quickly in and out of the video view during an rtsp
streaming session, a race condition occurs and MyHandler tries to
connect to a socket that has been reset. To avoid this,
checks are added.
- If there are errors during setupTrack 1, it is no use
setting up track 2. It will cause new errors.
- No assert for socket connect since there is a normal
status check already.
Change-Id: Ie06221d6c0d78ce0449f76c782ed5120fa646bfd
RTSPConnection.cpp
yHandler.h
|
| 4bb026ba585d5b37795bd9765459f0607b7aa60a |
24-Feb-2011 |
David Williams <david.williams@sonyericsson.com> |
Implemented support for RTSP 301 Redirect
RTSP 301 (Permament Redirect) support has been implemented.
Change-Id: If82ffc87f4e7dcbdf98e0a662a35cc086750fc1b
yHandler.h
|
| b90b748d7484f1d464cd9e15289d77b83beed10e |
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
AMRAssembler.cpp
H263Assembler.cpp
|
| 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
AMRAssembler.cpp
H263Assembler.cpp
|
| 78cc49b4c4b25ea51dc5f6a6878ea158056bcf32 |
20-Jan-2012 |
Lena Magnusson <lena.magnusson@sonyericsson.com> |
Unsolicited server responses cause RTSP streaming to crash
If the set up of the RTSP stream contains an incorrect or otherwise
problematic URL, some servers will send an unsolicited server response
that contains a negative number in the sequence number (CSeq) field.
This negative value is not returned from the function findPendingRequest(),
so the check in notifyResponseListener() will not work. Instead there will
be a crash when 0 is used as the index to find a matching request/response
pair that doesn’t exist.
The fix is to return the received sequence number also when it is an
unsolicited server-client message.
Change-Id: Iedaba8a63dece7b43bce007069baefbfd10970b8
RTSPConnection.cpp
|
| 8b96e5df9f085e285d23beb96fd41c3d4b8005a3 |
20-Jan-2012 |
Lena Magnusson <lena.magnusson@sonyericsson.com> |
Unsolicited server responses cause RTSP streaming to crash
If the set up of the RTSP stream contains an incorrect or otherwise
problematic URL, some servers will send an unsolicited server response
that contains a negative number in the sequence number (CSeq) field.
This negative value is not returned from the function findPendingRequest(),
so the check in notifyResponseListener() will not work. Instead there will
be a crash when 0 is used as the index to find a matching request/response
pair that doesn’t exist.
The fix is to return the received sequence number also when it is an
unsolicited server-client message.
Change-Id: Iedaba8a63dece7b43bce007069baefbfd10970b8
RTSPConnection.cpp
|
| 738198a16cfd7b125d15b0bab0708ba7fbf7e64a |
26-Sep-2011 |
Patric Frederiksen <patric.frederiksen@sonyericsson.com> |
Crash in android::MyHandler::parsePlayResponse
This fix handles problems with several asynchronous calls
within streaming. This case is when the phone has sent a
request to the server and while the response is being sent
back by the server the request is aborted by the user.
The fix is an if case that checks if we have aborted while
waiting for a response from the server. If we have aborted
we should ignore the late response instead of continuing.
Change-Id: I1264bb992f6abcaee1f10a89479e08b54a95e3c2
yHandler.h
|
| e1a31d16dda3460a34e5dfd65c4e96e422dbdbfc |
26-Sep-2011 |
Patric Frederiksen <patric.frederiksen@sonyericsson.com> |
Crash in android::MyHandler::parsePlayResponse
This fix handles problems with several asynchronous calls
within streaming. This case is when the phone has sent a
request to the server and while the response is being sent
back by the server the request is aborted by the user.
The fix is an if case that checks if we have aborted while
waiting for a response from the server. If we have aborted
we should ignore the late response instead of continuing.
Change-Id: I1264bb992f6abcaee1f10a89479e08b54a95e3c2
yHandler.h
|
| a45a600d69a5d8ab99eeb7e0dfa58c3cb99a2e61 |
19-Sep-2011 |
Erik Rydgren <erik.rydgren@sonyericsson.com> |
Use default values when MPEG4 audio config parsing fails.
MPEG4 audio packets may be multiplexed using the so called
LATM (Low Overhead Audio Transport Multiplex) scheme.
LATM parsing was recently introduced in Stagefright and it
has caused issues in cases when the LATM config element
cannot be parsed correctly. The main problem occurrs when
the AudioSpecificConfig part of the config element contains
more information than what is expected, causing the
frameLengthType parameter to get the wrong value. This fix
introduces default values of some config parameters that are
used in case config parsing fails.
Change-Id: I3cb35df76826f95ca0831dc08c2a1e7c6c2c586d
MPEG4AudioAssembler.cpp
|
| 8af5fe5a2431522a7d30bc546dcd31c0c64db70c |
19-Sep-2011 |
Erik Rydgren <erik.rydgren@sonyericsson.com> |
Use default values when MPEG4 audio config parsing fails.
MPEG4 audio packets may be multiplexed using the so called
LATM (Low Overhead Audio Transport Multiplex) scheme.
LATM parsing was recently introduced in Stagefright and it
has caused issues in cases when the LATM config element
cannot be parsed correctly. The main problem occurrs when
the AudioSpecificConfig part of the config element contains
more information than what is expected, causing the
frameLengthType parameter to get the wrong value. This fix
introduces default values of some config parameters that are
used in case config parsing fails.
Change-Id: I3cb35df76826f95ca0831dc08c2a1e7c6c2c586d
MPEG4AudioAssembler.cpp
|
| af5dd7753e62353411cf0daf3b513c38818e9662 |
02-Oct-2012 |
Andreas Huber <andih@google.com> |
ALooper::GetNowUs() now relies on systemTime instead of gettimeofday.
Change-Id: Ia31eb0940b02581327a8bf51af6df135f9ab6de3
related-to-bug: 7266324
RTPAssembler.cpp
|
| fa0e033ab5a0ab5d96e90c9f6d4d53bedc74514b |
02-Oct-2012 |
Andreas Huber <andih@google.com> |
ALooper::GetNowUs() now relies on systemTime instead of gettimeofday.
