| fb3ec448f1208d75edebff0e93fa97a8913ff29e |
06-Sep-2012 |
Jean-Baptiste Queru <jbq@google.com> |
am f66603e1: am 3f3ce8ad: am bf3e62b8: Merge "Make SimpleSessionDescription locale safe"
* commit 'f66603e109d439e3a537cd3804706609ce86970e':
Make SimpleSessionDescription locale safe
|
| 7314532349e402315af9b8f664432dd18292421f |
29-Aug-2012 |
Johan Redestig <johan.redestig@sonymobile.com> |
Make SimpleSessionDescription locale safe
Explicitly use Locale.US in SimpleSessionDescription to avoid
unexpected results in some locales.
Change-Id: Idb4a36a9e332d302e1b9b940355917c0f738e076
ava/android/net/sip/SimpleSessionDescription.java
|
| 845b4712f0e4d7ca802f21fba1adae0b1d0712e7 |
02-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Use audio_channel_mask_t more consistently
In AudioRecord::getMinFrameCount() and AudioSystem::getInputBufferSize(),
input parameter is channel mask instead of channel count.
Change-Id: I22a1c492113f3e689173c5ab97b2567cff3abe2b
ni/rtp/AudioGroup.cpp
|
| 9be0105fbc56eb1b1813bb7c5fe258a144867a43 |
22-Jun-2012 |
Scott Main <smain@google.com> |
docs: fix several links
Change-Id: I89d9fd64dc22c90680bb05415cc966c255165af9
ava/android/net/sip/package.html
|
| 870085cce0fde0a44f0c28a5097b8c827cafb0e2 |
04-Apr-2012 |
Chia-chi Yeh <chiachi@android.com> |
Merge "SIP: push the logic of finding local address down to SipSessionGroup."
|
| 2dd9134c6b2da47f4816bc0f72a0e4924aca4f84 |
04-Apr-2012 |
Chia-chi Yeh <chiachi@android.com> |
SIP: push the logic of finding local address down to SipSessionGroup.
This allows different accounts binding on different IP addresses,
such as one on IPv4 and another on IPv6.
Bug: 4169057
Change-Id: I0bb36669394c281330091673fa338adea8f782cd
ava/com/android/server/sip/SipService.java
ava/com/android/server/sip/SipSessionGroup.java
|
| e66950506c473e660f2e5762d7a71e13808be387 |
30-Mar-2012 |
Chia-chi Yeh <chiachi@android.com> |
RTP: refactor a little bit and fix few minor bugs.
Change-Id: I063644507f26996ded462972afcb550a4528dac8
ava/android/net/rtp/AudioGroup.java
ava/android/net/rtp/AudioStream.java
ava/android/net/rtp/RtpStream.java
ni/rtp/AudioGroup.cpp
ni/rtp/RtpStream.cpp
|
| ba4d0433319393d626d2169683209e4956a087e2 |
28-Mar-2012 |
James Dong <jdong@google.com> |
frameworks base Android.mk file changes
Change-Id: I7459b9e959a60751b8fa6e0d893cb2c820c064ce
ni/rtp/Android.mk
|
| fb982db41060a2914cddb43200f3ee53627f8762 |
28-Mar-2012 |
Chia-chi Yeh <chiachi@android.com> |
RTP: add a null-check in AudioStream.setDtmfType().
Change-Id: I52cbdea48affae3747942940451f4fd5ca47030f
ava/android/net/rtp/AudioStream.java
|
| 32d72b2f538673466f6e0ebf01886412e803dc4f |
21-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Whitespace"
|
| f743e1f6abdb018fc58c467cdf35cbb8b81cf8c4 |
14-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Whitespace
Fix indentation, and add blank lines in key places for clarity
Change-Id: I57a0a8142394f83203161aa9b8aa9276abf3ed7c
ni/rtp/AudioGroup.cpp
|
| ae75f994cc50837afe79d3bfbc576811e3602fef |
16-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Add libmedia_native
Change-Id: Ib8cff8abd73723b793f08da99ad59549f219e0e7
ni/rtp/Android.mk
|
| 4e42c5f41fdb67ec62fbecc48038c8fe30b57bcd |
13-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Remove dependency on audio_* location
Change-Id: I4bc66115fcb9ba22b057bd72db3f561dcb18a0d8
ni/rtp/Android.mk
|
| 597f8282ee1b86ba8f7384eb3060bac3b3f7cf92 |
12-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Fix build warnings
Change-Id: I543e730aff2d03c18c26b116c9fe9419259808af
ni/rtp/AudioGroup.cpp
|
| 3762c311729fe9f3af085c14c5c1fb471d994c03 |
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
ni/rtp/AudioGroup.cpp
ni/rtp/RtpStream.cpp
|
| 8564c8da817a845353d213acd8636b76f567b234 |
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
ni/rtp/AudioGroup.cpp
|
| 6215d3ff4b5dfa52a5d8b9a42e343051f31066a5 |
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
ni/rtp/EchoSuppressor.cpp
|
| 5baa3a62a97544669fba6d65a11c07f252e654dd |
20-Dec-2011 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGD(_IF) to (IF_)ALOGD(_IF) DO NOT MERGE
See https://android-git.corp.google.com/g/156016
Bug: 5449033
Change-Id: I4c4e33bb9df3e39e11cd985e193e6fbab4635298
ni/rtp/AudioGroup.cpp
|
| c2bd6162eddad0cdfdafc037142e043680ffa705 |
28-Dec-2011 |
Chia-chi Yeh <chiachi@android.com> |
SipService: grab Wi-Fi lock only when necessary.
Change-Id: Ie432049156e70b6748426b959b653f21bfc504a1
ava/com/android/server/sip/SipService.java
|
| 2b072677538de979961b5bf527109fdab1713731 |
23-Dec-2011 |
Joe Fernandez <joefernandez@google.com> |
am 201469f5: am bb7f590a: Merge "docs: Add developer guide cross-references, Project ACRE, round 4" into ics-mr1
* commit '201469f54522436be79d4d6665721049bfc74320':
docs: Add developer guide cross-references, Project ACRE, round 4
|
| 3aef8e1d1b2f0b87d470bcccf37ba4ebb6560c45 |
20-Dec-2011 |
Joe Fernandez <joefernandez@google.com> |
docs: Add developer guide cross-references, Project ACRE, round 4
Change-Id: I1b43414aaec8ea217b39a0d780c80a25409d0991
ava/android/net/sip/SipAudioCall.java
ava/android/net/sip/SipManager.java
ava/android/net/sip/SipProfile.java
ava/android/net/sip/package.html
|
| 98c54a3e06c3e1909fa0f6cb19c5201f116ed5bc |
22-Nov-2011 |
Chia-chi Yeh <chiachi@android.com> |
am 0e67685e: am bfd85f9a: Merge "SIP: turn off verbose logs." into ics-mr1
* commit '0e67685e53cbcf7682a0364cf34fd3ac0632596f':
SIP: turn off verbose logs.
|
| cb6ee06f62c20ae036a206667097f20b837b11ab |
19-Nov-2011 |
Chia-chi Yeh <chiachi@android.com> |
SIP: turn off verbose logs.
Bug: 5616713
Change-Id: Iaf2e6878731d10d7f4f2a7cd8af71f4517780642
ava/com/android/server/sip/SipHelper.java
ava/com/android/server/sip/SipService.java
ava/com/android/server/sip/SipSessionGroup.java
ava/com/android/server/sip/SipWakeLock.java
|
| 71f2cf116aab893e224056c38ab146bd1538dd3e |
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
ni/rtp/AudioGroup.cpp
|
| ee59e6a9fc69241b286acb7b55a22b8393c81222 |
28-Sep-2011 |
Chia-chi Yeh <chiachi@android.com> |
SipService: handle connectivity changes correctly.
This patch assumes that for the same network type,
there MUST be a DISCONNECTED between two CONNECTEDs.
Also removes the Wi-Fi scanning since the framework
already handles this when a WifiLock is held.
Bug: 4283795
Change-Id: I08481e70c651cffcbb516c8cc6584c919078fa4f
ava/com/android/server/sip/SipService.java
|
| 6d8b9b84ac83acfc193fd633ba961168867124fa |
14-Sep-2011 |
Chia-chi Yeh <chiachi@android.com> |
Merge "RTP: Update parameters for larger packet intervals."
|
| 7a685e89114ddfe35f87075dfe66a480c91c9de2 |
13-Sep-2011 |
Chia-chi Yeh <chiachi@android.com> |
Merge "SIP: fix keep-alive measurement and increase the timeout."
|
| d17b6d526648c372be761097e55c19767d5dba7d |
09-Sep-2011 |
Chia-chi Yeh <chiachi@android.com> |
SIP: fix keep-alive measurement and increase the timeout.
Bug: 5226511
Change-Id: I1283790581496b1ff4e583a8d9379cdc39f78c20
ava/com/android/server/sip/SipService.java
ava/com/android/server/sip/SipSessionGroup.java
|
| be57bfe853d07369f429b600039ea474b9ea5e31 |
07-Sep-2011 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Update parameters for larger packet intervals.
Also remove some duplicated code.
Change-Id: I64576e5442a962eb4b0dfa83b52a8127567ba597
ni/rtp/AudioGroup.cpp
|
| 81a5ec5b94d889656cc2f212102c441b91b2e3c0 |
08-Sep-2011 |
Chia-chi Yeh <chiachi@android.com> |
Merge "RTP: support payloads with larger packetization interval."
|
| fa6067f87c7405c987d5879554d529e7413910c0 |
06-Sep-2011 |
Eric Laurent <elaurent@google.com> |
Merge "VoIP JNI: Force AEC on for tuna board"
|
| 35d05dcba1e829782813b6ec21afceb5cffc22e6 |
06-Sep-2011 |
Chia-chi Yeh <chiachi@android.com> |
RTP: support payloads with larger packetization interval.
RFC 3551 section 4.2 said that a receiver should accept packets
representing between 0 and 200ms of audio data. Now we add the
ability to decode multiple frames in a payload as long as the
jitter buffer is not full. This change covers G711, GSM, and
GSM-EFR. AMR will be added later.
Bug: 3029736
Change-Id: Ifd194596766d14f02177925c58432cd620e44dd7
ni/rtp/AmrCodec.cpp
ni/rtp/AudioCodec.h
ni/rtp/AudioGroup.cpp
ni/rtp/G711Codec.cpp
ni/rtp/GsmCodec.cpp
|
| 54eabd6c929c6f56da28421839b0ef2945cda876 |
06-Sep-2011 |
Chia-chi Yeh <chiachi@android.com> |
SIP: avoid extreme small values in Min-Expires headers.
If the expiry time cannot be found in Contact header or Expires
header, use the default value of 3600 seconds, which is specified
in RFC 3261.
Change-Id: I2607a398b96743614b01713cfd9b28f40386fac1
ava/com/android/server/sip/SipSessionGroup.java
|
| 74e0a990ae3196b8195db2a399c22516c7dd0823 |
29-Aug-2011 |
Eric Laurent <elaurent@google.com> |
VoIP JNI: Force AEC on for tuna board
Force AEC on for tuna board because of the strong feedback
of Rx audio path, even when playing over earpiece or headset.
Change-Id: I9c14257d56103ba82d6cdb0b7d5a3f315638136e
ni/rtp/AudioGroup.cpp
|
| 5f760064e1975a50e4abb63e560731c8b2c7b56c |
30-Aug-2011 |
Chia-chi Yeh <chiachi@android.com> |
SIP: add the check for expiry time in Contact header.
There can be three expiry times in the same message header. We
choose the smaller value in Expires header and Contact header,
and then we obey the value defined in Min-Expires header. If
none of them is set, the default value is used.
Bug: 5178284
Change-Id: Ie9d4a48c93863e82e5197bb4a0db3f4fec56857c
ava/com/android/server/sip/SipSessionGroup.java
|
| dc5bbe965f7a66238c3f9a6694f4410b3d52af3b |
14-Aug-2011 |
Hung-ying Tyan <tyanh@google.com> |
Handle SIP authentication response for BYE.
Bug: 5159669
Change-Id: I029684334500d4d0db176783084c9b7d1db87e40
ava/android/net/sip/SipSession.java
ava/com/android/server/sip/SipSessionGroup.java
|
| 53ad2c7fe212a08ae05fb4d7f27d42f9a0a4b912 |
02-Aug-2011 |
Conley Owens <cco3@android.com> |
am 0793586b: am f8c1f129: am e1d27154: am f87743e7: Merge "Prevent NullPointerException cases while using SipService."
* commit '0793586bf8f4dce71d0b4d7ff2f212129b3f76fe':
Prevent NullPointerException cases while using SipService.
|
| 0793586bf8f4dce71d0b4d7ff2f212129b3f76fe |
02-Aug-2011 |
Conley Owens <cco3@android.com> |
am f8c1f129: am e1d27154: am f87743e7: Merge "Prevent NullPointerException cases while using SipService."
* commit 'f8c1f1298ac3ede518c8d29eeb6719746c6afaf0':
Prevent NullPointerException cases while using SipService.
|
| 25ccbb97ffd3298caede635f29445073e845cfc3 |
28-Jul-2011 |
Masahiko Endo <masahiko.endo@gmail.com> |
Prevent NullPointerException cases while using SipService.
Some SipService methods may return null, in such cases like no Wi-Fi
connection. Added minimum check to prevent NullPointerExceptions.
