bug:458435
This will temporarily start a thread for answering calls, we are
going to add a handler thread to handle this soon.
Change-Id: I9079038d671e1b1631c6e663fc2c3de297d97428
commit c80992e419ed567abef451042f09c4958534b90d
Author: Andreas Huber <andih@google.com>
Date: Wed May 11 14:00:07 2011 -0700
Support for the mp3 audio decoder as a software OMX component.
Change-Id: I66e10c4d0be4c3aecdef1c21b15a2c7359ceb807
commit a358d0e1bf2a88897887445f42ccdda0f5f2f528
Author: Andreas Huber <andih@google.com>
Date: Wed May 11 13:11:23 2011 -0700
Support for G.711 alaw and mulaw decoders as software OMX components
Change-Id: Ia5c76c02cb83a9f94ce39a27b2251e5880218f03
commit 79088b9c9a5c8b8c97ea66cb4f90a2b0f0d34553
Author: Andreas Huber <andih@google.com>
Date: Thu May 5 15:43:32 2011 -0700
Instead of using an RGB surface and conversion yuv420->rgb565
convert from OMX_COLOR_FormatYUV420Planar to HAL_PIXEL_FORMAT_YV12 instead.
Change-Id: I8c4fc3c54c963f0d4ba6377f3c4ab4e0013152e5
related-to-bug: 4394005
commit 69469d3bd84425777b11b9fc938c5e0c61af26a7
Author: Andreas Huber <andih@google.com>
Date: Tue May 10 15:46:42 2011 -0700
voip mustn't link against libstagefright.so
Change-Id: I4d0ba9a8b9dc9380b792a1bd04bcda231964862c
commit 2a9a9eeeeeb36ae3a9e680469c3016d509ff08c3
Author: Andreas Huber <andih@google.com>
Date: Tue May 10 14:37:10 2011 -0700
Remove most non-OMX software decoders by default
Change-Id: Ic56514bc1b56b8fa952e8c4a164ea7379ecb69d0
commit a4de62c37b335c318217765403a9fb282b20a216
Author: Andreas Huber <andih@google.com>
Date: Mon May 9 16:50:02 2011 -0700
Conditionally build the old-style software decoders.
Change-Id: I5de609e1d76c92d26d6eb81d1551462258f3f15f
commit 5d8b039f9449dc3dad1e77c42c80cc0b54b0c846
Author: Andreas Huber <andih@google.com>
Date: Mon May 9 16:13:12 2011 -0700
Support for MPEG4 and H.263 video decoders as soft OMX components.
Change-Id: I5e3a4835afab89f98e3aa128d013628f5830eafe
commit b25a1bfbeb0ff6e62e1cc694ce2599c91489c7d0
Author: Andreas Huber <andih@google.com>
Date: Mon May 9 11:49:10 2011 -0700
Boost Soft OMX thread priority, fix timestamp handling in vorbis Soft OMX decoder.
Change-Id: I68d26d4999f06fcc451d69e5303663fab0cba9e8
commit c0574362f8dc3319ce84d981097867062a698527
Author: Andreas Huber <andih@google.com>
Date: Mon May 9 11:28:53 2011 -0700
Support for the AMR decoders (NB and WB) as Soft OMX components.
Change-Id: Ia565f59833fb52653e23f26536e7e41fc329a754
commit 3e5575a8f0e27a490cb7bde77bd9456087837f08
Author: Andreas Huber <andih@google.com>
Date: Wed May 4 13:41:25 2011 -0700
Signal an error if the aac decoder failed to initialize from codec specific data.
Change-Id: I01da7831bdf722edd7d6dc5974486daa2cf2b209
related-to-bug: 4272179
commit f94aeaa9886e772ff4823e671ed237096649f4af
Author: Andreas Huber <andih@google.com>
Date: Tue May 3 13:07:38 2011 -0700
Software OMX nodes don't (yet?) support native_window mode.
Change-Id: I7d9ca9164ef4abf66b573ca21dba12d672f8b12d
commit eefdfabac8dc659e00daa56da69aea705c49cb67
Author: Andreas Huber <andih@google.com>
Date: Tue May 3 12:57:16 2011 -0700
Fixing the OMX tests to refer to appropriate files from test content.