Change-Id: Ia31eb0940b02581327a8bf51af6df135f9ab6de3
related-to-bug: 7266324
RTPAssembler.cpp
|
| cfaeeec0900014d97e15829e0fa52f865ee4c786 |
31-Aug-2012 |
Andreas Huber <andih@google.com> |
Add support for mpeg2 transport streams to the RTSP implementation.
Change-Id: I409d7133a53a71e62523b1acc2b03302fcf824a5
MPEG2TSAssembler.cpp
MPEG2TSAssembler.h
PacketSource.cpp
RTPSource.cpp
ndroid.mk
|
| 49694688c82214f5fd9e969e177c9e126a240a26 |
31-Aug-2012 |
Andreas Huber <andih@google.com> |
Add support for mpeg2 transport streams to the RTSP implementation.
Change-Id: I409d7133a53a71e62523b1acc2b03302fcf824a5
MPEG2TSAssembler.cpp
MPEG2TSAssembler.h
PacketSource.cpp
RTPSource.cpp
ndroid.mk
|
| 3677437296fd1547d762b1b227a3de83dbc960d6 |
27-Jul-2012 |
Tareq A. Siraj <tareq.a.siraj@intel.com> |
Fixed member access into incomplete type build error
Included the ARTPAssembler.h file to fix the 'member access into
incomplete type "android::ARTPAssembler"' error reported by clang.
Change-Id: I10cb1e38bf360858bb7ebdeae82ba1e64431f87d
Author: Tareq A. Siraj <tareq.a.siraj@intel.com>
Reviewed-by: Edwin Vane<edwin.vane@intel.com>
RTPConnection.cpp
|
| 8033393a74a6872ad8d702b10da34d98dde0bf41 |
20-Aug-2012 |
Patrik2 Carlsson <patrik2.carlsson@sonymobile.com> |
h264 streaming: make profile-level-id optional
profile-level-id is made optional according to rfc3984:
"If no profile-level-id is present, the Baseline Profile without
additional constraints at Level 1 MUST be implied."
Change-Id: If868468a48917ceccb963b8ac15767583da29723
PacketSource.cpp
|
| 3d51d5cb53cc630709a0ba78d0e60501a675f2d5 |
13-Jun-2012 |
James Dong <jdong@google.com> |
Add NOTICE and MODULE_LICENSE_APACH2 to libs build under /frameworks/av/
Change-Id: I0a3af3e2abdedebd5934f3d941d01c32cfc75e26
related-to-bug: 6647465
ODULE_LICENSE_APACHE2
OTICE
|
| 8647bbe4420ca487467318404127f52c567e346b |
17-May-2012 |
Andreas Huber <andih@google.com> |
Prefix MPEG4-generic audio data with ADTS headers
to work around limitations of the new AAC decoder.
Change-Id: I4988c7c39fedb7d04eb1ae2ba2d618aa6cb14e77
related-to-bug: 6488547
MPEG4AudioAssembler.cpp
MPEG4ElementaryAssembler.cpp
MPEG4ElementaryAssembler.h
PacketSource.cpp
RTPAssembler.cpp
RTPAssembler.h
|
| f95439afa8eb2484969d4a928b0fdd6a4d3a38d7 |
11-Apr-2012 |
Andreas Huber <andih@google.com> |
Changes to add support for H263-1999/2000 formats for streaming
contributed by sureshc@nvidia.com (and subsequently simplified)
Change-Id: Ia1c2ac9233f5414ce3e4a70e42e68c1c5c35eb9d
H263Assembler.cpp
|
| 559bf2836f5da25b75bfb229fec0d20d540ee426 |
28-Mar-2012 |
James Dong <jdong@google.com> |
AV Android make files changes
o plus a few file relocation: ActivityManager.cpp/h, SoundPool.h, etc
o remove some runtime dependencies to libandroid, libandroid_runtime, etc
Change-Id: I047a47c5fb361dd5cf85cd98798c39f629a75d10
ndroid.mk
|
| 3ee26944b082def647fe5bb2b75116ffb0267059 |
24-Mar-2012 |
James Dong <jdong@google.com> |
Remove JNI in LOCAL_C_INCLUDE from non-JNI related Android.mk files.
o related-to-bug: 6214141
Change-Id: Ic88d1732b3e014af47532a0809e01f6086e8464d
ndroid.mk
|
| 6c6b4d0d2b98a7ceee8b697daaf611f8df3254fb |
12-Mar-2012 |
James Dong <jdong@google.com> |
Switched to use the header files in /frameworks/native
and deleted the duplicate header files in /frameworks/base
o related-to-bug: 6044887
Change-Id: I17e0692d9a9b5c8796ded36677c833ca8ab36795
ndroid.mk
|
| 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
AMRAssembler.cpp
AVCAssembler.cpp
H263Assembler.cpp
MPEG4AudioAssembler.cpp
MPEG4ElementaryAssembler.cpp
RTPConnection.cpp
RTPSession.cpp
RTSPConnection.cpp
RawAudioAssembler.cpp
yHandler.h
|
| 7e73e44c2d2208a7079e562f7b0b9b73ef6a29f1 |
20-Jan-2012 |
Andreas Huber <andih@google.com> |
Starhub RTSP apparently does not establish time on all tracks
i.e. the "SR" RTCP packet is sent for only one of the two tracks.
fake timestamps if that's the case, previously we'd only fake timestamps
if we didn't receive _any_ "SR" packets.
