74f0f3551ecd596dc0f83146d218887082528fa8 |
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01-Nov-2015 |
henrik.lundin <henrik.lundin@webrtc.org> |
Delete a chain of methods in ViE, VoE and ACM The end goal is to remove AcmReceiver::SetInitialDelay. This change is in preparation for that goal. It turns out that AcmReceiver::SetInitialDelay was only invoked through the following call chain, where each method in the chain is never referenced from anywhere else (except from tests in some cases): ViEChannel::SetReceiverBufferingMode -> ViESyncModule::SetTargetBufferingDelay -> VoEVideoSync::SetInitialPlayoutDelay -> Channel::SetInitialPlayoutDelay -> AudioCodingModule::SetInitialPlayoutDelay -> AcmReceiver::SetInitialDelay The start of the chain, ViEChannel::SetReceiverBufferingMode was never referenced. This change deletes all the methods above except AcmReceiver::SetInitialDelay itself, which will be handled in a follow-up change. BUG=webrtc:3520 Review URL: https://codereview.webrtc.org/1421013006 Cr-Commit-Position: refs/heads/master@{#10471}
/external/webrtc/webrtc/voice_engine/voe_video_sync_impl.h
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0d266054acece70259fc1e85026194154f41e5a0 |
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04-May-2015 |
Jelena Marusic <jmarusic@webrtc.org> |
VoE: apply new style guide on VoE interfaces and their implementations Changes: 1. Ran clang-format on VoE interfaces and their implementations. 2. Replaced virtual with override in derived classes. R=henrika@webrtc.org Review URL: https://webrtc-codereview.appspot.com/49239004 Cr-Commit-Position: refs/heads/master@{#9130}
/external/webrtc/webrtc/voice_engine/voe_video_sync_impl.h
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822fbd8b68ffdb481b9557e2950ae8d6657c8ce6 |
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16-Aug-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 50918584. Together with Stefan's http://review.webrtc.org/1960004/. R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2048004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4556 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_video_sync_impl.h
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aa4d96a134a03f998d52fb9699845d9c644eb24b |
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16-Jul-2013 |
tnakamura@webrtc.org <tnakamura@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert r4301 R=mikhal@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1809004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4357 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_video_sync_impl.h
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66b2e5c05a3f2a93d634d1dbbcbb283fb218ca4f |
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05-Jul-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the rtp_rtcp implementation. This refactoring significantly reduces the receive-side RTP parser and receiver complexity, and makes it possible to implement RTX correctly by having two instances of receive-statistics. With this change the dead-or-alive and packet timeout APIs are removed. TEST=trybots, vie_auto_test, voe_auto_test BUG=1811 R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1745004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4301 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_video_sync_impl.h
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d900e8bea84c474696bf0219aed1353ce65ffd8e |
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03-Jul-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Proper spacing for end-of-namespace comments. BUG= R=mflodman@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1760006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4293 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_video_sync_impl.h
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e46c8d387587ba148e229a7bb18f1cc0708a2a87 |
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22-May-2013 |
turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
API to control target delay in NetEq jitter buffer. NetEq maintains the given delay unless channel conditions require a higher delay. TEST=unit-test, manual, trybots. R=henrik.lundin@webrtc.org, henrika@webrtc.org, mflodman@webrtc.org, mikhal@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1384005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4087 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_video_sync_impl.h
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1de01354e68da71bc62c81af17afeac8ed374a18 |
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11-Apr-2013 |
pwestin@webrtc.org <pwestin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/voice_engine/voe_video_sync_impl.h
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6388c3e2fdfc91b3648fb7d408a14ddb25e41cd1 |
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12-Feb-2013 |
turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Implement initial delay. This CL allows clients of VoE to set an initial delay. Playout of audio is delayed and the extra playout delay is maintained during the call. While packets are buffered (in NetEq) to acheive the desired delay. ACM will playout silence (zeros). Initial delay has to be set before any packet is pushed into ACM. TEST=ACM unit test is added, also a manual integration test is writen. Review URL: https://webrtc-codereview.appspot.com/1097009 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3506 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_video_sync_impl.h
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14b43beb7ce4440b30dcea31196de5b4a529cb6b |
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22-Oct-2012 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move src/ -> webrtc/ TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/915006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_video_sync_impl.h
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