Reason for revert:
Compile error on Android needs to be fixed before relanding.
Original issue's description:
> Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87.
>
> The revert of the original CL was commit 36d4c545007129446e551c45c17b25377dce89a4.
> Original review: https://codereview.webrtc.org/1413483003/
>
> The original CL changes a function on audio_processing.h that is used by Chrome, this CL adds back the old function.
>
> NOTRY=true
> TBR=glaznev@webrtc.org, henrik.lundin@webrtc.org, solenberg@google.com, henrikg@webrtc.org, perkj@webrtc.org
> BUG=webrtc:4741
>
> Committed: https://crrev.com/f4f5cb09277d5ef6aeac8341e5f54a055867803a
> Cr-Commit-Position: refs/heads/master@{#11093}
TBR=glaznev@webrtc.org,henrik.lundin@webrtc.org,solenberg@google.com,henrikg@webrtc.org,perkj@webrtc.org,kwiberg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1537213002
Cr-Commit-Position: refs/heads/master@{#11094}
Reason for revert:
Breaks Chrome-FYI bots because of a change in the StartDebugRecording function in audio_processing.h, that is called from Chrome.
Original issue's description:
> Added option to specify a maximum file size when recording an AEC dump.
>
> For applications with a strict filesize limit for debug files,
> I added an option to specify a maximum filesize for AEC dumps. An
> existing unit test is extended to check that the feature works as
> advertised.
>
> BUG=webrtc:4741
> TBR=glaznev@webrtc.org
>
> Committed: https://crrev.com/ae2c5ad12afc8cc29fe9c59dea432b697b871a87
> Cr-Commit-Position: refs/heads/master@{#11081}
TBR=pthatcher@webrtc.org,henrik.lundin@webrtc.org,henrikg@webrtc.org,solenberg@webrtc.org,andrew@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org,glaznev@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1533913004
Cr-Commit-Position: refs/heads/master@{#11087}
For applications with a strict filesize limit for debug files,
I added an option to specify a maximum filesize for AEC dumps. An
existing unit test is extended to check that the feature works as
advertised.
BUG=webrtc:4741
TBR=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1413483003
Cr-Commit-Position: refs/heads/master@{#11081}
We can now use std::move instead!
This CL leaves the Pass methods in place; a follow-up CL will add deprecation annotations to them.
Review URL: https://codereview.webrtc.org/1460043002
Cr-Commit-Position: refs/heads/master@{#11064}
This will let us transition to the new Initialize method in Chromium,
and then get rid of the old one.
Review URL: https://codereview.webrtc.org/1462253002
Cr-Commit-Position: refs/heads/master@{#10860}
The function to stop recording an AEC dump was missing from the PeerConnectionFactory interface (only a start function was provided). This CL adds the missing stop function.
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1415733005
Cr-Commit-Position: refs/heads/master@{#10372}
DtlsIdentityStoreImpl is updated to take KeyType into account, something which will be relevant after this CL lands:
https://codereview.webrtc.org/1189583002
The DtlsIdentityService[Interface] classes are about to be removed (to be removed when Chromium no longer implements and uses the interface). This was an unnecessary layer of complexity. The FakeIdentityService is now instead a FakeDtlsIdentityStore.
Where a service was previously passed around, a store is now passed around.
Identity generation is now commonly performed using DtlsIdentityStoreInterface. Previously, if a service was not specified, WebRtcSessionDescriptionFactory could fall back on its own generation code. Now, a store has to be provided for generation to occur.
For more information about the steps being taken to land this without breaking Chromium, see referenced bug.
BUG=webrtc:4899
R=magjed@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1176383004 .
Cr-Commit-Position: refs/heads/master@{#9696}
In order to do that, the signaling thread is also changed to wrap the current thread unless an external signaling thread thread is specified in the call to CreatePeerConnectionFactory.
This cleans up the PeerConnectionFactory and makes sure a user of the API will always access the factory on the signaling thread.
Note that both Chrome and the Android implementation use an external signaling thread.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8039 4adac7df-926f-26a2-2b94-8c16560cd09d