Change-Id: Ia7fae57ee893f2564cbfdedb6dc614938ab60ff7
Signed-off-by: Masahiko Endo <masahiko.endo@gmail.com>
ava/android/net/sip/SipManager.java
|
| 5fb3ba60afe68060ac1ed291f4a108fef8c622c3 |
25-Jul-2011 |
Eric Laurent <elaurent@google.com> |
Issue 3370834: No Echo canceler for SIP
Added detection of platfrom AEC in AudioGroup. If an AEC
is present, the SIP stack will use it, otherwise the echo suppressor
of the stack will be used.
Change-Id: I4aa45a8868466120f5f9fae71b491fe4ae1162c2
ni/rtp/Android.mk
ni/rtp/AudioGroup.cpp
|
| 307f15faafa5a38d9b3b314df22778cd11685d7b |
12-Jul-2011 |
repo sync <cywang@google.com> |
Add REFER handling.
Handle REFER requests including REFER with Replaces header.
bug:4958680
Change-Id: I96df95097b78bed67ab8abd309a1e57a45c6bc2f
ava/android/net/sip/SipAudioCall.java
ava/com/android/server/sip/SipHelper.java
ava/com/android/server/sip/SipSessionGroup.java
|
| 3eeb1a98f91dff53eaf00cfb6b6ee8f25917b232 |
05-Jul-2011 |
Hung-ying Tyan <tyanh@google.com> |
Merge "Keep last known keepalive interval to avoid duplicate effort."
|
| 9324e04dcf4d5dd4dd08b1a4d7d981e259df3fe0 |
04-Jul-2011 |
Hung-ying Tyan <tyanh@google.com> |
Merge "Do not hold wifi lock when SIP is also available over mobile network."
|
| f8c34ad3efd82974f166419b174431564658a7d0 |
04-Jul-2011 |
Hung-ying Tyan <tyanh@google.com> |
Merge "Do not keep alive for re-established call."
|
| 9edfa107575b5905c9ae0a2fa0d6f0cc19595300 |
01-Jul-2011 |
Hung-ying Tyan <tyanh@google.com> |
Keep last known keepalive interval to avoid duplicate effort.
The current implementation always starts with default minimum interval when
the measurement process starts. By keeping last known good interval, we can
save the time in re-measurement.
Change-Id: I8f1720acafaa7e101855fe0c66d5c7b0e578e0d7
ava/com/android/server/sip/SipService.java
|
| 8ba4566c01c5848b378d1d86e9041730f5b5a13f |
01-Jul-2011 |
Hung-ying Tyan <tyanh@google.com> |
Do not keep alive for re-established call.
Only need to keep alive for caller in a newly established call.
Change-Id: I36f9d9499c806c8701e3b78555de399b00593be8
ava/com/android/server/sip/SipSessionGroup.java
|
| f89654dd2847cc574dfa6a44806289f7e69e17b7 |
01-Jul-2011 |
Hung-ying Tyan <tyanh@google.com> |
Do not hold wifi lock when SIP is also available over mobile network.
Bug: 3111564
Change-Id: Ifc76e5c378d620e40ce4adf6ffa20807e9750fdb
ava/com/android/server/sip/SipService.java
|
| a6cec8feed5c135bb5f4d6193012d13258a067c4 |
30-Jun-2011 |
Hung-ying Tyan <tyanh@google.com> |
Synchronize SipWakeupTimer.onReceive()
to fix the race of two threads that change mPendingIntent; one assigns a new one
and the other nullifies it.
Change-Id: I5e01f83ea1ac437811d2073839adef9bd0a30ec9
ava/com/android/server/sip/SipWakeupTimer.java
|
| 129d0b08fdf9588f7c8feeb9db3def30973c092e |
29-Jun-2011 |
Hung-ying Tyan <tyanh@google.com> |
Make NAT port timeout measurement more flexible.
In two ways:
(1) When there's a session timeout, restart the measurement at a later time
instead of just stalling.
(2) When there's a port change, do not re-measure the interval if the current
interval works well in the past. We keep success count and decrement it
by half when there's a port change. When the count is below a threshold,
we restart the measurement process.
Change-Id: I7256464435a5e2d2a239bfccaa004e9ceb1d9ce5
ava/com/android/server/sip/SipService.java
|
| 99705b52ec952012bedc4aa8e1f62caff80a6a2f |
28-Jun-2011 |
Hung-ying Tyan <tyanh@google.com> |
Record external IP and port from SIP responses
and use them to create the contact header when sending OK response for INVITE.
Bug: 3461707
Change-Id: I5b254618f4920cf10a1460631bcd336778f344ec
ava/com/android/server/sip/SipHelper.java
ava/com/android/server/sip/SipSessionGroup.java
|
| 2093561a58e602450f6e4f2aae4831edd1b840f4 |
28-Jun-2011 |
repo sync <cywang@google.com> |
Support INVITE w/o SDP.
bug:3326873
Change-Id: Ie29d2c61b237fee2d8637f4ba3d293a22469cced
ava/android/net/sip/SipAudioCall.java
ava/com/android/server/sip/SipSessionGroup.java
|
| 233718c3c5a4f5b4f564af93cb2e42d80a900904 |
27-Jun-2011 |
Hung-ying Tyan <tyanh@google.com> |
Start keepalive process for the caller of a SIP call
so that the callee can send signals (on-hold or bye) back to the caller.
Without the keepalive, the NAT port for the caller will be timed out during the
call. And the signals will be dropped by the NAT device.
Change-Id: I21848d73469045b2ed9e7281556ab184c594c362
ava/com/android/server/sip/SipSessionGroup.java
|
| 1aceda35cc607856ec2e960e0c6cfc6aea87ab8e |
23-Jun-2011 |
repo sync <cywang@google.com> |
Support Invite w/ Replaces request.
bug:3326870
Change-Id: Idbfbe7e3cc6ba83874d42bfb7d149866f454e70a
ava/android/net/sip/ISipSessionListener.aidl
ava/android/net/sip/SipAudioCall.java
ava/android/net/sip/SipSession.java
ava/android/net/sip/SipSessionAdapter.java
ava/com/android/server/sip/SipHelper.java
ava/com/android/server/sip/SipSessionGroup.java
ava/com/android/server/sip/SipSessionListenerProxy.java
|
| e65f3a896f03bba5327ce4f3989c0422855450ca |
24-Jun-2011 |
Hung-ying Tyan <tyanh@google.com> |
Restart NAT port timeout measurement when keepalive fails and other fixes
Misc keepalive fixes including:
+ Restart NAT port timeout measurement when keepalive fails. The max interval
is set to the current keepalive interval.
+ When exception occurs during sending a keepalive, restarts registration.
+ When exception occurs during measurement, retry for a limited times before
giving up.
Change-Id: I7aa787a5ec7c4c9b4334aa1017371d9049b3520c
ava/com/android/server/sip/SipService.java
ava/com/android/server/sip/SipSessionGroup.java
|
| 4af085ff26fbe9e13f7002496fd505dbdb36b282 |
23-Jun-2011 |
Hung-ying Tyan <tyanh@google.com> |
Execute all the due wakeup events in SipWakeupTimer.
Events are sorted by periods. So events of larger periods may have trigger
time (i.e., when the event should be processed) earlier than the ones of smaller
periods. So need to scan the whole queue looking for due events. The scan takes
O(n) time but we expect the queue size to be small.
Change-Id: I08bd3bd9d4bb8decb78f3c91c943396463ca023a
ava/com/android/server/sip/SipWakeupTimer.java
|
| 12750701d0f90ed0166f5ddcf588c1235efe830a |
23-Jun-2011 |
Hung-ying Tyan <tyanh@google.com> |
Keep the keepalive process going after NAT port is changed.
This is a regression from the CL that makes the keep-alive process a reusable
component.
Change-Id: I1d580588e9e303c532bf620056fc0fe88a2fdcda
ava/com/android/server/sip/SipService.java
|
| 4a267a9158a62010cd76ab93681586ea8e3d6015 |
22-Jun-2011 |
Hung-ying Tyan <tyanh@google.com> |
Move the keepalive process to SipSessionImpl and make it reusable.
Reuse the new component in the original keepalive process and the NAT port
mapping timeout measurement process.
This is the foundation for fixing the following bug.
Bug: 3464181
Change-Id: If7e951c000503fa64843942ad062c4d853e20c8d
ava/com/android/server/sip/SipHelper.java
ava/com/android/server/sip/SipService.java
ava/com/android/server/sip/SipSessionGroup.java
|
| ac320b224590c8cdea93a50338aaef5faa1f2466 |
15-Jun-2011 |
Hung-ying Tyan <tyanh@google.com> |
Merge "Move WakeupTimer out of SipService."
|
| 5621554033089d1c07d53f56e8cd9787125d6e28 |
14-Jun-2011 |
Hung-ying Tyan <tyanh@google.com> |
Move WakeupTimer out of SipService.
This is to prepare to move keepalive process to SipSessionGroup before fixing
the following bug.
Bug: 3464181
Change-Id: I57d8f6effad76706b5a76e1269c53d558db88ae4
ava/com/android/server/sip/SipService.java
ava/com/android/server/sip/SipWakeupTimer.java
|
| c133781723f64d1321685d02ad6a208286bf0a42 |
13-Jun-2011 |
repo sync <cywang@google.com> |
Fix the issue of onNetwork in UI thread.
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
ava/com/android/server/sip/SipSessionGroup.java
|
| bb0a989c17cd6135c8d9c8566507521d4d927fe0 |
10-Mar-2011 |
Chung-yih Wang <cywang@google.com> |
Add KeepAlive Interval Measurement.
Change-Id: Id5ea2fcfa0bcd45198e773a5842d39eacc8ae400
ava/com/android/server/sip/SipService.java
ava/com/android/server/sip/SipSessionGroup.java
|
| 34bb419e5946ab28112e9e27a4d1b3928d31e0e2 |
11-May-2011 |
Dima Zavin <dima@android.com> |
update for new audio.h header location
Change-Id: Ic4c62c4037800802427eb7d3c7f5eb8b25d18876
Signed-off-by: Dima Zavin <dima@android.com>
ni/rtp/AudioGroup.cpp
|
| 4b3913a3e43d3180b21d77cc2f717b446350354f |
11-May-2011 |
Andreas Huber <andih@google.com> |
Squashed commit of the following:
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
ni/rtp/Android.mk
|
| b8df57d8767923f40bd52a0d2d1114e67fa76fa5 |
06-May-2011 |
Scott Main <smain@google.com> |
am d81214da: am a7a9c4cb: am 46524f83: Merge "docs: add package description for RTP" into honeycomb-mr1
* commit 'd81214dae45a4b38919296af41bf756e3931675a':
docs: add package description for RTP
|
| d81214dae45a4b38919296af41bf756e3931675a |
06-May-2011 |
Scott Main <smain@google.com> |
am a7a9c4cb: am 46524f83: Merge "docs: add package description for RTP" into honeycomb-mr1
* commit 'a7a9c4cbbc2315a59ad27b43c83c66e272dcc2f2':
docs: add package description for RTP
|
| de9acb76d9ea398d0ba4c5e62df554f5696eaa99 |
06-May-2011 |
Scott Main <smain@google.com> |
docs: add package description for RTP
Change-Id: I02c181a48101be288fb4aabf497f573f00038f90
ava/android/net/rtp/package.html
|
| 24fc2fb1c541e954b83fd31ea9f786a5e9b45501 |
20-Apr-2011 |
Dima Zavin <dima@android.com> |
audio/media: convert to using the audio HAL and new audio defs
Change-Id: Ibc637918637329e4f2b62f4ac7781102fbc269f5
Signed-off-by: Dima Zavin <dima@android.com>
ni/rtp/AudioGroup.cpp
|
| d8cbd16659d1a5d098acfcfa8ee98c95036aff63 |
05-Apr-2011 |
Eric Laurent <elaurent@google.com> |
am 7a492a9a: am b7a76e84: am a482d83c: Merge "Issue 4157048: mic gain for VoIP/SIP calls." into gingerbread
* commit '7a492a9ad42947a3a7b777b0eb6eec56f5bb942b':
Issue 4157048: mic gain for VoIP/SIP calls.
|
| 7a492a9ad42947a3a7b777b0eb6eec56f5bb942b |
05-Apr-2011 |
Eric Laurent <elaurent@google.com> |
am b7a76e84: am a482d83c: Merge "Issue 4157048: mic gain for VoIP/SIP calls." into gingerbread
* commit 'b7a76e84fde7fe534d46aaaa71e3224798354009':
Issue 4157048: mic gain for VoIP/SIP calls.
|
| b7a76e84fde7fe534d46aaaa71e3224798354009 |
04-Apr-2011 |
Eric Laurent <elaurent@google.com> |
am a482d83c: Merge "Issue 4157048: mic gain for VoIP/SIP calls." into gingerbread
* commit 'a482d83ccf35ccd6cc29a9e1ace3d77b5f28d013':
Issue 4157048: mic gain for VoIP/SIP calls.
|
| d7a724e6d89420408200c20937baa3b2bd902742 |
30-Mar-2011 |
Eric Laurent <elaurent@google.com> |
Issue 4157048: mic gain for VoIP/SIP calls.
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
ni/rtp/AudioGroup.cpp
|
| 397de169e5462bf0c62506827819f93336b3f123 |
29-Mar-2011 |
Brad Fitzpatrick <bradfitz@android.com> |
am fae5e289: am 6f67e7bf: am 2e383bc6: Merge "Making it possible to call SIP calls with special allowed chars."
