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gerrit-public.fairphone.software
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platform
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frameworks
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base
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3394db7d9432053e56d86b3f24e64deeb3099950
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voip
f1b1eec
Merge "SipService: mScreenOn is flipped to wrong value." into gingerbread
by Hung-ying Tyan
· 14 years ago
d6fc979
SipService: mScreenOn is flipped to wrong value.
by Hung-ying Tyan
· 14 years ago
ebc886c
Fix SipSessionGroup from throwing ConcurrentModificationException
by Hung-ying Tyan
· 14 years ago
685b61b
SipService: fix a missing switch-case break.
by Hung-ying Tyan
· 14 years ago
692cac9
SipHelper: add debug log for challenge responses.
by Hung-ying Tyan
· 14 years ago
c7fda18
Do not release the wifi lock if the screen is off.
by Chung-yih Wang
· 14 years ago
aa562ff
SipService: add permission check for using API
by Hung-ying Tyan
· 14 years ago
08faac3
Unhide SIP API.
by Hung-ying Tyan
· 14 years ago
4483232
Suppress harder for echo without affecting the volume of real voice.
by Chung-yih Wang
· 14 years ago
9db99a4
Make SipService broadcast SIP_SERVICE_UP when it's up.
by Hung-ying Tyan
· 14 years ago
16c29bd
Merge "SIP: Fix busy authentication loop." into gingerbread
by Hung-ying Tyan
· 14 years ago
ee8a884
SIP: Fix busy authentication loop.
by Hung-ying Tyan
· 14 years ago
fb116fb
Misc fixes for sim-eng build.
by Chung-yih Wang
· 14 years ago
718e003
Merge "SIP: add SERVER_UNREACHABLE error code." into gingerbread
by Hung-ying Tyan
· 14 years ago
c6548fd9
SIP: add SERVER_UNREACHABLE error code.
by Hung-ying Tyan
· 14 years ago
323d367
SipService: supply PendingIntent when open a profile.
by Hung-ying Tyan
· 14 years ago
4fc04f1
Merge "RTP: Add a baseline echo suppressor." into gingerbread
by Chia-chi Yeh
· 14 years ago
a8a1009
RTP: Add a baseline echo suppressor.
by Chia-chi Yeh
· 14 years ago
9ea96c6
SIP: minor fixes.
by Hung-ying Tyan
· 14 years ago
274e3b5
Merge "RTP: Start AudioRecord before AudioTrack to avoid being disabled." into gingerbread
by Chia-chi Yeh
· 14 years ago
063d02b
Merge "SipService: turn off verbose logging" into gingerbread
by Hung-ying Tyan
· 14 years ago
67ecb5b
RTP: Start AudioRecord before AudioTrack to avoid being disabled.
by Chia-chi Yeh
· 14 years ago
b031957
SipService: turn off verbose logging
by Hung-ying Tyan
· 14 years ago
21ae1ad
RTP: Minor fixes with polishing.
by Chia-chi Yeh
· 14 years ago
d29e075
Merge "Add uri field to SipManager.ListenerRelay" into gingerbread
by Hung-ying Tyan
· 14 years ago
9e1d308
Add uri field to SipManager.ListenerRelay
by Hung-ying Tyan
· 14 years ago
dfd1484
Merge "RTP: Adjust the jitter buffer to 512ms." into gingerbread
by Chia-chi Yeh
· 14 years ago
3520bd4
RTP: Adjust the jitter buffer to 512ms.
by Chia-chi Yeh
· 14 years ago
6a53489
SipService: add UID check.
by Hung-ying Tyan
· 14 years ago
0a537b7
Merge "RTP: Enable AMR codec." into gingerbread
by Chia-chi Yeh
· 14 years ago
2365b78
Merge "SIP: misc fixes." into gingerbread
by Hung-ying Tyan
· 14 years ago
f88fc1f
RTP: Enable AMR codec.
by Chia-chi Yeh
· 14 years ago
fb3a98b
SIP: misc fixes.
by Hung-ying Tyan
· 14 years ago
f4ae942
RTP: Enable GSM-EFR codec.
by Chia-chi Yeh
· 14 years ago
fe52989
RTP: Revise the workaround of private addresses and fix bugs.
by Chia-chi Yeh
· 14 years ago
e006e4d
Merge changes Iae1913fb,I38dbefef into gingerbread
by Chia-chi Yeh
· 14 years ago
a6f950c
RTP: Enable GSM codec.
by Chia-chi Yeh
· 14 years ago
78c11b3
RTP: Refactor out G711 codecs into another file.
by Chia-chi Yeh
· 14 years ago
320cdcb
Merge "RTP: Delay the initialization of AudioTrack and AudioRecord." into gingerbread
by Chia-chi Yeh
· 14 years ago
9083c84
RTP: Delay the initialization of AudioTrack and AudioRecord.
by Chia-chi Yeh
· 14 years ago
6057cd0
SIP: Feedback any provisional responses in addition to RING
by Hung-ying Tyan
· 14 years ago
624d5b4
SIP: add DisconnectCause.SERVER_ERROR
by Hung-ying Tyan
· 14 years ago
7e54ef7
Move SipService out of SystemServer to phone process.
by Hung-ying Tyan
· 14 years ago
fd144d7
Merge "SipAudioCall: remove SipManager dependency." into gingerbread
by Hung-ying Tyan
· 14 years ago
00a2206
SipService: handle cross-domain authentication error
by Hung-ying Tyan
· 14 years ago
bd22942
Fix the unhold issue especially if one is behind NAT.
by Chung-yih Wang
· 14 years ago
3a4197e
SipAudioCall: remove SipManager dependency.
