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gerrit-public.fairphone.software
/
platform
/
frameworks
/
base
/
64a55af0ac700baecb0877235eb42caac59a3560
/
voip
845b471
Use audio_channel_mask_t more consistently
by Glenn Kasten
· 12 years ago
9be0105
docs: fix several links
by Scott Main
· 12 years ago
870085c
Merge "SIP: push the logic of finding local address down to SipSessionGroup."
by Chia-chi Yeh
· 13 years ago
2dd9134
SIP: push the logic of finding local address down to SipSessionGroup.
by Chia-chi Yeh
· 13 years ago
e669505
RTP: refactor a little bit and fix few minor bugs.
by Chia-chi Yeh
· 13 years ago
ba4d043
frameworks base Android.mk file changes
by James Dong
· 13 years ago
fb982db
RTP: add a null-check in AudioStream.setDtmfType().
by Chia-chi Yeh
· 13 years ago
32d72b2
Merge "Whitespace"
by Glenn Kasten
· 13 years ago
f743e1f
Whitespace
by Glenn Kasten
· 13 years ago
ae75f99
Add libmedia_native
by Glenn Kasten
· 13 years ago
4e42c5f
Remove dependency on audio_* location
by Glenn Kasten
· 13 years ago
597f828
Fix build warnings
by Glenn Kasten
· 13 years ago
3762c31
Rename (IF_)LOGE(_IF) to (IF_)ALOGE(_IF) DO NOT MERGE
by Steve Block
· 13 years ago
8564c8d
Rename (IF_)LOGW(_IF) to (IF_)ALOGW(_IF) DO NOT MERGE
by Steve Block
· 13 years ago
6215d3f
Rename (IF_)LOGI(_IF) to (IF_)ALOGI(_IF) DO NOT MERGE
by Steve Block
· 13 years ago
5baa3a6
Rename (IF_)LOGD(_IF) to (IF_)ALOGD(_IF) DO NOT MERGE
by Steve Block
· 13 years ago
c2bd616
SipService: grab Wi-Fi lock only when necessary.
by Chia-chi Yeh
· 13 years ago
2b07267
am 201469f5: am bb7f590a: Merge "docs: Add developer guide cross-references, Project ACRE, round 4" into ics-mr1
by Joe Fernandez
· 13 years ago
3aef8e1d
docs: Add developer guide cross-references, Project ACRE, round 4
by Joe Fernandez
· 13 years ago
98c54a3
am 0e67685e: am bfd85f9a: Merge "SIP: turn off verbose logs." into ics-mr1
by Chia-chi Yeh
· 13 years ago
cb6ee06
SIP: turn off verbose logs.
by Chia-chi Yeh
· 13 years ago
71f2cf1
Rename (IF_)LOGV(_IF) to (IF_)ALOGV(_IF) DO NOT MERGE
by Steve Block
· 13 years ago
ee59e6a
SipService: handle connectivity changes correctly.
by Chia-chi Yeh
· 13 years ago
6d8b9b8
Merge "RTP: Update parameters for larger packet intervals."
by Chia-chi Yeh
· 13 years ago
7a685e8
Merge "SIP: fix keep-alive measurement and increase the timeout."
by Chia-chi Yeh
· 13 years ago
d17b6d5
SIP: fix keep-alive measurement and increase the timeout.
by Chia-chi Yeh
· 13 years ago
be57bfe
RTP: Update parameters for larger packet intervals.
by Chia-chi Yeh
· 13 years ago
81a5ec5
Merge "RTP: support payloads with larger packetization interval."
by Chia-chi Yeh
· 13 years ago
fa6067f
Merge "VoIP JNI: Force AEC on for tuna board"
by Eric Laurent
· 13 years ago
35d05dc
RTP: support payloads with larger packetization interval.
by Chia-chi Yeh
· 13 years ago
54eabd6
SIP: avoid extreme small values in Min-Expires headers.
by Chia-chi Yeh
· 13 years ago
74e0a99
VoIP JNI: Force AEC on for tuna board
by Eric Laurent
· 13 years ago
5f76006
SIP: add the check for expiry time in Contact header.