* commit 'fae5e2894ff3c09f27efac2a7ee6b9cfd4ed14b0':
Making it possible to call SIP calls with special allowed chars.
|
| fae5e2894ff3c09f27efac2a7ee6b9cfd4ed14b0 |
29-Mar-2011 |
Brad Fitzpatrick <bradfitz@android.com> |
am 6f67e7bf: am 2e383bc6: Merge "Making it possible to call SIP calls with special allowed chars."
* commit '6f67e7bf831147257e078dd72a22f2e43e009122':
Making it possible to call SIP calls with special allowed chars.
|
| b5c72ead014a509c0f84884d1f2dac1ff9deec8e |
22-Mar-2011 |
Magnus Strandberg <magnus.strandberg@sonyericsson.com> |
Making it possible to call SIP calls with special allowed chars.
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
ava/com/android/server/sip/SipHelper.java
|
| 3f9e08973f33a5640e52381431ef02aa184dd018 |
22-Mar-2011 |
Carl Shapiro <cshapiro@google.com> |
Include strings.h instead of string.h for the strcasecmp prototype.
Change-Id: I6b0ddc2408c30851edcffb36f1bc74245403ffc7
ni/rtp/AudioCodec.cpp
|
| 3070af08821ee86f06a9cc6b58dbb79c82946b94 |
14-Mar-2011 |
Iliyan Malchev <malchev@google.com> |
frameworks/base: remove LOCAL_PRELINK_MODULE
Change-Id: I54dd62ebef47e7690afa5a858f3cad941b135481
Signed-off-by: Iliyan Malchev <malchev@google.com>
ni/rtp/Android.mk
|
| 6defd2d47e81b206d76430266120294a40592b27 |
03-Mar-2011 |
Chia-chi Yeh <chiachi@android.com> |
NEW_API: Unhide RTP APIs.
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
ava/android/net/rtp/AudioCodec.java
ava/android/net/rtp/AudioGroup.java
ava/android/net/rtp/AudioStream.java
ava/android/net/rtp/RtpStream.java
|
| c52f5b2ec5e13ab3d9ab016e6cab757d4ecb45c7 |
03-Mar-2011 |
Chia-chi Yeh <chiachi@android.com> |
RTP: update javadocs.
Change-Id: If600df0eb1e6135aed9f3b2eacfb6bc9ed5d78ff
ava/android/net/rtp/AudioGroup.java
ava/android/net/rtp/AudioStream.java
|
| 89bc1fe73efb73b43758d41c9ff9f2f4902dd019 |
25-Feb-2011 |
Chung-yih Wang <cywang@google.com> |
Activate the wifi high perf. for sip calls.
bug:3487791
Change-Id: I7d8d146f8542cd7df387547c7ce3d5ded27f8e97
ava/android/net/sip/SipAudioCall.java
|
| fcd0e50da51074703929e9f7b700a2cd11bd67e0 |
21-Feb-2011 |
Chung-yih Wang <cywang@google.com> |
Add rport argument for a reinvite request.
bug:3461707
Change-Id: I69a4f84dde3929c754c838fd12e624b774f44826
ava/com/android/server/sip/SipHelper.java
|
| 9e25df44631e3c7881a6816cf26f34ea24055c72 |
10-Feb-2011 |
Chung-yih Wang <cywang@google.com> |
Make SIP AuthName APIs public.
bug:3326867
Change-Id: I766e6e28f6ad3e84de2c9e24850d472ad00271cc
ava/android/net/sip/SipProfile.java
|
| 2ba92c71b5684dce700cf848bf157153c156df1d |
15-Nov-2010 |
Jean-Michel Trivi <jmtrivi@google.com> |
do not merge bug 3370834 Cherrypick from master
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
ni/rtp/AudioGroup.cpp
|
| 14b6d0620b42d1bb3a55778ba452d838a0d89223 |
25-Jan-2011 |
Hung-ying Tyan <tyanh@google.com> |
Merge changes Ib70e0cf2,I0691cd70 into gingerbread
* changes:
SipService: registers broadcast receivers on demand.
SipService: release wake lock for cancelled tasks.
|
| f46013b67219b0b2e95fcebb0e51e9816ab0ce94 |
18-Jan-2011 |
Hung-ying Tyan <tyanh@google.com> |
Merge "Merge "SipService: registers broadcast receivers on demand."" into honeycomb
|
| e9b54077274d0c4066093cd90dabca59b3d9a157 |
07-Jan-2011 |
Hung-ying Tyan <tyanh@google.com> |
Merge "SipService: registers broadcast receivers on demand."
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
ava/com/android/server/sip/SipService.java
|
| 40f2cacbc9ab00d34c2a4f49519921bbf6b5293a |
06-Jan-2011 |
Hung-ying Tyan <tyanh@google.com> |
Merge "SipService: release wake lock for cancelled tasks."
Bug: 3327004
Change-Id: Ice47f973b5f2969f26eaa83a3e4795b2e153ba8b
ava/com/android/server/sip/SipService.java
|
| 0f7de88cb9eef781117fa2f2b69ba2698237637e |
06-Jan-2011 |
Chung-yih Wang <cywang@google.com> |
Merge "Add auth. username in SipProfile." from gingerbread.
bug:3326867
Change-Id: Ic67dd7d4858f28224e4f01ad8b65bcd3a3c15f10
ava/android/net/sip/SipProfile.java
ava/com/android/server/sip/SipSessionGroup.java
|
| f268a2f8488b6b111126a7043a5f1f559a566fa7 |
06-Jan-2011 |
Chung-yih Wang <cywang@google.com> |
Add auth. username in SipProfile.
bug:3326867
Change-Id: I2a62c75fb3f5e9c6ec2e00b29396e93b0c183d9b
ava/android/net/sip/SipProfile.java
ava/com/android/server/sip/SipSessionGroup.java
|
| f0bb1ce70f8d5f19f0d63be070997ef237a15fe6 |
07-Jan-2011 |
Hung-ying Tyan <tyanh@google.com> |
SipService: registers broadcast receivers on demand.
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
ava/com/android/server/sip/SipService.java
|
| d87be273aaea32995c87a6cbc6250cbfeeddd84d |
06-Jan-2011 |
Chia-chi Yeh <chiachi@android.com> |
Enable built-in echo canceler if available.
1. Always initialize AudioRecord with VOICE_COMMUNICATION.
2. If echo canceler is available, disable our echo suppressor.
Change-Id: Idf18d3833189a8478c1b252ebe6ce55e923280b3
ni/rtp/AudioGroup.cpp
|
| 4bf82df2f069b5a788689064bf8d3f6b612587d4 |
06-Jan-2011 |
Chia-chi Yeh <chiachi@android.com> |
Do not set back to AudioManager.MODE_NORMAL in SipAudioCall.
Change-Id: I8f68e01e5f8c73bb8afd44312cbfadb55aab4330
ava/android/net/sip/SipAudioCall.java
|
| 0c01e6e060d079b0a25a44c1159db63944afce17 |
06-Jan-2011 |
Hung-ying Tyan <tyanh@google.com> |
SipService: release wake lock for cancelled tasks.
Bug: 3327004
Change-Id: I0691cd70edf61f815ecb0613aca85babd89f6cc4
ava/com/android/server/sip/SipService.java
|
| d0da38079617e867db5d2bbdaaaa4cd49027d4eb |
05-Jan-2011 |
Chia-chi Yeh <chiachi@android.com> |
am dc78e3fe: am 3cf71376: RTP: Send silence packets on idle streams for every second.
* commit 'dc78e3fe7f2ffbc810cd54e86e3a83e279d74984':
RTP: Send silence packets on idle streams for every second.
|
| 3cf71376421f942d06b30101fbf0df7f3b23fbdd |
04-Jan-2011 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Send silence packets on idle streams for every second.
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
ni/rtp/AudioGroup.cpp
|
| 33808c6d2448bbc944905819c213f2debf18af5a |
22-Dec-2010 |
Hung-ying Tyan <tyanh@google.com> |
am aec9a33f: am e0bd2688: Merge "Check if VoIP API is supported in SipManager." into gingerbread
* commit 'aec9a33f1cfc7c32690bc8e24aefaeb137ab9859':
Check if VoIP API is supported in SipManager.
|
| 5bd3782f244212cd8ef51bf9f3578869b08b4e18 |
20-Dec-2010 |
Hung-ying Tyan <tyanh@google.com> |
Check if VoIP API is supported in SipManager.
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
ava/android/net/sip/SipAudioCall.java
ava/android/net/sip/SipManager.java
|
| 635b2b77b917c1bf20ce135ce6fcc98a6a7be084 |
20-Dec-2010 |
Hung-ying Tyan <tyanh@google.com> |
am d90bc225: am a936b256: Remove SIP realm/domain check
* commit 'd90bc225b9d6e4f8f69d984aa63062a7b20ac65c':
Remove SIP realm/domain check
|
| a936b256eb1611b5d8b88d0cd61f21225152cc82 |
16-Dec-2010 |
Hung-ying Tyan <tyanh@google.com> |
Remove SIP realm/domain check
as the realm may be different from the domain.
Bug: 3283834
Change-Id: I64c9f0d6d626afdb397c5d378d30afa9d6a64ca9
ava/com/android/server/sip/SipSessionGroup.java
|
| 58ee2acba8953814cc4bf65d2f28f7dd498b5779 |
16-Dec-2010 |
Hung-ying Tyan <tyanh@google.com> |
Check port in create peer's SIP profile.
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
ava/android/net/sip/SipProfile.java
ava/com/android/server/sip/SipSessionGroup.java
|
| eecf4a6f11129461088d620afadb6014edab3086 |
16-Dec-2010 |
Hung-ying Tyan <tyanh@google.com> |
Check port in create peer's SIP profile.
SipURI returns port -1 when port is not present in the URI.
Don't call SipProfile.Builder.setPort() when that happens.
Change-Id: Ic5fe7301195705a77010038cae20d6629b33135e
ava/android/net/sip/SipProfile.java
ava/com/android/server/sip/SipSessionGroup.java
|
| c030a164c8a890947985d15722fe3df8785f7d04 |
07-Dec-2010 |
Chung-yih Wang <cywang@google.com> |
am c9cc9ab5: am 5f86d7f5: Merge "Fix SIP bug of different transport/port used for requests." into gingerbread
* commit 'c9cc9ab590ef879877e466c0b5f5823e11bb4c47':
Fix SIP bug of different transport/port used for requests.
|
| f053292d7a46c30abbe6f12ca04dbc03ec964d80 |
03-Nov-2010 |
Chung-yih Wang <cywang@google.com> |
Fix SIP bug of different transport/port used for requests.
bug: http://b/3156148
Change-Id: I4fa5b274d2e90ebde12d9e99822dc193a65bad32
ava/android/net/sip/SipProfile.java
ava/com/android/server/sip/SipHelper.java
|
| 2aef9a1e847a7612549d9a0280cde6489e540f6b |
03-Dec-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 7da1ffc9: am e2abd103: Merge "Set AudioGroup mode according to audio settings" into gingerbread
* commit '7da1ffc9d2a51ef6120389a06351fd770ab45f46':
Set AudioGroup mode according to audio settings
|
| e2abd103a2d311738ff1dd1e1d9b8e6c52aa870c |
03-Dec-2010 |
Hung-ying Tyan <tyanh@google.com> |
Merge "Set AudioGroup mode according to audio settings" into gingerbread
|
| d6b0d689a77934a0124ed9d3a59c9534a5d4958b |
01-Dec-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 6034f9b2: am 06e8cdc0: Fix race between ending and answering a SIP call.
* commit '6034f9b2664799cb4f983657a78023b49efff825':
Fix race between ending and answering a SIP call.
|
| db4245291b15fd966b36c70f7f69ba4d22539803 |
01-Dec-2010 |
Hung-ying Tyan <tyanh@google.com> |
am ed34b244: am d7116ff1: Merge "Do not suppress error feedback during a SIP call." into gingerbread
* commit 'ed34b244f1665b604d2a291db504415b10a514d7':
Do not suppress error feedback during a SIP call.
|
| 06e8cdc0f81ead604d5adf9d7b3f982e10226fd2 |
25-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
Fix race between ending and answering a SIP call.
+ Also fix race between ending and changing (holding/unholding) a SIP call.
+ Remove an unused method.
Bug : 3128233
Change-Id: Ie18d8333a88f0d9906d54988243d909b58e07e4b
ava/com/android/server/sip/SipSessionGroup.java
|
| 4c7cc83827458945fe7a1f4bd2bfe0629f0d30ae |
01-Dec-2010 |
Chia-chi Yeh <chiachi@android.com> |
Merge "RTP: Prepare to unhide the APIs."
|
| 53aa6ef70d8692277f9403f94d43918ad9712dd0 |
30-Nov-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Prepare to unhide the APIs.
Polish things a little bit.