Change-Id: Id63d4940d453ba6c04c62e02ab9a0ad843936bc1
related-to-bug: 5669027
yHandler.h
|
| 29357bc2c0dd7c43ad3bd0c8e3efa4e6fd9bfd47 |
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
RTPSession.cpp
RTSPConnection.cpp
yHandler.h
DPPusher.cpp
tp_test.cpp
|
| 5ff1dd576bb93c45b44088a51544a18fc43ebf58 |
06-Jan-2012 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGW(_IF) to (IF_)ALOGW(_IF) DO NOT MERGE
See https://android-git.corp.google.com/g/157065
Bug: 5449033
Change-Id: I00a4b904f9449e6f93b7fd35eac28640d7929e69
RTPConnection.cpp
RTPSource.cpp
RTSPConnection.cpp
yHandler.h
|
| df64d15042bbd5e0e4933ac49bf3c177dd94752c |
04-Jan-2012 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGI(_IF) to (IF_)ALOGI(_IF) DO NOT MERGE
See https://android-git.corp.google.com/g/156801
Bug: 5449033
Change-Id: Ib08fe86d23db91ee153e9f91a99a35c42b9208ea
MPEG4AudioAssembler.cpp
PacketSource.cpp
RTPConnection.cpp
RTPSession.cpp
RTPWriter.cpp
RTSPConnection.cpp
SessionDescription.cpp
yHandler.h
DPPusher.cpp
|
| 6af1e76b61d04ed524b570f92091680a851207df |
12-Dec-2011 |
Andreas Huber <andih@google.com> |
Merge "Support for parsing of "folded" RTSP header values and some tweaks to the AMR assembler"
|
| aa5ba9a27f4c483ee116b7b296a681f4f8e23e62 |
10-Dec-2011 |
Andreas Huber <andih@google.com> |
am aa82c39b: am 0ba9380a: Merge "Fix Bitreader "putBits" implementation, make sure we emulate timestamps" into ics-mr1
* commit 'aa82c39bdb4ad9c1fdcb09f3bea11be5197d3ce6':
Fix Bitreader "putBits" implementation, make sure we emulate timestamps
|
| 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
AMRAssembler.cpp
RTSPConnection.cpp
|
| 1906e5c7492b9cbc88601365536a69e9a490c963 |
08-Dec-2011 |
Andreas Huber <andih@google.com> |
Fix Bitreader "putBits" implementation, make sure we emulate timestamps
if we don't receive npt time mapping from the rtsp server (i.e. live stream)
Change-Id: I5147d665bd90c9a303ad6ffdafbf770f930f917c
related-to-bug: 5660357
yHandler.h
|
| 78ff828e28c22715f5b6c320d967744cb4f51fd4 |
11-Nov-2011 |
Andreas Huber <andih@google.com> |
am 8a065423: am 23217182: Merge "Didn\'t mean to check this in..." into ics-mr1
* commit '8a0654231ff36d938bc3451190cf67231195f1d0':
Didn't mean to check this in...
|
| 516fb1dad0c434fd89624c418543d35436a5374c |
11-Nov-2011 |
Andreas Huber <andih@google.com> |
am 40461ee7: am cd556b82: Merge "Instead of asserting, signal a runtime error if the session doesn\'t contain" into ics-mr1
* commit '40461ee70161d8568663332f72be2353b04c34e7':
Instead of asserting, signal a runtime error if the session doesn't contain
|
| a36d8caf15d56a75906e9cc75b5e04463c1317a6 |
11-Nov-2011 |
Andreas Huber <andih@google.com> |
am 9c981cd3: am d9f25bc9: Merge "Disconnect on socket error on the RTSP control connection." into ics-mr1
* commit '9c981cd3d53238f10842368c1cd82d625b701a47':
Disconnect on socket error on the RTSP control connection.
|
| 91f230461288a2a5091182ef9e17079aabf8ebaa |
11-Nov-2011 |
Andreas Huber <andih@google.com> |
Didn't mean to check this in...
Change-Id: Ie5a1902ff2613cd349ca5724f63a3fe3306640c7
yHandler.h
|
| 73b1fd56d99b356b0effe8cf96ecf7446beb207f |
11-Nov-2011 |
Andreas Huber <andih@google.com> |
Merge "Instead of asserting, signal a runtime error if the session doesn't contain" into ics-mr1
|
| 4ab3045755d33ab24bd312cfbc888f300c5e01f9 |
11-Nov-2011 |
Andreas Huber <andih@google.com> |
Merge "DO NOT MERGE: Instead of asserting, remove active streams if their sockets" into ics-mr1
|
| 0fbe0577cfeda28bd016110e670708cce0752044 |
10-Nov-2011 |
Andreas Huber <andih@google.com> |
Disconnect on socket error on the RTSP control connection.
Change-Id: Ib52a69f9b0830b481c6f5c9b1991d1f4cb36ec7b
RTSPConnection.cpp
RTSPConnection.h
|
| 19de627354d465c4e9ccd1fcdcffd132861330b2 |
09-Nov-2011 |
Andreas Huber <andih@google.com> |
DO NOT MERGE: Instead of asserting, remove active streams if their sockets
return failure
Change-Id: Icb47adfd2fbe0398c473ba66e068186311c9cc79
related-to-bug: 5593654
RTPConnection.cpp
|
| f0c86a83c687074be79397e082e3775ca56641b1 |
10-Nov-2011 |
Andreas Huber <andih@google.com> |
Instead of asserting, signal a runtime error if the session doesn't contain
any playable tracks at all.
Change-Id: Ibbbe2fdcd53b7e020da80c84c8229856107a87e6
yHandler.h
|
| 7cad0b48243f86c516181d09185dc83223ae51d7 |
10-Nov-2011 |
Andreas Huber <andih@google.com> |
am 9e2949c6: am 2375d163: Merge "Send RTSP control connection keep-alive requests" into ics-mr1
* commit '9e2949c6ab4e791b5c20d5e85c3eff62f206a99b':
Send RTSP control connection keep-alive requests
|
| 8c308ffd781132c8417cebc3bf77c2e56a464e0b |
09-Nov-2011 |
Andreas Huber <andih@google.com> |
Instead of asserting, remove active streams if their sockets return failure
Change-Id: Ic5cc786f718cf921876b181927cf1b03e8373ff1
related-to-bug: 5593654
RTPConnection.cpp
|
| 908dbdee96856693decc04fa143c2ba525495d43 |
09-Nov-2011 |
Andreas Huber <andih@google.com> |
Send RTSP control connection keep-alive requests
default to 60 secs unless overridden by server's session-id response.