by Chia-chi Yeh
· 13 years ago
dc5bbe9
Handle SIP authentication response for BYE.
by Hung-ying Tyan
· 13 years ago
53ad2c7
am 0793586b: am f8c1f129: am e1d27154: am f87743e7: Merge "Prevent NullPointerException cases while using SipService."
by Conley Owens
· 13 years ago
0793586
am f8c1f129: am e1d27154: am f87743e7: Merge "Prevent NullPointerException cases while using SipService."
by Conley Owens
· 13 years ago
25ccbb9
Prevent NullPointerException cases while using SipService.
by Masahiko Endo
· 13 years ago
5fb3ba6
Issue 3370834: No Echo canceler for SIP
by Eric Laurent
· 13 years ago
307f15f
Add REFER handling.
by repo sync
· 13 years ago
3eeb1a9
Merge "Keep last known keepalive interval to avoid duplicate effort."
by Hung-ying Tyan
· 13 years ago
9324e04
Merge "Do not hold wifi lock when SIP is also available over mobile network."
by Hung-ying Tyan
· 13 years ago
f8c34ad
Merge "Do not keep alive for re-established call."
by Hung-ying Tyan
· 13 years ago
9edfa10
Keep last known keepalive interval to avoid duplicate effort.
by Hung-ying Tyan
· 13 years ago
8ba4566
Do not keep alive for re-established call.
by Hung-ying Tyan
· 13 years ago
f89654d
Do not hold wifi lock when SIP is also available over mobile network.
by Hung-ying Tyan
· 13 years ago
a6cec8f
Synchronize SipWakeupTimer.onReceive()
by Hung-ying Tyan
· 13 years ago
129d0b0
Make NAT port timeout measurement more flexible.
by Hung-ying Tyan
· 13 years ago
99705b5
Record external IP and port from SIP responses
by Hung-ying Tyan
· 13 years ago
2093561
Support INVITE w/o SDP.
by repo sync
· 13 years ago
233718c
Start keepalive process for the caller of a SIP call
by Hung-ying Tyan
· 13 years ago
1aceda3
Support Invite w/ Replaces request.
by repo sync
· 13 years ago
e65f3a8
Restart NAT port timeout measurement when keepalive fails and other fixes
by Hung-ying Tyan
· 13 years ago
4af085f
Execute all the due wakeup events in SipWakeupTimer.
by Hung-ying Tyan
· 13 years ago
1275070
Keep the keepalive process going after NAT port is changed.
by Hung-ying Tyan
· 13 years ago
4a267a9
Move the keepalive process to SipSessionImpl and make it reusable.
by Hung-ying Tyan
· 14 years ago
ac320b2
Merge "Move WakeupTimer out of SipService."
by Hung-ying Tyan
· 14 years ago
5621554
Move WakeupTimer out of SipService.
by Hung-ying Tyan
· 14 years ago
c133781
Fix the issue of onNetwork in UI thread.
by repo sync
· 14 years ago
bb0a989
Add KeepAlive Interval Measurement.
by Chung-yih Wang
· 14 years ago
34bb419
update for new audio.h header location
by Dima Zavin
· 14 years ago
4b3913a
Squashed commit of the following:
by Andreas Huber
· 14 years ago
b8df57d
am d81214da: am a7a9c4cb: am 46524f83: Merge "docs: add package description for RTP" into honeycomb-mr1
by Scott Main
· 14 years ago
d81214d
am a7a9c4cb: am 46524f83: Merge "docs: add package description for RTP" into honeycomb-mr1
by Scott Main
· 14 years ago
de9acb7
docs: add package description for RTP
by Scott Main
· 14 years ago
24fc2fb
audio/media: convert to using the audio HAL and new audio defs
by Dima Zavin
· 14 years ago
d8cbd16
am 7a492a9a: am b7a76e84: am a482d83c: Merge "Issue 4157048: mic gain for VoIP/SIP calls." into gingerbread
by Eric Laurent
· 14 years ago
7a492a9
am b7a76e84: am a482d83c: Merge "Issue 4157048: mic gain for VoIP/SIP calls." into gingerbread
by Eric Laurent
· 14 years ago
b7a76e8
am a482d83c: Merge "Issue 4157048: mic gain for VoIP/SIP calls." into gingerbread
by Eric Laurent
· 14 years ago
d7a724e
Issue 4157048: mic gain for VoIP/SIP calls.