Change-Id: I2c3cea8b34b9c858879bc722ea1f38082ba22b8d
ava/android/net/rtp/AudioGroup.java
ava/android/net/rtp/AudioStream.java
ava/android/net/rtp/RtpStream.java
ni/rtp/AmrCodec.cpp
ni/rtp/AudioGroup.cpp
|
| 1c8c173666313f8ab589fa54911661cbc41a5e8f |
01-Dec-2010 |
Hung-ying Tyan <tyanh@google.com> |
am c41b27e2: am 349f3509: Merge "Correct SipService.isOpened() implementation." into gingerbread
* commit 'c41b27e2748ee19620636a14721a1dc14c3b418c':
Correct SipService.isOpened() implementation.
|
| 121006789e9db9a35f0716e571a55fb0936c0783 |
01-Dec-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 5c85338d: am d9e12303: Merge "Notify SipSessions before closing SIP stack." into gingerbread
* commit '5c85338dcf85462534d85440ded100a8012ff9dd':
Notify SipSessions before closing SIP stack.
|
| ebf28fa3f086bd5d3fa8d988fe4b8a8faeddd710 |
01-Dec-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 0e58a952: am 0bba9535: Merge "Throw proper exceptions in SipManager" into gingerbread
* commit '0e58a9529895e270dae90e69486a59e41de714b8':
Throw proper exceptions in SipManager
|
| 342a9be0096752aa0ae80120a64751a7d0b9e7a2 |
01-Dec-2010 |
Chia-chi Yeh <chiachi@android.com> |
am e843dfa8: am bd399b0b: Merge "RTP: Pause echo suppressor when far-end volume is low." into gingerbread
* commit 'e843dfa8dcd0a7bfa956b75424bb5db834975a64':
RTP: Pause echo suppressor when far-end volume is low.
|
| fa81463e88d15859b557be6fef5982b049b92ab8 |
25-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
Set AudioGroup mode according to audio settings
Set AudioGroup mode according to holding, mute and speaker phone settings.
Bug: 3119690
Change-Id: I02803ae105409b7f8482e6c2ef3e67623bd54e03
ava/android/net/sip/SipAudioCall.java
|
| 4189d99b6e4877352049b7447b7f0734ef99b9e8 |
24-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
Do not suppress error feedback during a SIP call.
Bug: 3124788
Change-Id: Ia0a06f72336d1795515428eba0c9f875c32d13d1
ava/com/android/server/sip/SipSessionGroup.java
|
| 349f3509f7335bbdef443a75afa36fb3c2d9552c |
30-Nov-2010 |
Hung-ying Tyan <tyanh@google.com> |
Merge "Correct SipService.isOpened() implementation." into gingerbread
|
| d9e12303d279654afe16319f948f93490cd1b4d5 |
30-Nov-2010 |
Hung-ying Tyan <tyanh@google.com> |
Merge "Notify SipSessions before closing SIP stack." into gingerbread
|
| 0bba9535413f9ceefe03f1cef9ddaddccd05cae5 |
30-Nov-2010 |
Hung-ying Tyan <tyanh@google.com> |
Merge "Throw proper exceptions in SipManager" into gingerbread
|
| bd399b0bd205a1a3889bae1a619c6d4d4a0f4816 |
30-Nov-2010 |
Chia-chi Yeh <chiachi@android.com> |
Merge "RTP: Pause echo suppressor when far-end volume is low." into gingerbread
|
| 8d1b2a17d9935819ec96f1b5fca0e9945f564eaa |
03-Nov-2010 |
Hung-ying Tyan <tyanh@google.com> |
Throw proper exceptions in SipManager
instead of silently returning null and causing NPE in applications as returning
null is not documented in the javadoc.
Add connection to the connection list in SipCall after dial() succeeds so that
we don't need to clean up if it fails. The original code will cause the failed
connection to continue to live in the SipCall and in next dial() attempt, a new
connection is created and the in-call screen sees two connections in the call
and thus shows conference call UI.
Bug: 3157234, 3157387
Change-Id: Iabc3235f781c4f1e09384a67ad56b09ad2c12e5e
ava/android/net/sip/SipManager.java
|
| 262cdfca7a0940735d3a08779e2d01bfdf639294 |
02-Nov-2010 |
Hung-ying Tyan <tyanh@google.com> |
Correct SipService.isOpened() implementation.
Make it return true for all existing accounts.
Rename mOpened to mOpenedToReceiveCalls to make it less confusing.
Bug: 3155849
Change-Id: I327f411bf76afd73434ad1fa2ffef3db1e35d778
ava/com/android/server/sip/SipService.java
|
| e5bc8f617b48ab237bec22dd4572e678642f25eb |
29-Oct-2010 |
Scott Main <smain@google.com> |
am 9a8df805: am 1112632a: Merge "docs: revise javadocs for sip add a package description, revise class descriptions and edit some method docs" into gingerbread
* commit '9a8df8054b2e38a27d8e8e6b17365979218f0e3f':
docs: revise javadocs for sip add a package description, revise class descriptions and edit some method docs
|
| 02b1d685cc287d7c53141872b3d80be4ee5dd59e |
22-Oct-2010 |
Scott Main <smain@google.com> |
docs: revise javadocs for sip
add a package description, revise class descriptions and edit some method docs
Change-Id: Ice969a99c830349674c65d99e4b7a6f1d2f24a7e
ava/android/net/sip/SipAudioCall.java
ava/android/net/sip/SipErrorCode.java
ava/android/net/sip/SipException.java
ava/android/net/sip/SipManager.java
ava/android/net/sip/SipProfile.java
ava/android/net/sip/SipRegistrationListener.java
ava/android/net/sip/SipSession.java
ava/android/net/sip/package.html
|
| 0c7d30660c4573189570846a6ae0805d014fab56 |
27-Oct-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Pause echo suppressor when far-end volume is low.
Bug: 3136725
Change-Id: Ieeedd2836d3028045aacac963f44285491708cc3
ni/rtp/EchoSuppressor.cpp
|
| 5d0c5cf2d6c6e82bcdce95d72d9000a934b2f354 |
22-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
Notify SipSessions before closing SIP stack.
Bug: 3116480
Change-Id: I748d63382ade250aed27ccb09ea68c76a433fd27
ava/com/android/server/sip/SipSessionGroup.java
|
| 2754b4bde824bfbdc483ad627aaaea87971b053a |
22-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
am d4d3f36f: am 1257d330: Merge "Clean up pending sessions on incoming call in SipService" into gingerbread
Merge commit 'd4d3f36f4c25b41f4253eadd5e67035fe220cad3'
* commit 'd4d3f36f4c25b41f4253eadd5e67035fe220cad3':
Clean up pending sessions on incoming call in SipService
|
| 60c45d026907edbe340c8cf9e1723b3dd34f8b6a |
22-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
Clean up pending sessions on incoming call in SipService
Bug: 3122186
Change-Id: I25c9aa19d138f6940a29025d54e7bc2ffb7daa29
ava/com/android/server/sip/SipService.java
|
| 39df5331c7423455f1fb6c01b075b618f8bc00fd |
22-Oct-2010 |
Chia-chi Yeh <chiachi@android.com> |
am 044fcd64: am 703aae06: Merge "RTP: Fix non-zero DC in EchoSuppressor caused while aggregating samples." into gingerbread
Merge commit '044fcd64fe999dca0f986dfce9cb3b5b1da77f44'
* commit '044fcd64fe999dca0f986dfce9cb3b5b1da77f44':
RTP: Fix non-zero DC in EchoSuppressor caused while aggregating samples.
|
| 6d848f759e901264935ed7ba1094d865e3b2c16b |
21-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
am bdc15d8b: am 4056ab97: Merge "Add permission requirements to SipAudioCall and SipManager javadoc." into gingerbread
Merge commit 'bdc15d8b43e7763bf72780d0f683b223b8cf6625'
* commit 'bdc15d8b43e7763bf72780d0f683b223b8cf6625':
Add permission requirements to SipAudioCall and SipManager javadoc.
|
| 703aae06c0925b19657877bb1872bb2f28874969 |
21-Oct-2010 |
Chia-chi Yeh <chiachi@android.com> |
Merge "RTP: Fix non-zero DC in EchoSuppressor caused while aggregating samples." into gingerbread
|
| 8a68b52b9873f1f3d7114576c9f39a2b7b402152 |
21-Oct-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Fix non-zero DC in EchoSuppressor caused while aggregating samples.
Rewrite using integer arithmetic to get full 32-bit precision instead
of 23-bit in single precision floating-points.
Bug: 3029745
Change-Id: If67dcc403923755f403d08bbafb41ebce26e4e8b
ni/rtp/AudioGroup.cpp
ni/rtp/EchoSuppressor.cpp
ni/rtp/EchoSuppressor.h
|
| 164cd438fb21e82d0aacc06da940041f0b7f6a2c |
21-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 51028569: am 1180f2a0: Merge "Remove ringtone API from SipAudioCall." into gingerbread
Merge commit '5102856947595cffc1cceb11b9e4c5baf70b2e82'
* commit '5102856947595cffc1cceb11b9e4c5baf70b2e82':
Remove ringtone API from SipAudioCall.
|
| 16f5d6ee797543b42ea457c8a8724a569eac6595 |
21-Oct-2010 |
Chung-yih Wang <cywang@google.com> |
am aefcdde4: am 4944fdd7: Periodically scan wifi when wifi is not connected and wifi lock is grabbed in SipService.
Merge commit 'aefcdde4bdf2be74bdf9620359830faeed5419e6'
* commit 'aefcdde4bdf2be74bdf9620359830faeed5419e6':
Periodically scan wifi when wifi is not connected and wifi lock is
|
| 9d6d17fcd80ca43fb57c4bb5808e55e79f0fbe26 |
21-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 35d9e770: am 0a6e717f: Handle dialing a SIP call to self.
Merge commit '35d9e7701eea343d8cdfcd3c990ae74685b299b2'
* commit '35d9e7701eea343d8cdfcd3c990ae74685b299b2':
Handle dialing a SIP call to self.
|
| 385a753dead6ef15f2e30eae47f73e642b3ef7ed |
21-Oct-2010 |
Brad Fitzpatrick <bradfitz@android.com> |
resolved conflicts for merge of 368fdba4 to master
Change-Id: I42b7b433c86a71a5da5db67109f056a280077c9d
|
| e87b644402642bad7147f915849bfa0eadaea446 |
18-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
Add permission requirements to SipAudioCall and SipManager javadoc.
Bug: 3116259
Change-Id: I00a033794e9d3e1c2d2ccfe4e612cd50003ec2ee
ava/android/net/sip/SipAudioCall.java
ava/android/net/sip/SipManager.java
|
| 9b449e5606786f7c197679f8f9d25985308bfb72 |
20-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
Remove ringtone API from SipAudioCall.
(watch out auto-merge conflict for SipAudioCall).
Bug: 3113033, related CL: https://android-git/g/#change,75185
Change-Id: Ib48d3b990e229e0b341e47e10e76934e1a50d10f
ava/android/net/sip/SipAudioCall.java
ava/android/net/sip/SipManager.java
|
| 4944fdd771d2a629b3c1af8097df5eb2de02d9ee |
19-Oct-2010 |
Chung-yih Wang <cywang@google.com> |
Periodically scan wifi when wifi is not connected and wifi lock is
grabbed in SipService.
bug: http://b/3077454
Change-Id: I153974325c29e0f927c8eb7fdbc4725aaf10087d
ava/com/android/server/sip/SipService.java
|
| 0a6e717fb6846f66b8dc853e079f2166307bfc60 |
18-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
Handle dialing a SIP call to self.
Reply BUSY HERE response so server may redirect the call to the voice mailbox.
http://b/issue?id=3103072
http://b/issue?id=3109479
Change-Id: I81f5dd59ad87298dd9dda87084538ee460eabba8
ava/com/android/server/sip/SipHelper.java
ava/com/android/server/sip/SipService.java
ava/com/android/server/sip/SipSessionGroup.java
|
| 431bb2269532f2514861b908d5fafda8fa64da79 |
19-Oct-2010 |
Joe Onorato <joeo@google.com> |
Reduce logging.
Remember, the system and main logs are
- Shared resources
- Primarily for recording problems
- To be used only for large grained events during normal operation
Bug: 3104855
Change-Id: I136fbd101917dcbc8ebc3f96f276426b48bde7b7
ava/com/android/server/sip/SipService.java
|
| dc58e5211ff76677a545c5ce4c39a5fd246cffee |
19-Oct-2010 |
Chung-yih Wang <cywang@google.com> |
am cd6fe63f: am b4116c09: Fix the incorrect environment variable name for the thread pool size.
Merge commit 'cd6fe63fdc2e99da11b19a233afd81e2448d0db2'
* commit 'cd6fe63fdc2e99da11b19a233afd81e2448d0db2':
Fix the incorrect environment variable name for the thread pool size.
|
| 78c206c75095b4a9092aabed8e79702ae0c5dc86 |
19-Oct-2010 |
John Huang <jsh@google.com> |
am 085996c4: am 45bd8303: Merge "Uncomment SIP/VOIP feature check in SipManager." into gingerbread
Merge commit '085996c411b4d3878dfd97c59bfc4a17da08959b'
* commit '085996c411b4d3878dfd97c59bfc4a17da08959b':
Uncomment SIP/VOIP feature check in SipManager.
|
| 723e997ef9745ad05c2499509c21976a42f2f278 |
19-Oct-2010 |
John Huang <jsh@google.com> |
am 3e9bcb98: am 382717f3: Merge "Set the thread pool size of NIST sip stack to one." into gingerbread
Merge commit '3e9bcb98c4190b18d113e79ead071a86cd7ca480'
* commit '3e9bcb98c4190b18d113e79ead071a86cd7ca480':
Set the thread pool size of NIST sip stack to one.
|
| b4116c09fb9784551911ea0a10b4dd321ff9aa7d |
19-Oct-2010 |
Chung-yih Wang <cywang@google.com> |
Fix the incorrect environment variable name for the thread pool size.
bug: http://b/3099715
Change-Id: I531048414f22c8edcd9c4f815c12a0bdd6347640
ava/com/android/server/sip/SipSessionGroup.java
|
| 45bd8303fe20f20843f5a76ddf42a5ace54add58 |
18-Oct-2010 |
John Huang <jsh@google.com> |
Merge "Uncomment SIP/VOIP feature check in SipManager." into gingerbread
|
| a0cdfbf5b74a92611789b7ec08a84274b9011021 |
18-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
Uncomment SIP/VOIP feature check in SipManager.
http://b/issue?id=2971947
Change-Id: I3afa8eb03c4e347b382213dd388354365f766b2f
ava/android/net/sip/SipManager.java
|
| 66cc5355a137e291cc1e3c5d871e1d9cd35ee0ab |
18-Oct-2010 |
Chung-yih Wang <cywang@google.com> |
Set the thread pool size of NIST sip stack to one.
