Commit Graph

6 Commits

Author SHA1 Message Date
eb3603bd5e Don't always downsample to 16kHz in the reverse stream in APM
The first approach landed here: https://codereview.webrtc.org/1773173002
But it was partially reverted, because it affected the AEC performance, here: https://codereview.webrtc.org/1867483003/
The main difference of this approach is that it doesn't use the 3-band splitting filter in the reverse stream, which seems to be the culprit of the AEC regression.
Also, the 2-band splitting filter has been used for the 32kHz case for a long time without any problem, and this is expanded in the CL to cover the 48kHz case as well.

BUG=webrtc:5725
TBR=tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1865633005

Cr-Commit-Position: refs/heads/master@{#12451}
2016-04-20 22:28:01 +00:00
0bf612b3ec This CL is partially reverting the effects that
were added in https://codereview.webrtc.org/1773173002.

The reason for the revert is that for some scenarios
that CL causes problems in the coherence estimate used
in the AEC, which in turn causes echo leakage.

The reason for not reverting the actual CL is that
it would cause subsequent CLs to be reverted as well.
Therefore the choice was made to in this CK
instead revert the effects of that CL.

With the changes in this CL, the behavior is bitexact
to what it was before the CL mentioned above.

TBR=aluebs@webrtc.org

BUG=webrtc:5725

Review URL: https://codereview.webrtc.org/1867483003

Cr-Commit-Position: refs/heads/master@{#12259}
2016-04-06 09:47:52 +00:00
df6416aa50 Dont always downsample to 16kHz in the reverse stream in APM
TBR=tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1773173002

Cr-Commit-Position: refs/heads/master@{#12024}
2016-03-17 01:26:42 +00:00
477487410a Enable AudioProcessing48kHzSupport by default
Because of the Finch experiment, this will not affect Chrome's behaviour at all.
The SNRs in AudioProcessingTest.Formats were only increased to the next multiple of 5.

BUG=webrtc:3146
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43359004

Cr-Commit-Position: refs/heads/master@{#9263}
2015-05-22 18:59:59 +00:00
b1786dbab0 audio_processing: Added a new AEC delay metric value that gives the amount of poor delays
To more easily determine if for example the AEC is not working properly one could monitor how often the estimated delay is out of bounds. With out of bounds we mean either being negative or too large, where both cases will break the AEC.

A new delay metric is added telling the user how often poor delay values were estimated. This is measured in percentage since last time the metrics were calculated.

All APIs have been updated with a third parameter with EchoCancellation::GetDelayMetrics() giving the option to exclude the new metric not to break existing code.

The new metric has been added to audio_processing_unittests with an additional protobuf member, and reference files accordingly updated.
voe_auto_test has not been updated to display the new metric.

BUG=4246
TESTED=audioproc on files
R=aluebs@webrtc.org, andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39739004

Cr-Commit-Position: refs/heads/master@{#8230}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8230 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-03 06:07:21 +00:00
f17ee9c709 Add case to ApmTest.Process to test the extended filter mode
R=andrew@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40509004

Cr-Commit-Position: refs/heads/master@{#8192}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8192 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-29 00:04:18 +00:00