by Hung-ying Tyan
· 14 years ago
658bec9
SDP: remove dead code.
by Chia-chi Yeh
· 14 years ago
84a357b
Refactoring SIP classes to get ready for API review.
by Hung-ying Tyan
· 14 years ago
0b7d6de
Fix the build.
by repo sync
· 14 years ago
84f7f6b
SIP: Make SipAudioCallImpl use SimpleSessionDescription instead of javax.sdp.
by repo sync
· 14 years ago
e6c0c10
SDP: Add a simple class to help manipulate session descriptions.
by Chia-chi Yeh
· 14 years ago
7a69aef
RTP: Add log throttle for "no data".
by repo sync
· 14 years ago
4033a67
RTP: Update native part to reflect the API change.
by Chia-chi Yeh
· 14 years ago
37adc52
RTP: Add two getters to retrieve the current configuration from AudioStream.
by Chia-chi Yeh
· 14 years ago
32e106b
RTP: Extend codec capability and update the APIs.
by Chia-chi Yeh
· 14 years ago
8544560
SipPhone: fix missing-call DisconnectCause feedback
by Hung-ying Tyan
· 14 years ago
9796379
SIP: convert enum to static final int.
by Hung-ying Tyan
· 14 years ago
c4b8747
SIP: add config flag for wifi-only configuration.
by Hung-ying Tyan
· 14 years ago
afa583e
SipAudioCall: expose startAudio()
by Hung-ying Tyan
· 14 years ago
9352cf1
Add timer to SIP session creation process.
by Hung-ying Tyan
· 14 years ago
286bb5a
Fix links in SIP API javadoc.
by Hung-ying Tyan
· 14 years ago
ae076d3
SIP: add PEER_NOT_REACHABLE error feedback.
by Hung-ying Tyan
· 14 years ago
12bec5d
SipService: ignore connect event for non-active networks.
by Hung-ying Tyan
· 14 years ago
13f6270
SipAudioCall: use SipErrorCode instead of string in onError()
by Hung-ying Tyan
· 14 years ago
99bf4e4
SIP: remove dependency on javax.sip
by Hung-ying Tyan
· 14 years ago
d231aa8
SipService: deliver connectivity change to all sessions.
by Hung-ying Tyan
· 14 years ago
3d7606a
SIP: enhance timeout and registration status feedback.
by Hung-ying Tyan
· 14 years ago
25b52a2
SIP: remove dependency on javax.sip.SipException.
by Hung-ying Tyan
· 14 years ago
903e103
SIP: add SipErrorCode for error feedback.
by Hung-ying Tyan
· 14 years ago
f6936a3a
Merge "RTP: prevent buffer overflow in AudioRecord." into gingerbread
by Chia-chi Yeh
· 14 years ago
557b04d
RTP: prevent buffer overflow in AudioRecord.
by Chia-chi Yeh
· 14 years ago
643fce9
SipManager: always return true for SIP API and VOIP support query.
by Hung-ying Tyan
· 14 years ago
dc296b0
Merge "SipService: reduce the usage of javax.sdp.*." into gingerbread
by Chia-chi Yeh
· 14 years ago
95b15c3
SipService: reduce the usage of javax.sdp.*.
by Chia-chi Yeh
· 14 years ago
60264b30
SipProfile: remove outgoingCallAllowed flag.
by Hung-ying Tyan
· 14 years ago
3424c02
Add software features for SIP and VOIP
by Hung-ying Tyan
· 14 years ago
0858806
Add Wifi High Perf. mode during a call.
by Chung-yih Wang
· 14 years ago
14e0062
Merge "Revert "RTP: integrate the echo canceller from speex."" into gingerbread
by Chia-chi Yeh
· 14 years ago
7fa7ee1
Revert "RTP: integrate the echo canceller from speex."
by Chia-chi Yeh
· 14 years ago
5424c8d
Add dynamic uid info for tracking the sip service usage.
by Chung-yih Wang
· 14 years ago
37f709a
Merge "SipProfile: add isOutgoingCallAllowed() and new builder constructor" into gingerbread
by Hung-ying Tyan
· 14 years ago
cf95f5d
SipProfile: add isOutgoingCallAllowed() and new builder constructor
by Hung-ying Tyan
· 14 years ago
3294d44
Add confcall management to SIP calls
by Hung-ying Tyan
· 14 years ago
4ae6ec4
RTP: integrate the echo canceller from speex.
by Chia-chi Yeh
· 14 years ago
2880ef8
RTP: reduce the latency by overlapping AudioRecord and AudioTrack.
by Chia-chi Yeh
· 14 years ago
b879032
RTP: fix few leaks when fail to add streams into a group.
by Chia-chi Yeh
· 14 years ago
3459d30
RTP: remove froyo-compatible code.
by Chia-chi Yeh
· 14 years ago
cfd15dd
Fix the IN_CALL mode issue.
by Chung-yih Wang
· 14 years ago
ea4de5bd
SipAudioCall: perform local ops before network op in endCall()
by Hung-ying Tyan
· 14 years ago
8e63ddb
SIP: clean up unused class and fields.
by Hung-ying Tyan
· 14 years ago
4c5d28c
RTP: move into frameworks.
by Chia-chi Yeh
· 14 years ago
cde66df
Revert "Move SIP telephony related codes to framework."
by Chung-yih Wang
· 14 years ago
b631dcf
Move SIP telephony related codes to framework.
by Chung-yih Wang
· 14 years ago
363c2ab
Move the sip related codes to framework.
by Chung-yih Wang
· 14 years ago