Set the thread pool size to one to fix the out-of-order packets
seen in sip service when the device is waken up from sleep.
bug:http://b/3099715
Change-Id: Ia169e3fde77488068c369e3345ecf6a6d8ddf792
ava/com/android/server/sip/SipSessionGroup.java
|
| e1baa9c79ffd9240a20bab88c865afae5ce53e59 |
16-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
am ebad42d6: am 3fbfee2f: Merge "SipService: add wake lock for incoming INVITE packets." into gingerbread
Merge commit 'ebad42d6d35dc0dc07fe89650268453dbdff8a79'
* commit 'ebad42d6d35dc0dc07fe89650268453dbdff8a79':
SipService: add wake lock for incoming INVITE packets.
|
| 0601be0803538416857425a69ccb895866eb5e4a |
15-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 4f5eb955: am bd57eeaf: SipService: add wake lock for multiple components.
Merge commit '4f5eb9550ba2cc037c4aa81613574a38a956dd5c'
* commit '4f5eb9550ba2cc037c4aa81613574a38a956dd5c':
SipService: add wake lock for multiple components.
|
| 61312df60783ca49b664d5ae82f007bc779d132e |
15-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
am f3a935f6: am 3bb10442: Merge "Make SipService listen to WIFI state change events." into gingerbread
Merge commit 'f3a935f6aa421110c7b6f36f922223c195f751bc'
* commit 'f3a935f6aa421110c7b6f36f922223c195f751bc':
Make SipService listen to WIFI state change events.
|
| 4fe18a4998ad1d3d0a62a6b35f9225304c34ef00 |
15-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 8a362186: am f1b1eec9: Merge "SipService: mScreenOn is flipped to wrong value." into gingerbread
Merge commit '8a3621866d8a04b0a229eb5765ea9799e68fd90c'
* commit '8a3621866d8a04b0a229eb5765ea9799e68fd90c':
SipService: mScreenOn is flipped to wrong value.
|
| 28f63c06894b9ca9252f43bc54a098c0a785d4b4 |
14-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
SipService: add wake lock for incoming INVITE packets.
+ Keep the wake lock for 500ms. (Some measurements on N1 indicate 160~180ms
needed to bring up InCallScreen but since INVITE doesn't come in frequently
we can be more generous just to be safe.)
+ Move MyWakeupLock out of SipService so SipSessionGroup can use it without
awkward inter-dependency with SipService.
+ Add acquire(int timeout) to be used to create the "timed" wake lock.
http://b/issue?id=3081828
Change-Id: Iffd1d78d1a5cae9f795252ada75310917095204d
ava/com/android/server/sip/SipService.java
ava/com/android/server/sip/SipSessionGroup.java
ava/com/android/server/sip/SipWakeLock.java
|
| 379694f528cbf947b4dbe3fecffced4162cc74e4 |
15-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 06e55977: am 907f6f1d: Merge "Fix SipSessionGroup from throwing ConcurrentModificationException" into gingerbread
Merge commit '06e559779edd93a83100824b36c9bf67a27db178'
* commit '06e559779edd93a83100824b36c9bf67a27db178':
Fix SipSessionGroup from throwing ConcurrentModificationException
|
| bd57eeafe034cf850225db403700b5dc5db5ebcc |
13-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
SipService: add wake lock for multiple components.
+ Add MyWakeLock to maintain a global wake lock for multiple components.
+ Use a Set to store components that want to hold the lock.
+ When the first component enters the set, we grab the global wake lock.
+ When the set becomes empty, we release the global lock.
+ In places like no account being opened to receive calls, we reset the
wake lock just to be safe from possible leakage.
+ Make MyExecutor aware of the wake lock. It will grab the wake lock on
behalf of the task so that tasks don't need to worry about the lock.
+ Connectivity receiver is modified to be executed in MyExecutor.
+ WakeupTimer handler is already protected by AlarmManager's wake lock but
all the timeout handlers that register themselves to the WakeupTimer are
to be executed in MyExecutor to be protected by the wake lock.
+ Remove unnecessary code in the Keepalive and registration processes. Since
both processes are executed in MyExecutor submitted by the WakeupTimer (as
they are timeout handlers registered to the WakeupTimer), they don't need
to add themselves to MyExecutor explicitly in their run() callbacks.
+ Make the keepalive process wait for at most 3 seconds instead of forever for
server response. It could cause the wake lock to be held longer than necessary
and is a potential cause for ANR.
http://b/issue?id=3081828
Related bug:
http://b/issue?id=3087153
Change-Id: Idee0ddb837e67daa0d5092c012bb242bd7c18431
ava/com/android/server/sip/SipService.java
ava/com/android/server/sip/SipSessionGroup.java
|
| 8635bb5405a50b86badcb3e674032d6f444d4944 |
14-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
am c74d3917: am ece7e11b: Merge "SipService: fix a missing switch-case break." into gingerbread
Merge commit 'c74d39173e1071dbffe713e008b95784ac4312bc'
* commit 'c74d39173e1071dbffe713e008b95784ac4312bc':
SipService: fix a missing switch-case break.
|
| 4f8fd10f761d562ad3a4e01e78fc7046d3c9936c |
13-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
Make SipService listen to WIFI state change events.
+ Grab a WIFI lock if any account is opened to receive calls and WIFI is enabled
+ Release the WIFI lock if no account is opened to receive calls or WIFI is
disabled
+ Remove screen on/off event receiver
http://b/issue?id=3077454
Change-Id: Ifdf60a850bcf4106c75ec1e7563b26d8b33d7e92
ava/com/android/server/sip/SipService.java
|
| f1b1eec9de1ca27b660f6c00487de07a30eecf64 |
13-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
Merge "SipService: mScreenOn is flipped to wrong value." into gingerbread
|
| d6fc979edbba6c65cac1085fb5f2b8b972713758 |
13-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
SipService: mScreenOn is flipped to wrong value.
http://b/issue?id=3077454
Change-Id: I23b6f70730074689b939e449c2c202ce8ffb586f
ava/com/android/server/sip/SipService.java
|
| ebc886c857a702d788c2bdfb9e0b3d20746ad745 |
12-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
Fix SipSessionGroup from throwing ConcurrentModificationException
http://b/issue?id=3087256
Change-Id: I67df64105db7c1295649f1f3ce77f99025ce3d44
ava/com/android/server/sip/SipSessionGroup.java
|
| 685b61b7117dbae94c5ceb5de4546ad23a4d3d0f |
12-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
SipService: fix a missing switch-case break.
Change-Id: I638eecd8000293d4cb37b3595c02ca33df4924eb
ava/com/android/server/sip/SipService.java
|
| 82b400387114634e5b0b8c08ac142cb69ccf14cf |
12-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
am df08c2f0: am 692cac9f: SipHelper: add debug log for challenge responses.
Merge commit 'df08c2f03e7cf7be7002d34efe8e4e8d24e406de'
* commit 'df08c2f03e7cf7be7002d34efe8e4e8d24e406de':
SipHelper: add debug log for challenge responses.
|
| ec17ab33542d0962f043155d9271a9c8b725497a |
11-Oct-2010 |
Chung-yih Wang <cywang@google.com> |
am 833db408: am dc2e5208: Merge "Do not release the wifi lock if the screen is off." into gingerbread
Merge commit '833db40866ebf27be33aa387d08a2cb0b9a4246d'
* commit '833db40866ebf27be33aa387d08a2cb0b9a4246d':
Do not release the wifi lock if the screen is off.
|
| 692cac9fdd7b179ba807351772fdf2339c000dfe |
10-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
SipHelper: add debug log for challenge responses.
Change-Id: If0143a0f076ef30b1b8998e477df933923bfa7b1
ava/com/android/server/sip/SipHelper.java
|
| e06be941185e33392dde3dcaba85c67ce5423578 |
10-Oct-2010 |
Chung-yih Wang <cywang@google.com> |
am 77880ae4: am 44b7ef54: Merge "SipService: add permission check for using API" into gingerbread
Merge commit '77880ae4187d92506848249585687fc9d0c0dd25'
* commit '77880ae4187d92506848249585687fc9d0c0dd25':
SipService: add permission check for using API
|
| 3d59480dc201c893c6da5c3934b14a2d95a1bef9 |
10-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
am ea445758: am 08faac3c: Unhide SIP API.
Merge commit 'ea445758efba6b728d5e597402e9d9538f3ef451'
* commit 'ea445758efba6b728d5e597402e9d9538f3ef451':
Unhide SIP API.
|
| 2473f94718d8d7f2c8f0eb4a705816829952509b |
10-Oct-2010 |
Chung-yih Wang <cywang@google.com> |
am 4b7ff734: am 4483232f: Suppress harder for echo without affecting the volume of real voice.
Merge commit '4b7ff734611666a68471c97fabb6f516efab25cd'
* commit '4b7ff734611666a68471c97fabb6f516efab25cd':
Suppress harder for echo without affecting the volume of real voice.
|
| c7e4b2d5bb9c9ab2d4a8efa0fd07be9d83987d36 |
09-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 841d6ff9: am 62ec9834: Merge "Make SipService broadcast SIP_SERVICE_UP when it\'s up." into gingerbread
Merge commit '841d6ff9e05daccbc60daa1618a27e9db3a4fb32'
* commit '841d6ff9e05daccbc60daa1618a27e9db3a4fb32':
Make SipService broadcast SIP_SERVICE_UP when it's up.
|
| 1d8db8a0a83f3a09dd74afa3070df8bf4b8a6962 |
09-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 909a974f: am 16c29bd7: Merge "SIP: Fix busy authentication loop." into gingerbread
Merge commit '909a974f8e5df4b6fc2cf8de6c64633406095c6e'
* commit '909a974f8e5df4b6fc2cf8de6c64633406095c6e':
SIP: Fix busy authentication loop.
|
| c7fda188aee215f1842111f5b9f711f7b844b4d4 |
09-Oct-2010 |
Chung-yih Wang <cywang@google.com> |
Do not release the wifi lock if the screen is off.
We need to be able to receive calls if the device is able to
reassociate with any AP later on.
Change-Id: Ib7aafb98386bf250ed9b5ec0a5b519594efa1649
ava/com/android/server/sip/SipService.java
|
| 7ff6f97f84b3df7d1f2c5e730189d26ba35bc688 |
09-Oct-2010 |
Chung-yih Wang <cywang@google.com> |
resolved conflicts for merge of 4790a2e2 to master
Change-Id: I659ccd9a51e24f217f715178a98eaf6592c258d7
|
| aa562ffdb8f728569e6957b742f271eb7303f878 |
08-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
SipService: add permission check for using API
Change-Id: Ifd85ba07f1b913011cb3e80e5027c67bfe3db280
ava/com/android/server/sip/SipService.java
|
| 08faac3c26e12863858e1534985dd950193f755f |
15-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
Unhide SIP API.
Change-Id: I09468e3149a242a3b1e085ad220eb74f84ac6c68
ava/android/net/sip/SipAudioCall.java
ava/android/net/sip/SipErrorCode.java
ava/android/net/sip/SipException.java
ava/android/net/sip/SipManager.java
ava/android/net/sip/SipProfile.java
ava/android/net/sip/SipRegistrationListener.java
ava/android/net/sip/SipSession.java
|
| 4483232f57ebdc444bf045120c302235a211e737 |
08-Oct-2010 |
Chung-yih Wang <cywang@google.com> |
Suppress harder for echo without affecting the volume of real voice.
Change-Id: Ia3ce98eedd487a9e879ff0a4907b8c15b5707429
ni/rtp/EchoSuppressor.cpp
|
| 9db99a4dc10ac0d5d3751f03ea51c0fed217d2f8 |
07-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
Make SipService broadcast SIP_SERVICE_UP when it's up.
http://b/issue?id=3062010
Change-Id: I13419fa3a8fdfba1977260f703e4dcaa42a6606c
ava/android/net/sip/SipManager.java
ava/com/android/server/sip/SipService.java
|
| 16c29bd7f2ce147fd2f39f1f36df88d1b04a5387 |
07-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
Merge "SIP: Fix busy authentication loop." into gingerbread
|
| f209cd70623f837026fb6c41e40a421291be62d0 |
07-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
am a785a59c: am 718e0033: Merge "SIP: add SERVER_UNREACHABLE error code." into gingerbread
Merge commit 'a785a59c831256f274627f8f8eb77f9d54508916'
* commit 'a785a59c831256f274627f8f8eb77f9d54508916':
SIP: add SERVER_UNREACHABLE error code.
|
| 828c89ba8e76224bc61702fa3b9c093300825ba0 |
07-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 3cb2d3be: am 1862af57: Merge "SipService: supply PendingIntent when open a profile." into gingerbread
Merge commit '3cb2d3be6cb501c77c7a5765d954363125857cca'
* commit '3cb2d3be6cb501c77c7a5765d954363125857cca':
SipService: supply PendingIntent when open a profile.
|
| ee8a884f3504c981be8a1d6888b4590a0a394e05 |
06-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
SIP: Fix busy authentication loop.