by Eric Laurent
· 14 years ago
397de16
am fae5e289: am 6f67e7bf: am 2e383bc6: Merge "Making it possible to call SIP calls with special allowed chars."
by Brad Fitzpatrick
· 14 years ago
fae5e28
am 6f67e7bf: am 2e383bc6: Merge "Making it possible to call SIP calls with special allowed chars."
by Brad Fitzpatrick
· 14 years ago
b5c72ea
Making it possible to call SIP calls with special allowed chars.
by Magnus Strandberg
· 14 years ago
3f9e089
Include strings.h instead of string.h for the strcasecmp prototype.
by Carl Shapiro
· 14 years ago
3070af0
frameworks/base: remove LOCAL_PRELINK_MODULE
by Iliyan Malchev
· 14 years ago
6defd2d
NEW_API: Unhide RTP APIs.
by Chia-chi Yeh
· 14 years ago
c52f5b2
RTP: update javadocs.
by Chia-chi Yeh
· 14 years ago
89bc1fe
Activate the wifi high perf. for sip calls.
by Chung-yih Wang
· 14 years ago
fcd0e50
Add rport argument for a reinvite request.
by Chung-yih Wang
· 14 years ago
9e25df4
Make SIP AuthName APIs public.
by Chung-yih Wang
· 14 years ago
2ba92c7
do not merge bug 3370834 Cherrypick from master
by Jean-Michel Trivi
· 14 years ago
14b6d06
Merge changes Ib70e0cf2,I0691cd70 into gingerbread
by Hung-ying Tyan
· 14 years ago
f46013b
Merge "Merge "SipService: registers broadcast receivers on demand."" into honeycomb
by Hung-ying Tyan
· 14 years ago
e9b5407
Merge "SipService: registers broadcast receivers on demand."
by Hung-ying Tyan
· 14 years ago
40f2cac
Merge "SipService: release wake lock for cancelled tasks."
by Hung-ying Tyan
· 14 years ago
0f7de88
Merge "Add auth. username in SipProfile." from gingerbread.
by Chung-yih Wang
· 14 years ago
f268a2f
Add auth. username in SipProfile.
by Chung-yih Wang
· 14 years ago
f0bb1ce
SipService: registers broadcast receivers on demand.
by Hung-ying Tyan
· 14 years ago
d87be27
Enable built-in echo canceler if available.
by Chia-chi Yeh
· 14 years ago
4bf82df
Do not set back to AudioManager.MODE_NORMAL in SipAudioCall.
by Chia-chi Yeh
· 14 years ago
0c01e6e
SipService: release wake lock for cancelled tasks.
by Hung-ying Tyan
· 14 years ago
d0da380
am dc78e3fe: am 3cf71376: RTP: Send silence packets on idle streams for every second.
by Chia-chi Yeh
· 14 years ago
3cf7137
RTP: Send silence packets on idle streams for every second.
by Chia-chi Yeh
· 14 years ago
33808c6
am aec9a33f: am e0bd2688: Merge "Check if VoIP API is supported in SipManager." into gingerbread
by Hung-ying Tyan
· 14 years ago
5bd3782
Check if VoIP API is supported in SipManager.
by Hung-ying Tyan
· 14 years ago
635b2b7
am d90bc225: am a936b256: Remove SIP realm/domain check
by Hung-ying Tyan
· 14 years ago
a936b25
Remove SIP realm/domain check
by Hung-ying Tyan
· 14 years ago
58ee2ac
Check port in create peer's SIP profile.
by Hung-ying Tyan
· 14 years ago
eecf4a6
Check port in create peer's SIP profile.
by Hung-ying Tyan
· 14 years ago
c030a16
am c9cc9ab5: am 5f86d7f5: Merge "Fix SIP bug of different transport/port used for requests." into gingerbread
by Chung-yih Wang
· 14 years ago
f053292
Fix SIP bug of different transport/port used for requests.
by Chung-yih Wang
· 14 years ago
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