Change-Id: I7c3aff5b787dbb57cc0dccf9db3c75e5cf7e778c
related-to-bug: 5562303
yHandler.h
|
| 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
AMRAssembler.cpp
AVCAssembler.cpp
MPEG4ElementaryAssembler.cpp
RTPConnection.cpp
RTPSource.cpp
RTPWriter.cpp
RTSPConnection.cpp
RawAudioAssembler.cpp
SessionDescription.cpp
yHandler.h
|
| 2bfdd428c56c7524d1a11979f200a1762866032d |
12-Oct-2011 |
Andreas Huber <andih@google.com> |
NuPlayer is now taking on the task of streaming over RTSP.
Change-Id: Ie204db8810807f1e7981959e34dc0149e5d9563a
PacketSource.cpp
PacketSource.h
RTSPController.cpp
ndroid.mk
yHandler.h
|
| a23456b306f35b9ecf973bf5818ca39295e9e029 |
08-Jul-2011 |
Ashish Sharma <ashishsharma@google.com> |
Network traffic accounting for chromium stack support in mediaserver.
- Atribute network activity to uid calling the mediaplayer
- Enables logging of chromium network stack in logcat
Change-Id: I2d28c8392248a056b3cee305dd4d4475ebba4337
RTSPConnection.cpp
yHandler.h
|
| f89d780df70b7fbb8465bce4913c46cca019721f |
05-Aug-2011 |
Andreas Huber <andih@google.com> |
Eliminate superfluous memcpys by wrapping an ABuffer in a MediaBuffer
Change-Id: I1313f117cd7cdfaf7d6ec25413a0b4b8ea495037
related-to-bug: 5122973
PacketSource.cpp
|
| dab718bba3945332dc75e268e1e7f0fe2eb91c4a |
14-Jul-2011 |
Andreas Huber <andih@google.com> |
Remove legacy http support from stagefright, chromium is the new hotness.
Change-Id: I6725d42d38b91e6a1cbca43174870f445aeb3d99
RTSPConnection.cpp
yHandler.h
|
| 9b80c2bdb205bc143104f54d0743b6eedd67b14e |
01-Jul-2011 |
Andreas Huber <andih@google.com> |
Charge network traffic to the uid of the process using the MediaPlayer.
Change-Id: I2bcb54b8232afd3fc7ee16289f37c7a7b3f23067
related-to-bug: 4517282
RTSPConnection.cpp
RTSPConnection.h
RTSPController.cpp
yHandler.h
|
| ac5767a96df9fae46a37ffba62611472135a0f6d |
30-Jun-2011 |
Andreas Huber <andih@google.com> |
Revert "Parse RTP-Info even for live streams."
This reverts commit d873413ff9f742f259c29d7d0b58265db6b24529.
SessionDescription.cpp
yHandler.h
|
| a6925e6149faf4a936a5b557a769d117454413d8 |
01-Jun-2011 |
Andreas Huber <andih@google.com> |
Parse RTP-Info even for live streams.
Change-Id: Ib2c39ce8d5366f5ea350e71b7a54f5f7c2b510b9
SessionDescription.cpp
yHandler.h
|
| 386d609dc513e838c7e7c4c46c604493ccd560be |
19-May-2011 |
Andreas Huber <andih@google.com> |
Support mpeg1,2 audio and mpeg1,2,4 video content extraction from .ts streams.
Change-Id: I9d2ee63495f161e30daba7c3aab16cb9d8ced6a5
PacketSource.cpp
|
| e681b91c27439907f216cb6c88426929bc5194bf |
29-Mar-2011 |
Andreas Huber <andih@google.com> |
Add a user-agent header to our RTSP requests.
Change-Id: I02f8ff6a4a37fa59cc8c5fcfd3afb64ee11ba576
related-to-bug: 4173725
RTSPConnection.cpp
RTSPConnection.h
|
| fcea8f7a7d178e5426aa06586cff54726e18d1f6 |
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
PacketSource.cpp
RTPSource.cpp
RawAudioAssembler.cpp
RawAudioAssembler.h
ndroid.mk
|
| 55e26193c885b7d5acdae9978848e6587987790f |
22-Feb-2011 |
Andreas Huber <andih@google.com> |
Support more MPEG4-LATM audio functionality.
related-to-bug: 3474610
Change-Id: I6dab40e8b465922c62be9ee7f168718822c6caac
Now skipping extra header that the spec claimed shouldn't be present in LATM...
MPEG4AudioAssembler.cpp
MPEG4AudioAssembler.h
|
| 5ef152132b477a07fa31b2ddd39f4cf7a29f68b4 |
16-Feb-2011 |
Andreas Huber <andih@google.com> |
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: I557865ac00d0fb65ffa69363eb1eceaabe522a1a
related-to-bug: 3353752
RTSPConnection.cpp
RTSPConnection.h
|
| de9a20c274983d4f7a688acb68d5dfc6a432eb10 |
15-Feb-2011 |
Andreas Huber <andih@google.com> |
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.
related-to-bug: 3457201
Change-Id: I0a523f38e6959bf00b8b140a70bb65fcc414c9c1
yHandler.h
|
| dc468c5f9d72ce54de0070493e9a23efb8907e06 |
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
MPEG4ElementaryAssembler.cpp
PacketSource.cpp
RTPConnection.cpp
RTPSource.cpp
SessionDescription.cpp
yHandler.h
|
| f1958f9442bc937e1f8c8d9175901500b944b021 |
14-Feb-2011 |
Andreas Huber <andih@google.com> |
Enable cancelling the rtsp connection process early.
Change-Id: Ie2059c54541ad8c675944d71b39c772b0f6f04c8
related-to-bug: 3452699
RTSPController.cpp
|
| 864d06670089f79bc177a51fd53de9db0e21fc99 |
10-Feb-2011 |
Andreas Huber <andih@google.com> |
Fix the build.