Change-Id: I5b61c3498749bfb876abbd3946a5132356e3f6ff
commit f31b7326aef14b6a1b7946520a9688f092e844d5
Author: Andreas Huber <andih@google.com>
Date: Tue May 3 11:08:38 2011 -0700
Soft OMX components are now dynamiclly loaded/unloaded, not directly linked against.
Change-Id: I1e2ecfbfab67a8869886f738eaf0c7b3c948b6d9
commit b7f0343879e4df06f0a1c9bfece24df557954e2f
Author: Andreas Huber <andih@google.com>
Date: Mon May 2 15:58:36 2011 -0700
Support for the AVC software decoder as an OMX component.
Change-Id: I13c12df435ba4afbd968a9fc659f66b91c818bc2
commit 5bb9e616d6c8e1b13d531fe996b9a9affdfb2977
Author: Andreas Huber <andih@google.com>
Date: Fri Apr 29 12:05:37 2011 -0700
Fix Vorbis OMX decoder's component role.
Change-Id: I5e871e5e11b3f951c93590210e63fd7987c467b5
commit 089c91f2333062e196c7afd5fb0ca914878aa474
Author: Andreas Huber <andih@google.com>
Date: Fri Apr 29 12:05:18 2011 -0700
Support vorbis_decoder OMX testing.
Change-Id: I1985be178a12ae3f8768bc72067d9236238be170
commit 56e241fa36fc37219bc536b823bdc2ab82dc1fad
Author: Andreas Huber <andih@google.com>
Date: Fri Apr 29 12:01:46 2011 -0700
SoftVorbis OMX component now respects the number of valid frames per page.
Change-Id: I82a117a064d9b083fc58a54ad900a987a763ef03
commit fcd618ec520c376fdb78f4cbb44b8d9f5d213e2b
Author: Andreas Huber <andih@google.com>
Date: Fri Apr 29 10:59:38 2011 -0700
Support for the vorbis audio decoder as a soft OMX component.
Change-Id: Iaeb057e58ca306d3dce205c0445b74d5aefef492
commit d1fcc3203fc8003ad79c6e96b3a1fc4261743f16
Author: Andreas Huber <andih@google.com>
Date: Fri Apr 29 10:07:50 2011 -0700
VPX decoder now properly resizes buffers after a port settings change.
Change-Id: I110749a31b6cba087891d8e5dfe420830bdbf831
commit 35c7168243cb69849d88911144a2c7fdfed5c54e
Author: Andreas Huber <andih@google.com>
Date: Thu Apr 28 13:23:34 2011 -0700
Support for the VPX video decoder as a Software OMX component.
Change-Id: Ic345add2d6d768d4af631160153f2e9b97fcea71
commit 923b2534b4211fc5405377b5190bfa6f2dd27f32
Author: Andreas Huber <andih@google.com>
Date: Thu Apr 28 11:34:40 2011 -0700
Table-based registration of soft omx components.
Change-Id: I7f45f0fa5b3a7950776e69c66349731f7674e937
commit 04a88f3edb2266a463da9c4481b80178be460902
Author: Andreas Huber <andih@google.com>
Date: Thu Apr 28 11:22:31 2011 -0700
Apparently OMX_GetParameter is valid in any state other than OMX_StateInvalid
OMX_SetParameter is still constrained to OMX_StateLoaded or a disabled port.
Change-Id: I1032d7cf4011982d306aa369d4158a82830d26fb
commit 9d70ca68445e7c40f5c9b2d12466e468f514de88
Author: Andreas Huber <andih@google.com>
Date: Wed Apr 27 15:03:18 2011 -0700
Use the new soft OMX aac decoder for HTTP live playback.
Change-Id: Ifbcfb732a9edb855cb46b49f6d0ac942170ee28f
commit 213fe4a10ea93cce08e8622dc3908053f29878a1
Author: Andreas Huber <andih@google.com>
Date: Tue Apr 12 16:39:45 2011 -0700
Foundation for supporting software decoders as OMX components
Change-Id: I7fdab256563b35d1d090617abaea9a26b198d816
Change-Id: I83e9236beed4af985d10333c203f065df9e09a42
Herring board exhibits a strong echo even in non speakerphone modes.
To compensate the lack of AEC or AES when not in speakerphone, the mic gain
had been reduced in the ADC. But this has an adverse effect on other VoIP applications
that have their own AEC and are penalized by the weak mic gain.
