pwestin@webrtc.org 1de01354e6 Adding playout buffer status to the voe video sync ...

Review URL: https://webrtc-codereview.appspot.com/1311004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3835 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-04-11 20:23:35 +00:00

..

mock

WebRtc_Word32 -> int32_t in voice_engine/

2013-04-09 10:09:10 +00:00

voe_audio_processing.h

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

voe_base.h

Remove UDP transport API from VoE

2013-04-03 15:43:57 +00:00

voe_call_report.h

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

voe_codec.h

VoE Changes to enable dual_streaming.

2012-12-11 02:15:12 +00:00

voe_dtmf.h

Remove DTMF detection. Talk team has been in the loop and there is no need for

2013-03-12 22:27:27 +00:00

voe_encryption.h

Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRTP APIs. Test stubs are left in place as we still have the (De)RegisterExternalEncryption() APIs, although they are currently untested.

2013-03-28 09:14:36 +00:00

voe_errors.h

VoE Changes to enable dual_streaming.

2012-12-11 02:15:12 +00:00

voe_external_media.h

WebRtc_Word32 -> int32_t in voice_engine/

2013-04-09 10:09:10 +00:00

voe_file.h

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

voe_hardware.h

Removed CPU APIs from VoEHardware. Code is now only used by test applications.

2013-03-28 15:58:49 +00:00

voe_neteq_stats.h

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

voe_network.h

Remove UDP transport API from VoE

2013-04-03 15:43:57 +00:00

voe_rtp_rtcp.h

Add API to retreive last received RTP timestamp to VoiceEngine.

2012-12-12 21:31:41 +00:00

voe_video_sync.h

Adding playout buffer status to the voe video sync

2013-04-11 20:23:35 +00:00

voe_volume_control.h

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00