Change-Id: I9b777ffb260eb0f3790ae0907e4a443d33fa3f2f
ndroid.mk
|
| 100a4408968b90e314526185d572c72ea4cc784a |
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
RTPAssembler.cpp
RTPConnection.cpp
RTPConnection.h
RTPSession.cpp
RTPSource.cpp
RTPSource.h
ndroid.mk
yHandler.h
|
| 783e5cd85d4bd40b1a04dfdfed256c5dcb2525cc |
28-Jan-2011 |
Andreas Huber <andih@google.com> |
More robust parsing of NPT time ranges in RTSP.
Change-Id: I3674501d2fd66aaface805c0a8678c74671a6dd3
related-to-bug: 3217210
SessionDescription.cpp
SessionDescription.h
yHandler.h
|
| 9202cca86e9017cc5ce30970c92a91ab32a0835e |
27-Jan-2011 |
Andreas Huber <andih@google.com> |
This particular RTSP server streams MPEG4-LATM audio with extra trailing bytes.
And now we're just ignoring them. Yay standards.
Change-Id: I76529ad8d585f143d6f99621ff671d179caf7b35
related-to-bug: 3353752
MPEG4AudioAssembler.cpp
|
| 21a6f9ffee8b3c014abfe165b8f7fd2224f49e1f |
18-Jan-2011 |
Andreas Huber <andih@google.com> |
Implement parsing of vbv buffering info in RTSP.
Change-Id: I7d871cafda2c4c65670a40ad9ab4f24317f8568a
related-to-bug: 3351915
PacketSource.cpp
|
| 934ca8cb1bcffcf1781a576ca625d2f901e5f0a9 |
12-Jan-2011 |
Andreas Huber <andih@google.com> |
Fail to parse duration instead of asserting, if the server response cannot be parsed.
Change-Id: I42324468edca5ccce29486059091da8e64f36326
related-to-bug: 3338518
SessionDescription.cpp
|
| 674ebd0b4e1143e38392a4e3bb38b4679a4577bc |
19-Nov-2010 |
James Dong <jdong@google.com> |
Removed uncessary FILE structure pointer for I/O
o also move the fd owner from caller to callee in the Writers
Change-Id: I510ccfdd0fcc58f1777fea4ed1349fd251852c65
RTPWriter.cpp
|
| fc9ac988e08a8b4c42e58999300265989f26f24c |
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
MPEG4AudioAssembler.cpp
MPEG4AudioAssembler.h
RTPSource.cpp
SessionDescription.cpp
|
| c21143636f2c6078c8ad6b096f69a9208591342b |
25-Oct-2010 |
Andreas Huber <andih@google.com> |
We don't have access to the md5 implementation on the simulator, let's disable digest authentication in rtsp for simulator targets.
Change-Id: I02a9b4af929601c899f04cee9864d0dd0716de62
RTSPConnection.cpp
|
| 4579b7d49f6dd4f37e6043e59debfd72d69b8e7b |
21-Oct-2010 |
Andreas Huber <andih@google.com> |
Support for BASIC and DIGEST authentication schemes in RTSP. Support for malformed packet descriptions that end lines in LF only, instead of CRLF.
Change-Id: I57eaefdc4b300a8f56bbe5cf3a34c424e8efe63a
related-to-bug: 3084183
RTSPConnection.cpp
RTSPConnection.h
SessionDescription.cpp
ndroid.mk
yHandler.h
|
| 8ac0cb9dc8a46f9b2badabc91cb5f7871e2215a9 |
18-Oct-2010 |
Jean-Baptiste Queru <jbq@google.com> |
Merge fb474872 from gingerbread-plus-aosp
Change-Id: I1bbb845a86a7b7df44ea175df3af22e5f47c44e3
|
| 56cfa2376ae87cba730ea7ce4a9e0ca4f0d07627 |
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
|
| a44501ea0896c2508bd6b807185d9049be6752f3 |
15-Oct-2010 |
Andreas Huber <andih@google.com> |
am 8e4f3c76: am 646e0d5a: Merge "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..." into gingerbread
Merge commit '8e4f3c76dd7f5596fa2fe10bdf008d4c02353160'
* commit '8e4f3c76dd7f5596fa2fe10bdf008d4c02353160':
Some webcams output rtp streams but never send any rtcp data in violation of
|
| f61551f4fc79e7da879802e3974afa9b03ffb5d0 |
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
|
| 43a2b3b5fd4e15ffed4235f348d5eba168e8432c |
12-Oct-2010 |
Andreas Huber <andih@google.com> |
am 5b0d0630: am 1010da2e: Merge "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." into gingerbread
Merge commit '5b0d063010b364102ffb7a533e2b76ecfd9636d5'
* commit '5b0d063010b364102ffb7a533e2b76ecfd9636d5':
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.
|
| 2bc940b4f961e588459c83862b2c6bea314a4027 |
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
|
| 250e051e564e3b6f5a88314379d5e145a2b5615f |
11-Oct-2010 |
Andreas Huber <andih@google.com> |
am cac43e8a: am beffefa2: Merge "RTSP seeking is now asynchronous, MediaPlayer is not notified that the seek is complete until it actually is. Ignore seek requests on live streams." into gingerbread
Merge commit 'cac43e8a2ce59c1151d5a2028330b2a769591d22'
* commit 'cac43e8a2ce59c1151d5a2028330b2a769591d22':
RTSP seeking is now asynchronous, MediaPlayer is not notified that the seek is complete until it actually is. Ignore seek requests on live streams.
|
| e31aa743466972764f9db5a88a713621ff0a29ae |
11-Oct-2010 |
Andreas Huber <andih@google.com> |
am e0c8545a: am 0fd4e216: Merge "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." into gingerbread
Merge commit 'e0c8545a2369881fe09582337a9de3db2db1a951'
* commit 'e0c8545a2369881fe09582337a9de3db2db1a951':
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.
