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gerrit-public.fairphone.software / platform / frameworks / base / 3bdc7209b10cfae095cb5db8681a14e792a1d165 / voip
  1. 3559d1d am ea16e72b: am b355714a: Merge "SipManager: always return true for SIP API and VOIP support query." into gingerbread by Hung-ying Tyan · 14 years ago
  2. 643fce9 SipManager: always return true for SIP API and VOIP support query. by Hung-ying Tyan · 14 years ago
  3. 3d67c56 resolved conflicts for merge of 12eaf9d5 to master by repo sync · 14 years ago
  4. dc296b0 Merge "SipService: reduce the usage of javax.sdp.*." into gingerbread by Chia-chi Yeh · 14 years ago
  5. 95b15c3 SipService: reduce the usage of javax.sdp.*. by Chia-chi Yeh · 14 years ago
  6. e93a492 am 18dfae22: am 60264b30: SipProfile: remove outgoingCallAllowed flag. by Hung-ying Tyan · 14 years ago
  7. 60264b30 SipProfile: remove outgoingCallAllowed flag. by Hung-ying Tyan · 14 years ago
  8. f83d4f1 resolved conflicts for merge of 3e4975a5 to master by Hung-ying Tyan · 14 years ago
  9. 3424c02 Add software features for SIP and VOIP by Hung-ying Tyan · 14 years ago
  10. b6d4723 Revert "Add Wifi High Perf. mode during a call." by Chung-yih Wang · 14 years ago
  11. 0858806 Add Wifi High Perf. mode during a call. by Chung-yih Wang · 14 years ago
  12. 14e0062 Merge "Revert "RTP: integrate the echo canceller from speex."" into gingerbread by Chia-chi Yeh · 14 years ago
  13. 7fa7ee1 Revert "RTP: integrate the echo canceller from speex." by Chia-chi Yeh · 14 years ago
  14. 5424c8d Add dynamic uid info for tracking the sip service usage. by Chung-yih Wang · 14 years ago
  15. 37f709a Merge "SipProfile: add isOutgoingCallAllowed() and new builder constructor" into gingerbread by Hung-ying Tyan · 14 years ago
  16. cf95f5d SipProfile: add isOutgoingCallAllowed() and new builder constructor by Hung-ying Tyan · 14 years ago
  17. 3294d44 Add confcall management to SIP calls by Hung-ying Tyan · 14 years ago
  18. 4ae6ec4 RTP: integrate the echo canceller from speex. by Chia-chi Yeh · 14 years ago
  19. 2880ef8 RTP: reduce the latency by overlapping AudioRecord and AudioTrack. by Chia-chi Yeh · 14 years ago
  20. b879032 RTP: fix few leaks when fail to add streams into a group. by Chia-chi Yeh · 14 years ago
  21. 3459d30 RTP: remove froyo-compatible code. by Chia-chi Yeh · 14 years ago
  22. cfd15dd Fix the IN_CALL mode issue. by Chung-yih Wang · 14 years ago
  23. ea4de5bd SipAudioCall: perform local ops before network op in endCall() by Hung-ying Tyan · 14 years ago
  24. 8e63ddb SIP: clean up unused class and fields. by Hung-ying Tyan · 14 years ago
  25. 4c5d28c RTP: move into frameworks. by Chia-chi Yeh · 14 years ago
  26. cde66df Revert "Move SIP telephony related codes to framework." by Chung-yih Wang · 14 years ago
  27. b631dcf Move SIP telephony related codes to framework. by Chung-yih Wang · 14 years ago
  28. 363c2ab Move the sip related codes to framework. by Chung-yih Wang · 14 years ago