Add a retry count and give up after two attempts.
Also stop auto registration when server is unreachable.
And rename onError() to restartLater() for better readability.
http://b/issue?id=3066573
Change-Id: Icfa65c58546a1e2bf8e59e29584a3926c53c479b
ava/com/android/server/sip/SipService.java
ava/com/android/server/sip/SipSessionGroup.java
|
| fd1e4ad52ca5c7371538390c7debe36492542051 |
06-Oct-2010 |
Marco Nelissen <marcone@google.com> |
Fix simulator build, part 1/n
Change-Id: If0a42ab262ee6aa6381ce95bd49baf232adb01c5
ni/rtp/AmrCodec.cpp
ni/rtp/EchoSuppressor.cpp
|
| fb116fbea3b5f4ea725c0c3e82e70fd82e0a45c5 |
06-Oct-2010 |
Chung-yih Wang <cywang@google.com> |
Misc fixes for sim-eng build.
Change-Id: I0c5dac1097abc924e66dab92d7d03d5051b4fd29
ni/rtp/AmrCodec.cpp
ni/rtp/EchoSuppressor.cpp
|
| 718e0033e69fa7d1db12242324ab9098ac430bf5 |
05-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
Merge "SIP: add SERVER_UNREACHABLE error code." into gingerbread
|
| c6548fd9eda7b58f5a2e2a9c01e3c7cafd42fafb |
05-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
SIP: add SERVER_UNREACHABLE error code.
Let SipSession return it when UnknownHostException is caught.
Add DisconnectCause.SERVER_UNREACHABLE in Connection and have SipPhone report
it when receiving SERVER_UNREACHABLE from SipSession.
http://b/issue?id=3061691
Change-Id: I944328ba3ee30c0a9386e89b5c4696d4d9bde000
ava/android/net/sip/SipErrorCode.java
ava/com/android/server/sip/SipSessionGroup.java
|
| 323d3671ac813df8dd173f3f4d6cb681ee29f740 |
05-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
SipService: supply PendingIntent when open a profile.
The SipService used to take an action string and broadcasts an intent with
that action string when an incoming call is received. The design is not safe
(as the intent may be sniffed) and inflexible (can only received by
BroadcastReceiver). Now we use PendingIntent to fix all these.
Companion CL: https://android-git.corp.google.com/g/#change,71800
Change-Id: Id12e5c1cf9321edafb171494932cd936eae10b6e
ava/android/net/sip/ISipService.aidl
ava/android/net/sip/SipManager.java
ava/com/android/server/sip/SipService.java
|
| e1ccf7c953c22a3e96a9e7f9483901d84c8c5f4c |
04-Oct-2010 |
Chia-chi Yeh <chiachi@android.com> |
am fbd5a59d: am 4fc04f16: Merge "RTP: Add a baseline echo suppressor." into gingerbread
Merge commit 'fbd5a59da9a455bc1c54a80bd5b3afeb426a8e3d'
* commit 'fbd5a59da9a455bc1c54a80bd5b3afeb426a8e3d':
RTP: Add a baseline echo suppressor.
|
| 4fc04f160f5ad99ce618084c689b239a2644deca |
04-Oct-2010 |
Chia-chi Yeh <chiachi@android.com> |
Merge "RTP: Add a baseline echo suppressor." into gingerbread
|
| a8a10096a1501e901676632d78f699cdebe9f4f6 |
04-Oct-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Add a baseline echo suppressor.
Change-Id: I832f1f572f141fd928afe671b12d0b59f2a8e0b1
ni/rtp/Android.mk
ni/rtp/AudioGroup.cpp
ni/rtp/EchoSuppressor.cpp
ni/rtp/EchoSuppressor.h
|
| 51d2adab83837425dae8062b7ff2a5bd1e732dd9 |
04-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 1f34ffd7: am 5cab38ba: Merge "SIP: minor fixes." into gingerbread
Merge commit '1f34ffd7e36de5d1a12d4a3901c3ac4e4e56cb99'
* commit '1f34ffd7e36de5d1a12d4a3901c3ac4e4e56cb99':
SIP: minor fixes.
|
| 9ea96c6cade1f25d4d77dcbd24854df431548b36 |
03-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
SIP: minor fixes.
+ Log error instead of crashing app process in SipManager's ListenerRelay.
+ Terminate dialog and transaction in SipSessionGroup.reset().
+ Remove redundant reset() in SipSessionGroup.
Change-Id: Ifbf29d2c9607ffe1a1a50b0c131ee3a4e81a0d0e
ava/android/net/sip/SipManager.java
ava/com/android/server/sip/SipSessionGroup.java
|
| 82fb4ef335ec63a519a0658cea233ed8a3265020 |
01-Oct-2010 |
Chia-chi Yeh <chiachi@android.com> |
am c38e6ae4: am 274e3b5d: Merge "RTP: Start AudioRecord before AudioTrack to avoid being disabled." into gingerbread
Merge commit 'c38e6ae40096ec60bc42de663f39dc061b9c90f4'
* commit 'c38e6ae40096ec60bc42de663f39dc061b9c90f4':
RTP: Start AudioRecord before AudioTrack to avoid being disabled.
|
| 1041bb7eec5163781442624fdc2fef42b41f7c54 |
01-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 9af6b536: am 063d02bb: Merge "SipService: turn off verbose logging" into gingerbread
Merge commit '9af6b53676061db6fc9c18300dc7d8258f7306ab'
* commit '9af6b53676061db6fc9c18300dc7d8258f7306ab':
SipService: turn off verbose logging
|
| 274e3b5d752d1a4fd05352e50ad098d2f5b49a36 |
01-Oct-2010 |
Chia-chi Yeh <chiachi@android.com> |
Merge "RTP: Start AudioRecord before AudioTrack to avoid being disabled." into gingerbread
|
| 063d02bb03a9260314b529490866528433148738 |
01-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
Merge "SipService: turn off verbose logging" into gingerbread
|
| 67ecb5b90c7d944a485ed35f3e968ab0ae49f5b4 |
01-Oct-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Start AudioRecord before AudioTrack to avoid being disabled.
Change-Id: I96be89fda41d77e2cf5bfc1c2f14e2b109001b57
ni/rtp/AudioGroup.cpp
|
| b031957d528840df0ccbd28651ecbf3c64d42718 |
01-Oct-2010 |
Hung-ying Tyan <tyanh@google.com> |
SipService: turn off verbose logging
Change-Id: I264662ba17d215d532f58b6ee793e569fe67c334
ava/com/android/server/sip/SipService.java
|
| 2e88d0c4bc22412645d115945683ae6d7d2a33e3 |
30-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
am 2b133fc0: am 21ae1ad6: RTP: Minor fixes with polishing.
Merge commit '2b133fc07533a853f7de23da4f60a766f4233bed'
* commit '2b133fc07533a853f7de23da4f60a766f4233bed':
RTP: Minor fixes with polishing.
|
| e0ed9dbcb8f3b67f66a1b2a1df264e3aee0bb81c |
30-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
am c79e74ec: am d29e0754: Merge "Add uri field to SipManager.ListenerRelay" into gingerbread
Merge commit 'c79e74ec1d30f95de89568ee645a9b1577ae73b3'
* commit 'c79e74ec1d30f95de89568ee645a9b1577ae73b3':
Add uri field to SipManager.ListenerRelay
|
| bf45f19435851c8c578d6edabb761e1f4c51da8e |
30-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
am f6381ec1: am dfd1484e: Merge "RTP: Adjust the jitter buffer to 512ms." into gingerbread
Merge commit 'f6381ec1da1166e3350d897faae654afb2c21a5a'
* commit 'f6381ec1da1166e3350d897faae654afb2c21a5a':
RTP: Adjust the jitter buffer to 512ms.
|
| f3da1ea405e6a3f908a9e7773c7650cdb305f4dd |
30-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 34552149: am 6a53489a: SipService: add UID check.
Merge commit '34552149e4b997d4ed3383fc153faff2bb189066'
* commit '34552149e4b997d4ed3383fc153faff2bb189066':
SipService: add UID check.
|
| a77c9541d008cfffed71cb8e3a9382001cf7fe9c |
30-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
am cbee6229: am 0a537b78: Merge "RTP: Enable AMR codec." into gingerbread
Merge commit 'cbee622954de5e9e0c07557f8ec9aaa741110043'
* commit 'cbee622954de5e9e0c07557f8ec9aaa741110043':
RTP: Enable AMR codec.
|
| d161479237010cb2b7bc8dab0fbbce2cf0170ecf |
30-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 947d2abd: am 2365b78e: Merge "SIP: misc fixes." into gingerbread
Merge commit '947d2abd82ef68c661fc29fd5167e4c0ba749f71'
* commit '947d2abd82ef68c661fc29fd5167e4c0ba749f71':
SIP: misc fixes.
|
| 5a7c6d298e9f8963e3b82f84da15f16a4a83f8ff |
30-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
am 1c2eab2d: am 955ab37c: Merge "RTP: Enable GSM-EFR codec." into gingerbread
Merge commit '1c2eab2d86faa9c647a9893f761a50cfa28d9d57'
* commit '1c2eab2d86faa9c647a9893f761a50cfa28d9d57':
RTP: Enable GSM-EFR codec.
|
| 21ae1ad6a695d6f1f253797fcf2a77b975b82cd3 |
30-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Minor fixes with polishing.
Change-Id: I50641373989e512fb489b5017edbcfd7848fe8b9
ni/rtp/AudioGroup.cpp
ni/rtp/G711Codec.cpp
ni/rtp/RtpStream.cpp
|
| d29e0754183e4b3945eb7cabae91cd3df47ae4d6 |
30-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
Merge "Add uri field to SipManager.ListenerRelay" into gingerbread
|
| 9e1d308e993d451882456e44cfaacae63df7a496 |
30-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
Add uri field to SipManager.ListenerRelay
in case mSession is not available.
Change-Id: Ifee2c129e48aa1177f648f176413ab6aa5606770
ava/android/net/sip/SipManager.java
|
| dfd1484e3ba9c305730ccb39859919ca0e97d720 |
30-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
Merge "RTP: Adjust the jitter buffer to 512ms." into gingerbread
|
| 3520bd43139f4571cf96af126dba13681633bcb0 |
30-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Adjust the jitter buffer to 512ms.
Change-Id: Ia91c1aa1a03b65dbd329ea98383f370844e2b0c0
ni/rtp/AudioGroup.cpp
|
| 8ff07224535602884e5e92b954bb3e38b67c7c19 |
30-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
am 1254b9c5: am cd386649: Merge "RTP: Revise the workaround of private addresses and fix bugs." into gingerbread
Merge commit '1254b9c534c5f027f8928fbb3e743e57d55bd13d'
* commit '1254b9c534c5f027f8928fbb3e743e57d55bd13d':
RTP: Revise the workaround of private addresses and fix bugs.
|
| 6a53489ae594d7cc373a00687d6ea2f23d0634df |
13-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SipService: add UID check.
Only allow creator or radio user to access profiles.
Change-Id: I548938f117926bcc878419142d1b5d818a4e70df
ava/com/android/server/sip/SipService.java
|
| 0a537b78d3fb4db86411d745b2696459d6b98ef6 |
30-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
Merge "RTP: Enable AMR codec." into gingerbread
|
| 2365b78e64feaa9527efb15bf4ac207a837f2b45 |
30-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
Merge "SIP: misc fixes." into gingerbread
|
| f88fc1fa907f720df4a3e915509e688e9e4cf1f8 |
30-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Enable AMR codec.
Change-Id: I49e6bdc1b67306b44173f2f346f8372a50264870
ava/android/net/rtp/AudioCodec.java
ni/rtp/AmrCodec.cpp
ni/rtp/AudioCodec.cpp
|
| fb3a98b1d8d0ad040980d509c4c5341928b9460b |
30-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SIP: misc fixes.
+ Fix keepalive timer event leak due to the race between stopping timer and
the async'ed timeout handler
+ SipSessionImpl: set state before handling an event to ensure we get correct
state when some error occurs during handling the event.
+ Fix potential NPE in SipManager.ListenerRelay.getUri().
Change-Id: I021ee34f83059fd4fbb64b30bea427a5462aa51b
ava/android/net/sip/SipManager.java
ava/com/android/server/sip/SipService.java
ava/com/android/server/sip/SipSessionGroup.java
|
| f4ae94229d736c7dbd3c5c36d484213d51545702 |
29-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Enable GSM-EFR codec.
Change-Id: I9d84009e4557a0a82c1f9d7d543922741be97c77
ava/android/net/rtp/AudioCodec.java
ni/rtp/AmrCodec.cpp
ni/rtp/Android.mk
ni/rtp/AudioCodec.cpp
|
| fe5298992a52f93bb8365d345cdd82d88a4b49f2 |
29-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Revise the workaround of private addresses and fix bugs.
Change-Id: Ie654b569f47049aa452eca8d3e6d4a98ac18469c
ni/rtp/AudioGroup.cpp
|
| dcf2be6cf660269c77f51ff0e0f336726d1625c6 |
29-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
am ebfe5632: am e006e4d2: Merge changes Iae1913fb,I38dbefef into gingerbread
Merge commit 'ebfe5632db275a89b49ab828064ba90db59702cf'
* commit 'ebfe5632db275a89b49ab828064ba90db59702cf':
RTP: Enable GSM codec.