|
| 1c8ef86f2c25272488c171f1469f996ebf335edc |
11-Oct-2010 |
Andreas Huber <andih@google.com> |
am 14ea1048: am c5912acc: Merge "Disable the access unit timeout temporarily while a seek operation is in progress." into gingerbread
Merge commit '14ea1048e7e8a4b40836b5601bc86b91663525cb'
* commit '14ea1048e7e8a4b40836b5601bc86b91663525cb':
Disable the access unit timeout temporarily while a seek operation is in progress.
|
| 0dcd837af4169bdb6fb2a0c384722dc4f57433c6 |
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
|
| c68a48c474f609df3eeb7d9738675d6ac8835e0a |
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
|
| a9d9dd2425c32f6868c35f49a3e8f29aafba931a |
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
|
| 3f94dacbd43b48bb629a79e45e738ead37c5debd |
22-Sep-2010 |
Andreas Huber <andih@google.com> |
am af909581: am 67738486: Merge "Remove stagefright foundation\'s incompatible logging interface and update callsites." into gingerbread
Merge commit 'af90958184fc5cfa1a4190e28bcfc4fdd4a5bcd6'
* commit 'af90958184fc5cfa1a4190e28bcfc4fdd4a5bcd6':
Remove stagefright foundation's incompatible logging interface and update callsites.
|
| 6e4c5c499999c04c2477b987f9e64f3ff2bf1a06 |
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
|
| ac5f724d00c8ac2040f01485873b6373f8994354 |
16-Sep-2010 |
Andreas Huber <andih@google.com> |
am 7ff94577: am 9909b948: Merge "Various fixes to improve resilience of the rtsp stack against spurious errors instead of asserting." into gingerbread
Merge commit '7ff945775210c60e6f113fb00903449cbb05c68a'
* commit '7ff945775210c60e6f113fb00903449cbb05c68a':
Various fixes to improve resilience of the rtsp stack against spurious errors instead of asserting.
|
| 6f85dba3768089679ff5e35ad2f1841918d0adb2 |
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
|
| 6faf0cd82346b23075d1f8b9f70f7af43f2c5f04 |
02-Sep-2010 |
Andreas Huber <andih@google.com> |
am fd0eed00: am a2511da9: Merge "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." into gingerbread
Merge commit 'fd0eed007d99178092ede56ec2c4799046615f70'
* commit 'fd0eed007d99178092ede56ec2c4799046615f70':
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.
|
| c9e894872c298b25fe9d74e68aa1e7287a541ac3 |
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
|
| 56f2c6e529bc62d55fc8baa7d1b52326307474d4 |
01-Sep-2010 |
Andreas Huber <andih@google.com> |
am 47f2cf62: am 412fc7cd: Merge "Keep gtalk video chat specific code consistent with rtsp changes." into gingerbread
Merge commit '47f2cf620731226a9311db0f864a4e1404e54b96'
* commit '47f2cf620731226a9311db0f864a4e1404e54b96':
Keep gtalk video chat specific code consistent with rtsp changes.
|
| 389636ce967af15e72817e2133907a2cb2efd1ae |
01-Sep-2010 |
Andreas Huber <andih@google.com> |
Keep gtalk video chat specific code consistent with rtsp changes.
Change-Id: I5f3f46c2150e16b26674432e427f79c04a69cd8e
RTPSession.cpp
|
| 3ef9f98aebb76018d2ee48ae4ac727a05efa63df |
01-Sep-2010 |
Andreas Huber <andih@google.com> |
am 6b52911c: am 48ac68e1: Merge "Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr." into gingerbread
Merge commit '6b52911cc7ba548fd3a240ca61eba510a8581e6f'
* commit '6b52911cc7ba548fd3a240ca61eba510a8581e6f':
Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr.
|
| 06124758ff402512f3c7a5fb2b35d8d09a0d6c2e |
31-Aug-2010 |
Andreas Huber <andih@google.com> |
Merge "Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr." into gingerbread
|
| 16c4e8c778d8518af4c0cbefadc5d5b1272c1762 |
31-Aug-2010 |
Andreas Huber <andih@google.com> |
am e1a3cddd: am 99fa510e: Merge "Better detection of connection problems - timeout if no rtcp packets arrive within a certain time, not a final frame (which may take longer)" into gingerbread
Merge commit 'e1a3cddd94749a42457a8f32cf21f663f07e4edf'
* commit 'e1a3cddd94749a42457a8f32cf21f663f07e4edf':
Better detection of connection problems - timeout if no rtcp packets arrive within a certain time, not a final frame (which may take longer)
|
| 4dba3e90f211eb5f5af19b10c5d3fc8c967b0086 |
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
|
| e7d3e90d8761f52a6acfdcd926f0392aca8ebb52 |
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
|
| ca999e0f936fc83f321e31ae13f93348d3f7454c |
31-Aug-2010 |
Andreas Huber <andih@google.com> |
am 03e83d4a: am 68ae91cb: 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
Merge commit '03e83d4ad909f5c07fb2011e03348a413453e909'
* commit '03e83d4ad909f5c07fb2011e03348a413453e909':
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.
|
| 5d5f5dfcc16756fe80a7c46cff0949fce9d54fe9 |
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
|
| b186054757f4743eb9a6d6e81d262b9c7b36bec7 |
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
|
| b62029edb6e0f97759ffb6d8f587267bee2dc31b |
31-Aug-2010 |
Andreas Huber <andih@google.com> |
am 987556bc: am abb8398e: Merge "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." into gingerbread
Merge commit '987556bc9bc1a61415b6e65bd600b8daf5b24d30'
* commit '987556bc9bc1a61415b6e65bd600b8daf5b24d30':
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.
|
| 7aef03379179c109c2547c33c410bfc93c8db576 |
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
|
| 9d876aca5ede85e6d9ccb82f11fae2834955c6f9 |
30-Aug-2010 |
Andreas Huber <andih@google.com> |
am 7ed9104c: am f6639c46: Finetune some rtsp timeout constants.
