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| author | grunell <grunell@chromium.org> | 2015-08-07 04:41:59 -0700 |
|---|---|---|
| committer | Commit bot <commit-bot@chromium.org> | 2015-08-07 11:42:37 +0000 |
| commit | 1e8d8de4a6cabcd7b908bbb794b1ce50d7cbd7a6 (patch) | |
| tree | eaa7f11eaccd9c9716da93e63f191b61385329a7 /media/audio | |
| parent | d00b04448dc7a412f8264e238fa2bee49fe3c723 (diff) | |
| download | chromium_src-1e8d8de4a6cabcd7b908bbb794b1ce50d7cbd7a6.zip chromium_src-1e8d8de4a6cabcd7b908bbb794b1ce50d7cbd7a6.tar.gz chromium_src-1e8d8de4a6cabcd7b908bbb794b1ce50d7cbd7a6.tar.bz2 | |
Revert of Modifies size of ring buffer in shared memory on the audio capture side (patchset #3 id:40001 of https://codereview.chromium.org/1254793002/ )
Reason for revert:
This was an speculative change and should now be reverted. There is no sign that this change would fix or improve the issue.
Original issue's description:
> Modifies size of ring buffer in shared memory on the audio capture side.
>
> We have seen reports in Chrome where segments of repeated input audio has
> damaged AEC performance in WebRTC clients. This patch is an attempt to limit
> the number of possible places in Chrome where such a patter could be created.
>
> It is a speculative patch and the idea is that it should live in Canary
> for a while so we can see if there are any positive effects.
>
> TBR=tommi
> BUG=b/13976602
> TEST=manual tests of different WebRTC clients running in loopback to ensure that no
> audio artifacts can be detected
>
> Committed: https://crrev.com/bda4fc7018c3ce93d88b74461a065f9eaf0b8b5f
> Cr-Commit-Position: refs/heads/master@{#340261}
TBR=tommi@chromium.org,dalecurtis@chromium.org,henrika@chromium.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=b/13976602
Review URL: https://codereview.chromium.org/1277173002
Cr-Commit-Position: refs/heads/master@{#342336}
Diffstat (limited to 'media/audio')
| -rw-r--r-- | media/audio/audio_input_device.cc | 8 |
1 files changed, 1 insertions, 7 deletions
diff --git a/media/audio/audio_input_device.cc b/media/audio/audio_input_device.cc index 07eb11c..9f46b55 100644 --- a/media/audio/audio_input_device.cc +++ b/media/audio/audio_input_device.cc @@ -17,13 +17,7 @@ namespace media { // in order to avoid data overwriting. This number can be any positive number, // dependent how fast the renderer process can pick up captured data from // shared memory. -// TODO(henrika): figure out a suitable size of this ring buffer. -// We have seen reports in Chrome where segments of repeated input audio has -// damaged AEC performance in WebRTC clients. By setting its value to 1, we -// reduce the number of places in Chrome where such a patteren could possibly -// be created. The original value of kRequestedSharedMemoryCount was 10. -// See b/13976602 for details. -static const int kRequestedSharedMemoryCount = 1; +static const int kRequestedSharedMemoryCount = 10; // Takes care of invoking the capture callback on the audio thread. // An instance of this class is created for each capture stream in |