RTP: Refactor out G711 codecs into another file.
|
| e006e4d2c91c50795df0a02366b31610b4e97cb1 |
29-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
Merge changes Iae1913fb,I38dbefef into gingerbread
* changes:
RTP: Enable GSM codec.
RTP: Refactor out G711 codecs into another file.
|
| a6f950c9682ffffc00ca976aafeeedf391718b1d |
29-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Enable GSM codec.
Change-Id: Iae1913fb0643f1c66b5d16f24d51924d363e5ef5
ava/android/net/rtp/AudioCodec.java
ni/rtp/Android.mk
ni/rtp/AudioCodec.cpp
ni/rtp/GsmCodec.cpp
|
| 9783052ec1260985f4e08f4dada6331445ce538e |
29-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
am df31e03c: am 320cdcb1: Merge "RTP: Delay the initialization of AudioTrack and AudioRecord." into gingerbread
Merge commit 'df31e03c47cd4caf45d8a58cf1fe5893da48ec6c'
* commit 'df31e03c47cd4caf45d8a58cf1fe5893da48ec6c':
RTP: Delay the initialization of AudioTrack and AudioRecord.
|
| 0b3968ae539f084a7b29afffcd9dc3f02e81c811 |
29-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 0d447760: am 6d028dd2: Merge "SIP: Feedback any provisional responses in addition to RING" into gingerbread
Merge commit '0d44776016cecf1c7c826c4784f8f867a56235f0'
* commit '0d44776016cecf1c7c826c4784f8f867a56235f0':
SIP: Feedback any provisional responses in addition to RING
|
| 78c11b3cf170fdd35ff6984bc2a64c01e2457503 |
28-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Refactor out G711 codecs into another file.
Change-Id: I38dbefef2315a28d44683e86a51e69f38e3f20ec
ni/rtp/Android.mk
ni/rtp/AudioCodec.cpp
ni/rtp/G711Codec.cpp
|
| 320cdcb122505ba703326a102f9b13d2f2f8847a |
28-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
Merge "RTP: Delay the initialization of AudioTrack and AudioRecord." into gingerbread
|
| 9083c84af1742cfc9228add21ec72310e67e6086 |
28-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Delay the initialization of AudioTrack and AudioRecord.
Related to http://b/3043844.
Change-Id: I2c4fd9f64e6eba597d68b2ea1ceeff83103697db
ni/rtp/AudioGroup.cpp
|
| 6c6eacda8066728537f2d8828e4c123f91ddfc27 |
28-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
am f7e13400: am 624d5b4e: SIP: add DisconnectCause.SERVER_ERROR
Merge commit 'f7e13400b24bdc5461e640cfb2c9cb2bbb2b6236'
* commit 'f7e13400b24bdc5461e640cfb2c9cb2bbb2b6236':
SIP: add DisconnectCause.SERVER_ERROR
|
| 6057cd00d95c756b78f22c67279cb982bc0674ef |
28-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SIP: Feedback any provisional responses in addition to RING
The only exception is TRYING.
Also remove an unused import in SipSessionGroup.
http://b/issue?id=3021865
Change-Id: I160982b0c4b417362f1fb961217db90c3a585ce5
ava/com/android/server/sip/SipSessionGroup.java
|
| 624d5b4e8c20516516d0bff74479b9f5abdfe61c |
28-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SIP: add DisconnectCause.SERVER_ERROR
and fix how SipErrorCode.SERVER_ERROR is determinted from server response, not
from local exceptions.
http://b/issue?id=3041332
Change-Id: Idce67e29858d5c7573b98b7fa1fac074913d71d6
ava/com/android/server/sip/SipSessionGroup.java
|
| a57afb6a6c9f4fb451535c3d6f49c3bdf4b59125 |
27-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
resolved conflicts for merge of 2a36a778 to master
Change-Id: Ia70adeef06afddd29c827405fb5657bf9f5a29a3
|
| 7e54ef71db3320a751571bba5259fba816399421 |
25-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
Move SipService out of SystemServer to phone process.
Companion CL: https://android-git/g/#change,70187
http://b/issue?id=2998069
Change-Id: I90923ac522ef363a4e04292f652d413c5a1526ad
ava/com/android/server/sip/SipHelper.java
ava/com/android/server/sip/SipService.java
ava/com/android/server/sip/SipSessionGroup.java
ava/com/android/server/sip/SipSessionListenerProxy.java
|
| 5a474a2bb8bc23fcc8d05e8b9ec3f4306dd63db1 |
27-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
am 44669d31: am fd144d76: Merge "SipAudioCall: remove SipManager dependency." into gingerbread
Merge commit '44669d31d1d5b094d7b7d3e393281440ea0c9536'
* commit '44669d31d1d5b094d7b7d3e393281440ea0c9536':
SipAudioCall: remove SipManager dependency.
|
| 031d8786824a385fa47750e5e8aa75f40d70cae9 |
27-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
am fe2d279c: am 00a22064: SipService: handle cross-domain authentication error
Merge commit 'fe2d279c5ef571340f20d433badd9f68072299af'
* commit 'fe2d279c5ef571340f20d433badd9f68072299af':
SipService: handle cross-domain authentication error
|
| fd144d7667d9d050b7fb158276ae4623d4ea83b8 |
27-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
Merge "SipAudioCall: remove SipManager dependency." into gingerbread
|
| 00a22064efef4f574e439079aae2deae1a087a31 |
25-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SipService: handle cross-domain authentication error
and add new CROSS_DOMAIN_AUTHENTICATION error code and OUT_OF_NETWORK
DisconnectCause.
http://b/issue?id=3020185
Change-Id: Icc0a341599d5a72b7cb2d43675fbddc516544978
ava/android/net/sip/SipErrorCode.java
|
| 5e18ad0c53faf88357c83bae66ab9d04c0388bb9 |
27-Sep-2010 |
Chung-yih Wang <cywang@google.com> |
am 4a04a312: am bd229420: Fix the unhold issue especially if one is behind NAT.
Merge commit '4a04a3129bd30a996dd302b982aeca8f228f57e8'
* commit '4a04a3129bd30a996dd302b982aeca8f228f57e8':
Fix the unhold issue especially if one is behind NAT.
|
| bd2294204e3edaede3fe81eb9b11c05c4fafe627 |
23-Sep-2010 |
Chung-yih Wang <cywang@google.com> |
Fix the unhold issue especially if one is behind NAT.
+call startAudio() when call is established.
Change-Id: Ib6a1e34017fb83007ce275da1991058e8b803833
ni/rtp/AudioGroup.cpp
|
| 3a4197e642e9c70f1fe00c2cba30f0f957d36bfc |
24-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SipAudioCall: remove SipManager dependency.
Change-Id: I2dc8bf427e52f64529ee0e0261362b975a8917c6
ava/android/net/sip/SipAudioCall.java
ava/android/net/sip/SipManager.java
|
| a97c5f7779bbd53896b5312c9dd04c505511781d |
24-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
Merge "fix build"
|
| fb0264096e08aeeb350c9a2762b34d14361ba38e |
24-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
fix build
Change-Id: Iff05b5ea7f535f532eec2af1edf78fdf8acfa21c
ava/android/net/sip/SipAudioCall.java
|
| 658bec956785e074edc4f6c9fe739c366e37be33 |
23-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
SDP: remove dead code.
Change-Id: I2a5764a2b9cabc54b0ac18666e494c1cb39c4e9b
ava/android/net/sip/SdpSessionDescription.java
ava/android/net/sip/SessionDescription.aidl
ava/android/net/sip/SessionDescription.java
|
| 84a357bb6a8005e1c5e924e96a8ecf310e77c47c |
15-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
Refactoring SIP classes to get ready for API review.
+ replace SipAudioCall and its Listener interfaces with real implementations,
+ remove SipAudioCallImpl.java, most of it is has become part of SipAudioCall,
+ add SipSession and its Listener classes to wrap ISipSession and ISipSessionListener,
+ move SipSessionState to SipSession.State,
+ make SipManager keep context and remove the context argument from many methods of its,
+ rename SipManager.getInstance() to newInstance(),
+ rename constant names for action strings and extra keys to follow conventions,
+ set thread names for debugging purpose.
Change-Id: Ie1790dc0e8f49c06c7fc80d33fec0f673a9c3044
ava/android/net/sip/SipAudioCall.java
ava/android/net/sip/SipAudioCallImpl.java
ava/android/net/sip/SipManager.java
ava/android/net/sip/SipProfile.java
ava/android/net/sip/SipSession.java
ava/android/net/sip/SipSessionState.java
|
| 0b7d6de1559a4a78af76ab501e0a15afc396c2b9 |
23-Sep-2010 |
repo sync <chiachi@android.com> |
Fix the build.
Change-Id: I82210cb2d41f532583f83ea17e6f2d8d49280a30
ava/android/net/sip/SimpleSessionDescription.java
|
| 84f7f6ba3913a4ad8546d425197a6d64593b91cf |
23-Sep-2010 |
repo sync <chiachi@android.com> |
SIP: Make SipAudioCallImpl use SimpleSessionDescription instead of javax.sdp.
Change-Id: I7efff4f29ca84c3e7c17ef066b7186b514a777b2
ava/android/net/sip/SipAudioCallImpl.java
|
| e6c0c109588771a97aba51d06fdf73557b06dfd3 |
20-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
SDP: Add a simple class to help manipulate session descriptions.
Change-Id: I1631ee20e8b4a9ad8e2184356b5d13de66e03db1
ava/android/net/sip/SimpleSessionDescription.java
|
| 7a69aeffda29bd1a7ebc5993eeb4e9ee224f096a |
22-Sep-2010 |
repo sync <chiachi@android.com> |
RTP: Add log throttle for "no data".
Change-Id: I14d9886a40fa780514cbc6c5bac6fb2a670f55f4
ni/rtp/AudioGroup.cpp
|
| 4033a67d0e99d422336574fc5c982d349632b117 |
16-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Update native part to reflect the API change.
Change-Id: Ic2858920ad77d7312f2429f89ca509a481363431
ni/rtp/AudioCodec.cpp
ni/rtp/AudioCodec.h
ni/rtp/AudioGroup.cpp
|
| 37adc522f6bc074a688ffbef420a8627ef9a4b5b |
21-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Add two getters to retrieve the current configuration from AudioStream.
Change-Id: Iff588130653242f6ddd6a6b663df775ecb276768
ava/android/net/rtp/AudioStream.java
|
| 32e106b7bdd57c82ee67705871f6116d92bce79b |
16-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Extend codec capability and update the APIs.
Change-Id: I37ba9d83c2de3c5dae2bfc1b7513df7f6fee3c5c
ava/android/net/rtp/AudioCodec.java
ava/android/net/rtp/AudioGroup.java
ava/android/net/rtp/AudioStream.java
ava/android/net/rtp/RtpStream.java
|
| 8544560ccc43de7ff49d91866f461f5572f0b147 |
20-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SipPhone: fix missing-call DisconnectCause feedback
also fix delivering bad news before closing a SipAudioCallImpl object so that
apps can get the current audio-call object state before it's closed:
http://b/issue?id=3009262
Change-Id: I94c19dae8b4f252de869e614ec462b19b4ff2077
ava/android/net/sip/SipAudioCallImpl.java
|
| 97963794af1e18674dd111e3ad344d90b16c922c |
17-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SIP: convert enum to static final int.
Converts SipErrorCode and SipSessionState.
Change-Id: Iee3a465649ea89d395b2336bbd673c25113e5f93
ava/android/net/sip/ISipSession.aidl
ava/android/net/sip/ISipSessionListener.aidl
ava/android/net/sip/SipAudioCall.java
ava/android/net/sip/SipAudioCallImpl.java
ava/android/net/sip/SipErrorCode.java
ava/android/net/sip/SipManager.java
ava/android/net/sip/SipRegistrationListener.java
ava/android/net/sip/SipSessionAdapter.java
ava/android/net/sip/SipSessionState.java
|
| c4b87477c076d61062950becc132b7483e3fb198 |
19-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SIP: add config flag for wifi-only configuration.
http://b/issue?id=2994029
Change-Id: I328da9b0f8b70d660dbcefffdac8250341792101
ava/android/net/sip/SipManager.java
|
| afa583e6557557577188c3e40146ac8d6f2aa7c7 |
17-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SipAudioCall: expose startAudio()
so that apps can start audio when time is right.
Change-Id: I7ae96689d3a8006b34097533bc2434bc3814b82a
ava/android/net/sip/SipAudioCall.java
ava/android/net/sip/SipAudioCallImpl.java
|
| 9352cf1a4d46492fc48a20f7d825a9bcb6e8b365 |
16-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
Add timer to SIP session creation process.
+ add timer parameter to ISipSession.make/changeCall(),
+ add timer paramter to SipAudioCall.make/answer/hold/continueCall()'s,
+ add timer parameter to SipManager.makeAudioCall(),
+ modify implementation in SipSessionGroup, SipAudioCallImpl accordingly,
+ make SipPhone to use it with 8-second timeout.
http://b/issue?id=2994748
Change-Id: I661a887e5810087ddc5e2318335e2fa427f80ec6
ava/android/net/sip/ISipSession.aidl
ava/android/net/sip/SipAudioCall.java
ava/android/net/sip/SipAudioCallImpl.java
ava/android/net/sip/SipManager.java
|
| 286bb5a00bdb9f0cb0815aef441ec72f231c84ea |
14-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
Fix links in SIP API javadoc.