Merge commit '7ed9104c3acb172a480ebd7fd456fe69efd1ec3d'
* commit '7ed9104c3acb172a480ebd7fd456fe69efd1ec3d':
Finetune some rtsp timeout constants.
|
| c5c4286bebffa4c2a9539c8e09207c3130351531 |
30-Aug-2010 |
Andreas Huber <andih@google.com> |
am 6df6d606: am df992ac9: Merge "ALoopers can now be named (useful to distinguish threads)." into gingerbread
Merge commit '6df6d60681be9d524ce7fc07f2511008de424d27'
* commit '6df6d60681be9d524ce7fc07f2511008de424d27':
ALoopers can now be named (useful to distinguish threads).
|
| e56121bc4cb29c91d736eab181b1f51c4f125e78 |
30-Aug-2010 |
Andreas Huber <andih@google.com> |
Finetune some rtsp timeout constants.
Change-Id: Ice731c5097c2a2dee8a7f0cd45b547cd34f532c6
yHandler.h
|
| 9fbd6ae6b6d9f3eb791a3385df6fed3524531bd4 |
28-Aug-2010 |
Andreas Huber <andih@google.com> |
am 05c1cada: am 577615c9: Merge "Better support for rtsp (normal play-)time display. Better seek support, timeout if no packets arrive for too long." into gingerbread
Merge commit '05c1cadaeaf272a70acc889bfccd607648058470'
* commit '05c1cadaeaf272a70acc889bfccd607648058470':
Better support for rtsp (normal play-)time display. Better seek support, timeout if no packets arrive for too long.
|
| 437ab8c4b66a6c9dc47faa257df90089ebef10a9 |
28-Aug-2010 |
Andreas Huber <andih@google.com> |
am e25e0361: am e250c220: Merge "We accidentally always aborted after 10 secs, even if the connection was fine." into gingerbread
Merge commit 'e25e03612e1a2988ed83f24d2658cf0898fd1944'
* commit 'e25e03612e1a2988ed83f24d2658cf0898fd1944':
We accidentally always aborted after 10 secs, even if the connection was fine.
|
| a814c1fdc2acf0ed2ee3b175110f6039be7c4873 |
28-Aug-2010 |
Andreas Huber <andih@google.com> |
ALoopers can now be named (useful to distinguish threads).
Change-Id: Ieabaddb2e3a9e3a7a5bc36e55cd0721b60dbd50e
RTPWriter.cpp
yHandler.h
|
| 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
AMRAssembler.cpp
AVCAssembler.cpp
H263Assembler.cpp
MPEG4AudioAssembler.cpp
MPEG4ElementaryAssembler.cpp
PacketSource.cpp
PacketSource.h
RTPAssembler.cpp
RTPAssembler.h
RTSPController.cpp
yHandler.h
|
| cc6adf524c1bb3bfaa5be464b50b8bcca899761c |
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
|
| 7cb54d6f0e6c89f45e3db0bd9246f35836d67b8f |
27-Aug-2010 |
Andreas Huber <andih@google.com> |
am 74ae6973: am 17a765a1: Merge "Support for RTP packets arriving interleaved with RTSP responses." into gingerbread
Merge commit '74ae6973f8d5b7bc7bc4a7dcac5ddce90f382cd4'
* commit '74ae6973f8d5b7bc7bc4a7dcac5ddce90f382cd4':
Support for RTP packets arriving interleaved with RTSP responses.
|
| 0792ce7e0924ebb0dbe7b7cfcd79d12cbdb03ed2 |
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
|
| 5ac7b5def64625fdc9cfaf1bbdd013f5ada241f3 |
25-Aug-2010 |
Andreas Huber <andih@google.com> |
am 67ca90b3: am 6b6ae996: Merge "A first shot at proper support for seeking of rtsp streams." into gingerbread
Merge commit '67ca90b339feb8bb6889ca289a9dbc82c447b0d2'
* commit '67ca90b339feb8bb6889ca289a9dbc82c447b0d2':
A first shot at proper support for seeking of rtsp streams.
|
| cce326fe43411855aca2f719e505b051bc4b61b3 |
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
|
| d9734dc5f25730944ec4e62bb028092e1841e4a3 |
24-Aug-2010 |
Andreas Huber <andih@google.com> |
am 31e71131: am 3e22ef1e: Merge "Better handling of rtsp connection and disconnection." into gingerbread
Merge commit '31e71131049c943a388134e796087e109248efcc'
* commit '31e71131049c943a388134e796087e109248efcc':
Better handling of rtsp connection and disconnection.
|
| 1b543242102ef3c28145c6ad50ee8e8ce2fb26d3 |
23-Aug-2010 |
Andreas Huber <andih@google.com> |
Better handling of rtsp connection and disconnection.
Change-Id: Ib126af6c14c5a212a51a5ee3c4a0a7d1860ad167
RTSPController.cpp
yHandler.h
|
| 263ebfd8a17266eedc84eb879edb6a6a3395f760 |
21-Aug-2010 |
James Dong <jdong@google.com> |
am c8d2fa70: am cbd038fe: Merge "Make MediaWriter stop and pause return errors if necessary" into gingerbread
Merge commit 'c8d2fa704abebdbf0bd8aac185216dc068950217'
* commit 'c8d2fa704abebdbf0bd8aac185216dc068950217':
Make MediaWriter stop and pause return errors if necessary
|
| 9934d0cf66861d331adcad28dc4713874e607a76 |
21-Aug-2010 |
Andreas Huber <andih@google.com> |
am 873ebfb8: am 223e4f73: Merge "Support for MP4V-ES packetization format according to RFC3016." into gingerbread
Merge commit '873ebfb825cb498d9ff3012d1d31b02e31a79980'
* commit '873ebfb825cb498d9ff3012d1d31b02e31a79980':
Support for MP4V-ES packetization format according to RFC3016.
|
| 9b92412737095ab6a06f01a0c6daaebb79dffb55 |
21-Aug-2010 |
Andreas Huber <andih@google.com> |
am b29ebd39: am f0ad5484: Merge "In the absence of width/height information in the sdp, extract the dimensions from the avc codec specific data." into gingerbread
Merge commit 'b29ebd397e25a7176bcc1c81980f17b0190ebe7f'
* commit 'b29ebd397e25a7176bcc1c81980f17b0190ebe7f':
In the absence of width/height information in the sdp, extract the dimensions from the avc codec specific data.
|
| 91d113e8daa9d71c4ea8afd595a3921e03787cbf |
21-Aug-2010 |
Andreas Huber <andih@google.com> |
am 6bcffcd2: am 8c192fe9: Merge "Better support for rtsp streamed through VLC. Temporarily make the socket blocking to read all of the session description." into gingerbread
Merge commit '6bcffcd2dc410db780c152c70a01b22da6ca58be'
* commit '6bcffcd2dc410db780c152c70a01b22da6ca58be':
Better support for rtsp streamed through VLC. Temporarily make the socket blocking to read all of the session description.
