solenberg
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11ace15c19
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The VoE functionality to apply receive-side processing to VoE channels is unused. I'm removing it so we can avoid instantiating a full APM per channel (and thus also for webrtc::AudioSendStream and webrtc::AudioReceiveStream), and then never use it.
The following APIs are removed from VoEAudioProcessing:
virtual int SetRxNsStatus(int channel,
bool enable,
NsModes mode = kNsUnchanged) = 0;
virtual int GetRxNsStatus(int channel, bool& enabled, NsModes& mode) = 0;
virtual int SetRxAgcStatus(int channel,
bool enable,
AgcModes mode = kAgcUnchanged) = 0;
virtual int GetRxAgcStatus(int channel, bool& enabled, AgcModes& mode) = 0;
virtual int SetRxAgcConfig(int channel, AgcConfig config) = 0;
virtual int GetRxAgcConfig(int channel, AgcConfig& config) = 0;
virtual int RegisterRxVadObserver(int channel,
VoERxVadCallback& observer) = 0;
virtual int DeRegisterRxVadObserver(int channel) = 0;
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/2295113002
Cr-Commit-Position: refs/heads/master@{#14227}
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2016-09-15 11:29:21 +00:00 |
|
Jelena Marusic
|
0d266054ac
|
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}
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2015-05-04 12:15:41 +00:00 |
|
bjornv@webrtc.org
|
cc64a9cc4f
|
voice_engine: Updates GetEcDelayMetrics() w.r.t. new metric
As of r8230 (https://webrtc-codereview.appspot.com/39739004/) a new Echo Delay Metric was added calculating the fraction of poor values that may cause the AEC to fail. There are currently two methods for GetDelayMetrics() in webrtc::AutioProcessing and one is deprecated.
This CL updates
- GetEcDelayMetrics()
- voe_auto_test
- talk/media/(fake)webrtcvoiceengine
BUG=N/A
TESTED=locally and trybots
R=pbos@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41749004
Cr-Commit-Position: refs/heads/master@{#8251}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8251 4adac7df-926f-26a2-2b94-8c16560cd09d
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2015-02-05 12:53:24 +00:00 |
|
henrikg@webrtc.org
|
863b536100
|
Allow opening an AEC dump from an existing file handle.
This is necessary for Chromium to be able enable the dump with the sanbox enabled. It will open the file in the browser process and pass the handle to the render process.
This changes FileWrapper to deal with the case were the file handle is not managed by the wrapper.
BUG=2567
R=andrew@webrtc.org, henrika@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5239 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2013-12-06 16:05:17 +00:00 |
|
pbos@webrtc.org
|
d900e8bea8
|
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
|
2013-07-03 15:12:26 +00:00 |
|
pbos@webrtc.org
|
956aa7e087
|
Include files from webrtc/.. paths in voice_engine/
BUG=1662
R=henrikg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1434005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4079 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2013-05-21 13:52:32 +00:00 |
|
pbos@webrtc.org
|
9213521ea9
|
Remove const for plain data types in voice_engine/
BUG=1644
R=henrikg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1463004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4018 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-05-14 08:31:39 +00:00 |
|
andrew@webrtc.org
|
14b43beb7c
|
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
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2012-10-22 18:19:23 +00:00 |
|