Change-Id: I839280fe18502bb576f6e9c9a7948077c02fa570
ava/android/net/sip/ISipSession.aidl
ava/android/net/sip/SipAudioCall.java
ava/android/net/sip/SipErrorCode.java
ava/android/net/sip/SipException.java
ava/android/net/sip/SipManager.java
ava/android/net/sip/SipProfile.java
ava/android/net/sip/SipRegistrationListener.java
|
| ae076d3981fda732d54b6c6e37e5659b2e7ba130 |
15-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SIP: add PEER_NOT_REACHABLE error feedback.
http://b/issue?id=3002033
Change-Id: Ib64b08919d214acbab89945ac19dc113a68e62ad
ava/android/net/sip/SipErrorCode.java
|
| 12bec5ddf58ad3a69728810480e6194c806567d6 |
14-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SipService: ignore connect event for non-active networks.
+ sanity check and remove redundant code.
Change-Id: I4d3e226851ad7fc4d88ddcd0a5c58f7e33b6c14a
ava/android/net/sip/SipAudioCallImpl.java
|
| 13f6270eb14b409709c936b828e2a2fd40e427c4 |
14-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SipAudioCall: use SipErrorCode instead of string in onError()
and fix callback in setListener().
Change-Id: Ic2622df992a2ad45cb1e3f71736f320897ae8fb3
ava/android/net/sip/SipAudioCall.java
ava/android/net/sip/SipAudioCallImpl.java
|
| 99bf4e45c4566172189735b34b368b76660ca57a |
14-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SIP: remove dependency on javax.sip
and change errorCodeString to errorCode in
SipRegistrationListener.onRegistrationFailed().
Change-Id: Id9618f5a4b0effaed04f8b0dc60347499d9e4501
ava/android/net/sip/ISipSession.aidl
ava/android/net/sip/SipManager.java
ava/android/net/sip/SipProfile.java
ava/android/net/sip/SipRegistrationListener.java
|
| d231aa880ab006d51ffe03454c1fc082f1c97bb8 |
13-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SipService: deliver connectivity change to all sessions.
+ add DATA_CONNECTION_LOST to SipErrorCode
+ convert it to Connection.DisconnectCause.LOST_SIGNAL in SipPhone
http://b/issue?id=2992548
Change-Id: Ie8983c1b81077b21f46304cf60b8e61df1ffd241
ava/android/net/sip/SipErrorCode.java
|
| 3d7606aa607b24817e37c264f2141ed7b2d50be0 |
12-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SIP: enhance timeout and registration status feedback.
http://b/issue?id=2984419
http://b/issue?id=2991065
Change-Id: I2d3b1dd3a70079ff347f7256f4684aea07847f4e
ava/android/net/sip/SipErrorCode.java
|
| 25b52a2f97df112c2836972d0b6d9a4c7a9c4a4e |
13-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SIP: remove dependency on javax.sip.SipException.
Change-Id: I77d289bef1b5e7f1ec0c0408d0bbf96c21085cd7
ava/android/net/sip/SipAudioCall.java
ava/android/net/sip/SipAudioCallImpl.java
ava/android/net/sip/SipException.java
ava/android/net/sip/SipManager.java
|
| 903e1031605d715e904811b0dd06cc6a518f0048 |
09-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SIP: add SipErrorCode for error feedback.
Change-Id: I8b071d4933479b780a403d0bfa30511f4c23ca8f
ava/android/net/sip/ISipSessionListener.aidl
ava/android/net/sip/SipAudioCall.java
ava/android/net/sip/SipAudioCallImpl.java
ava/android/net/sip/SipErrorCode.java
ava/android/net/sip/SipRegistrationListener.java
ava/android/net/sip/SipSessionAdapter.java
|
| f6936a3a52b6bb6de27f75d4e38d116e896b7f4d |
09-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
Merge "RTP: prevent buffer overflow in AudioRecord." into gingerbread
|
| 557b04de23238fb496b5ca58e21331c842e95660 |
08-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: prevent buffer overflow in AudioRecord.
This change simply reduces the receive timeout of DeviceSocket. It works
because AudioRecord will block us till there is enough data, which makes
AudioSocket overlap AudioRecord.
Change-Id: I4700224fb407e148ef359a9d99279e10240128d0
ni/rtp/AudioGroup.cpp
|
| 643fce978152c6c5ded316a8c9de6531b7d4cee7 |
03-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SipManager: always return true for SIP API and VOIP support query.
Change-Id: I397a804e0aa598aee77a8ce28ada1b11e10fbaea
http://b/issue?id=2972054
ava/android/net/sip/SipManager.java
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| dc296b0d4bd6fef8764c10fb4cd59c85bc5186f6 |
02-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
Merge "SipService: reduce the usage of javax.sdp.*." into gingerbread
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| 95b15c35608fe3ea679c8a478c6cbd841623371e |
02-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
SipService: reduce the usage of javax.sdp.*.
After this change, SipAudioCallImpl is the only place still using it.
Change-Id: I5693bffa54f9e19cbfa70b45dfcf40fba04dedbb
ava/android/net/sip/ISipSession.aidl
ava/android/net/sip/ISipSessionListener.aidl
ava/android/net/sip/SdpSessionDescription.java
ava/android/net/sip/SipAudioCall.java
ava/android/net/sip/SipAudioCallImpl.java
ava/android/net/sip/SipManager.java
ava/android/net/sip/SipSessionAdapter.java
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| 60264b306453a3043442719b970f2edb3f46f51b |
01-Sep-2010 |
Hung-ying Tyan <tyanh@google.com> |
SipProfile: remove outgoingCallAllowed flag.
Change-Id: I37a215bafce57adf6911c81fd38db324bac686ec
ava/android/net/sip/SipProfile.java
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| 3424c02e6b931a8bbd651ae75217bebd008b2605 |
27-Aug-2010 |
Hung-ying Tyan <tyanh@google.com> |
Add software features for SIP and VOIP
and block SipService creation and SIP API if the feature is not available.
Change-Id: Icf780af1ac20dda4d8180cea3e5b20e21a8350bc
ava/android/net/sip/SipManager.java
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| 0858806ffcb9ff34725abb79106aa1de27d1bf60 |
26-Aug-2010 |
Chung-yih Wang <cywang@google.com> |
Add Wifi High Perf. mode during a call.
To prevent the wifi from entering low-power mode due to the screen off
triggered by the proximity sensor.
Change-Id: I490bc594d800bc30c256e52ef3bce08bf86bc7b1
ava/android/net/sip/SipAudioCallImpl.java
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| 14e00621c81da6a0391da47afce77945b27c7231 |
26-Aug-2010 |
Chia-chi Yeh <chiachi@android.com> |
Merge "Revert "RTP: integrate the echo canceller from speex."" into gingerbread
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| 7fa7ee11f6c274903241897c284337ba8b158988 |
26-Aug-2010 |
Chia-chi Yeh <chiachi@android.com> |
Revert "RTP: integrate the echo canceller from speex."
This reverts commit 4ae6ec428f3570b9020b35ada6a62f94af66d888.
ni/rtp/Android.mk
ni/rtp/AudioGroup.cpp
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| 5424c8dcacf1c227fe7deb0185510614122ab447 |
25-Aug-2010 |
Chung-yih Wang <cywang@google.com> |
Add dynamic uid info for tracking the sip service usage.
Change-Id: Ibc340401b63799326b08aee6eba602a3e753b13f
ava/android/net/sip/SipProfile.java
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| 37f709aeb0424948a8f69577c6fad39dc95d7733 |
25-Aug-2010 |
Hung-ying Tyan <tyanh@google.com> |
Merge "SipProfile: add isOutgoingCallAllowed() and new builder constructor" into gingerbread
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| cf95f5d26363d4cd3815d31f5798f932a7720c17 |
23-Aug-2010 |
Hung-ying Tyan <tyanh@google.com> |
SipProfile: add isOutgoingCallAllowed() and new builder constructor
Change-Id: I7ced47079fd2b00c7160b152eb4c1d34399e39dc
ava/android/net/sip/SipProfile.java
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| 3294d44b96f63f647fba3a03604eb028e28a42bc |
18-Aug-2010 |
Hung-ying Tyan <tyanh@google.com> |
Add confcall management to SIP calls
and fix the bug of re-assigning connectTime's in SipConnection,
and adding synchronization for SipPhone to be thread-safe,
and set normal audio mode when call not on hold instead of on hold in SipAudioCallImpl,
and fix re-entrance problem in CallManager.setAudioMode() for in-call mode.
Change-Id: I54f39dab052062de1ce141e5358d892d30453a3a
ava/android/net/sip/SipAudioCall.java
ava/android/net/sip/SipAudioCallImpl.java
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| 4ae6ec428f3570b9020b35ada6a62f94af66d888 |
24-Aug-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: integrate the echo canceller from speex.
Currently the filter_length is set to one second.
Will change that when we have a better idea.
Change-Id: Ia942a8fff00b096de8ff0049a448816ea9a68068
ni/rtp/Android.mk
ni/rtp/AudioGroup.cpp
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| 2880ef86e5210832ef44f2d45c46ada1891372e5 |
24-Aug-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: reduce the latency by overlapping AudioRecord and AudioTrack.
Change-Id: I00d750ee514ef68d5b2a28bd1893417ed70ef1fc
ni/rtp/AudioGroup.cpp
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| b8790323473bef75a27d2da6fde2497b3bfe19eb |
19-Aug-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: fix few leaks when fail to add streams into a group.
Change-Id: Iefb3fe219ad48641da37a83c8d14e9ebf1d3086c
ava/android/net/rtp/AudioGroup.java
ni/rtp/AudioGroup.cpp
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| 3459d3037cc0c482a27422f1cc000b5e9d289ae8 |
18-Aug-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: remove froyo-compatible code.
Change-Id: I6822a4e4749a5909959658c29253242b4018aeb0
ni/rtp/AudioGroup.cpp
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| cfd15dd3c8554cbbcb5822a0fdf6ca31d6b28acf |
16-Aug-2010 |
Chung-yih Wang <cywang@google.com> |
Fix the IN_CALL mode issue.
If the sip call is on-holding, we should not set the audio to
MODE_NORMAL, or it will affect the audio if there is an active pstn
call.
Change-Id: If1bcba952617bf8427bc9e2d64d483ba1ee37370
ava/android/net/sip/SipAudioCall.java
ava/android/net/sip/SipAudioCallImpl.java
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| ea4de5bd25b394a1bac6f27b43c4982aace2011e |
10-Aug-2010 |
Hung-ying Tyan <tyanh@google.com> |
SipAudioCall: perform local ops before network op in endCall()
Change-Id: I1808f715d56c0979cea7741cb5bdb3831774d3ef
ava/android/net/sip/SipAudioCallImpl.java
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| 8e63ddb4c78dc4453d64ea6e94c109db703185e4 |
09-Aug-2010 |
Hung-ying Tyan <tyanh@google.com> |
SIP: clean up unused class and fields.
Change-Id: I79ed7fb324fea9a52946340055b5ea1d389a926a
ava/android/net/sip/BinderHelper.java
ava/android/net/sip/SipManager.java
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| 4c5d28cee0537c83ff0e5bc0daaae78f68dfc7c8 |
06-Aug-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: move into frameworks.
Change-Id: Ic9c17b460448c746b21526ac10b647f281ae48e9
ni/rtp/Android.mk
ni/rtp/AudioCodec.cpp
ni/rtp/AudioCodec.h
ni/rtp/AudioGroup.cpp
ni/rtp/RtpStream.cpp
ni/rtp/rtp_jni.cpp
ni/rtp/util.cpp
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| cde66df44240cfe5a7bec12ac52464c3bf26c14f |
05-Aug-2010 |
Chung-yih Wang <cywang@google.com> |
Revert "Move SIP telephony related codes to framework."
This reverts commit b631dcf3eb449ddec756bea330f4e70b996ffb9e.
ava/android/net/sip/SipManager.java
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| b631dcf3eb449ddec756bea330f4e70b996ffb9e |
05-Aug-2010 |
Chung-yih Wang <cywang@google.com> |
Move SIP telephony related codes to framework.
+ hardcode the sip service for build dependency.
Change-Id: Ib0e9717c9b87eb6e06ffa3a7b01ae31184de61bb
ava/android/net/sip/SipManager.java
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| 363c2ab82cca4f095e9e0c8465e28f6d27a24bf8 |
05-Aug-2010 |
Chung-yih Wang <cywang@google.com> |
Move the sip related codes to framework.
Change-Id: Ib81dadc39b73325c8438f078c7251857a83834fe
ava/android/net/rtp/AudioCodec.java
ava/android/net/rtp/AudioGroup.java
ava/android/net/rtp/AudioStream.java
ava/android/net/rtp/RtpStream.java
ava/android/net/sip/BinderHelper.java
ava/android/net/sip/ISipService.aidl
ava/android/net/sip/ISipSession.aidl
ava/android/net/sip/ISipSessionListener.aidl
ava/android/net/sip/SdpSessionDescription.java
ava/android/net/sip/SessionDescription.aidl
ava/android/net/sip/SessionDescription.java
ava/android/net/sip/SipAudioCall.java
ava/android/net/sip/SipAudioCallImpl.java
ava/android/net/sip/SipManager.java
ava/android/net/sip/SipProfile.aidl
ava/android/net/sip/SipProfile.java
ava/android/net/sip/SipRegistrationListener.java
ava/android/net/sip/SipSessionAdapter.java
ava/android/net/sip/SipSessionState.java
|