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| e0b77ce97ef84c47ae408e92f2afb7509a5051b6 |
19-Aug-2010 |
James Dong <jdong@google.com> |
Merge "Make MediaWriter stop and pause return errors if necessary" into gingerbread
|
| 37187916a486504acaf83bea30147eb5fbf46ae5 |
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
|
| 62cb04d23642a2ea7c005f050494c8ef3c370dd3 |
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
|
| 85f12e9b9062402d6110df3f7099707912040edb |
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
|
| ef7af7fec702db2fde72b16dedf9064585e6db77 |
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
|
| cc760e477378117ef34fb2833d0b6521925b38ad |
12-Aug-2010 |
Andreas Huber <andih@google.com> |
am 3bf8c342: am ae3a1f45: Merge "Fix the h.263 assembler to properly subset a buffer\'s range if it already has a range applied." into gingerbread
Merge commit '3bf8c3427f4c728bb88e5e266b85c96e3e727203'
* commit '3bf8c3427f4c728bb88e5e266b85c96e3e727203':
Fix the h.263 assembler to properly subset a buffer's range if it already has a range applied.
|
| db3a7e67a82b48b9b7e2bfa639fc117f75682a76 |
12-Aug-2010 |
Andreas Huber <andih@google.com> |
am 53895c6a: am 66aa0f3d: Merge "APacketSource is too verbose." into gingerbread
Merge commit '53895c6a0e8ecb4e835aab7eca7480779c224356'
* commit '53895c6a0e8ecb4e835aab7eca7480779c224356':
APacketSource is too verbose.
|
| a6238a1e5b603ca2ccf3b2297c9bc8a141cf8559 |
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
|
| 6dc387a8c3f031f9f17d1138295368946563f7a5 |
12-Aug-2010 |
Andreas Huber <andih@google.com> |
APacketSource is too verbose.
Change-Id: I48ca7b070d89e43405d05e5f41e650db587e12b4
PacketSource.cpp
|
| 5d8e9cd46d21d8cddebe82831b99927363fa896a |
10-Aug-2010 |
Andreas Huber <andih@google.com> |
am 4dc41bb4: am 18f0174f: Merge "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." into gingerbrea
Merge commit '4dc41bb445860cfcb8c0dfbecdc8f0f5f15f5e28'
* commit '4dc41bb445860cfcb8c0dfbecdc8f0f5f15f5e28':
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.
|
| f8ca90452ff3e252f20de38f1c3eee524c808c3e |
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
|
| c16acb7a9467831caf2c7c268a3fe57ec4bc69aa |
05-Aug-2010 |
Andreas Huber <andih@google.com> |
am 870678a9: am 2c37f3d3: Merge "Better support for fake timestamps in RTP, H.263 video now also requests FIR." into gingerbread
Merge commit '870678a954e1e2a96caf76453c20de808253ffd1'
* commit '870678a954e1e2a96caf76453c20de808253ffd1':
Better support for fake timestamps in RTP, H.263 video now also requests FIR.
|
| b6b546e72818988865d508e380d4445da71c4503 |
05-Aug-2010 |
Andreas Huber <andih@google.com> |
am c6d1519e: am fb861523: Merge "Specification of codec specific data as part of the session description is now optional." into gingerbread
Merge commit 'c6d1519e549740abd56df7a98b5348bd9095ae46'
* commit 'c6d1519e549740abd56df7a98b5348bd9095ae46':
Specification of codec specific data as part of the session description is now optional.
|
| 982a93173bc84f005172152d823cbb59dfcbeb12 |
05-Aug-2010 |
Andreas Huber <andih@google.com> |
am 1f513d88: am c17f35dd: Merge "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." into gingerbread
Merge commit '1f513d8821670a33d6361ea521b6756163a3f9bf'
* commit '1f513d8821670a33d6361ea521b6756163a3f9bf':
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.
|
| ff53123821a3ec2e71fdb1a971ea2cbae3119826 |
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
|
| 33a8457868eb00b94b37b53321a80d9307202a9d |
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
|
| 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
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
|
| f661058d77d1484e5911d1962f8e1e8466240687 |
22-Jul-2010 |
Andreas Huber <andih@google.com> |
am b72d3180: am 81046c8c: Merge "Various changes to improve rtsp networking, reduce packet loss and adapt to ALooper API changes." into gingerbread
Merge commit 'b72d3180dc8d41d6269664bea808b04410bbe40f'
* commit 'b72d3180dc8d41d6269664bea808b04410bbe40f':
Various changes to improve rtsp networking, reduce packet loss and adapt to ALooper API changes.
|
| 348a8eab84f4bba76c04ca83b2f5418467aa1a48 |
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
|
| 4e2ffa400b82559cab2c5717c8dcdff393d334a9 |
15-Jul-2010 |
Mike Lockwood <lockwood@android.com> |
Fixes for simulator build on lucid
strchr and strrchr now return const char* instead of char*
Change-Id: I5ca831b8951af7e6306eb9d9d6f78ed2ec13d649
Signed-off-by: Mike Lockwood <lockwood@android.com>
RTSPConnection.cpp
SessionDescription.cpp
yHandler.h
|
| cf7b9c7aae758ac0b99833915053c63c2ac46e09 |
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
|