This workaround enables an acceptable mic gain for other VoIP apps while offering a
SIP call experience which is not worse than it was with the residual echo that was
present even with mic gain reduction.
Change-Id: I33fd37858758e94e42ef5b545d3f0dc233220bf1
Since String.replaceFirst uses regex and since SIP user names are
allowed to include regex charaters such as '+', the code must
fist convert the string to a literal pattern String before using
replaceFirst method.
Change-Id: I25eac852bd620724ca1c5b2befc023af9dae3c1a
This change unhides RTP related classes including AudioCodec,
AudioGroup, AudioStream, and RtpStream. This allows developers
to control audio streams directly and also makes conference
calls possible with the combination of the public SIP APIs.
Change-Id: Idfd4edf65a1cbf3245ec2786fbc03b06438b0fb3
Cherripick from master CL 79833, 79417, 78864, 80332, 87500
Add new audio mode and recording source for audio communications
other than telelphony.
The audio mode MODE_IN_CALL signals the system the device a phone
call is currently underway. There was no way for audio video
chat or VoIP applications to signal a call is underway, but not
using the telephony resources. This change introduces a new mode
to address this. Changes in other parts of the system (java
and native) are required to take this new mode into account.
The generic AudioPolicyManager is updated to not use its phone
state variable directly, but to use two new convenience methods,
isInCall() and isStateInCall(int) instead.
Add a recording source used to designate a recording stream for
voice communications such as VoIP.
Update the platform-independent audio policy manager to pass the
nature of the audio recording source to the audio policy client
interface through the AudioPolicyClientInterface::setParameters()
method.
SIP calls should set the audio mode to MODE_IN_COMMUNICATION,
Audio mode MODE_IN_CALL is reserved for telephony.
SIP: Enable built-in echo canceler if available.
1. Always initialize AudioRecord with VOICE_COMMUNICATION.
2. If echo canceler is available, disable our echo suppressor.
Note that this CL is intentionally not correcting the
getAudioSourceMax() return value in MediaRecorder.java as the
new source is hidden here.
Change-Id: Ie68cd03c50553101aa2ad838fe9459b2cf151bc8
The previous implementation registers receivers when SipService starts up.
If the user doesn't use SIP at all, SipService will still process connecivity
and wifi state change events, which involves holding wake lock and thus
consumes power unnecessarily.
With this CL, SipService is completely idle if the user doesn't use SIP at all.
It registers receivers only when at least one account is opened.
Bug: 3326998
Change-Id: Idea43747f8204b0ccad3fc05a1b1c0b29c9b2557
The previous implementation registers receivers when SipService starts up.
If the user doesn't use SIP at all, SipService will still process connecivity
and wifi state change events, which involves holding wake lock and thus
consumes power unnecessarily.
With this CL, SipService is completely idle if the user doesn't use SIP at all.
It registers receivers only when at least one account is opened.
Bug: 3326998
Change-Id: Ib70e0cf2c808e0ebab4c3c43dcab5532d24e5eeb
Originally a stream does not send packets when it is receive-only or there is
nothing to mix. However, this causes some problems with certain firewalls and
proxies. A firewall might remove a port mapping when there is no outgoing
packet for a preiod of time, and a proxy might wait for incoming packets from
both sides before start forwarding. To solve these problems, we send out a
silence packet on the stream for every second. It should be good enough to
keep the stream alive with relatively low resources.
Bug: 3119690
Change-Id: Ib9c55e5dddfba28928bd9b376832b68bda24c0e4
This is to make SipManager.isVoipSupported() effective.
Also add NPE check now that we may return null SipAudioCall when VOIP is not
supported.
Bug: 3251016
Change-Id: Icd551123499f55eef190743b90980922893c4a13
SipURI returns port -1 when port is not present in the URI.
Don't call SipProfile.Builder.setPort() when that happens.
Bug: 3291248
Change-Id: I8e608cbc56ea82862df55fdba885f6a864db83ab
SipURI returns port -1 when port is not present in the URI.
Don't call SipProfile.Builder.setPort() when that happens.
Change-Id: Ic5fe7301195705a77010038cae20d6629b33135e
+ Also fix race between ending and changing (holding/unholding) a SIP call.
+ Remove an unused method.
Bug : 3128233
Change-Id: Ie18d8333a88f0d9906d54988243d909b58e07e4b
