Log - b8182a382df3efaf18c9810dd7c517b2601edadb - platform/frameworks/base

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