This CL adds support for handling highly reverberant echoes in
AEC3. The functionality is hardcoded to be have no effect (via
a decay factor of 0), but this CL will be followed by other CLs
for which nonzero decay factors will be allowed.
Apart from this change, this CL also refactors the residual
echo estimation code to make it shorter and more readable.
The changes introduced herein are bitexact (for a decay factor
of 0).
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2804223002
Cr-Commit-Position: refs/heads/master@{#17589}
This CL ensures that the number of bands
for the render side matches that for the capture side
when AEC3 is active. Without this, there was problems
when the render rate is different from the capture rate.
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2800033003
Cr-Commit-Position: refs/heads/master@{#17586}
Reason for revert:
Trying to re-land after solving some related issues.
There are no changes compared to the original CL.
Original issue's description:
> Revert of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #16 id:300001 of https://codereview.webrtc.org/2747863003/ )
>
> Reason for revert:
> I will try to reland next week because it is causing some problems.
>
> Original issue's description:
> > To accommodate some downstream WebRTC users we need to loosen
> > the coupling between our code and the //third_party/protobuf.
> >
> > This includes using typedefs to define strings instead of
> > assuming std::string.
> >
> > After this refactoring it will be possible to link with other
> > protobuf implementations than the current one.
> >
> > We moved the PRESUBMIT check to another CL [1]. The goal of this
> > presubmit is to avoid the direct usage of google::protobuf outside
> > of the webrtc/base/protobuf_utils.h header file.
> >
> > [1] - https://codereview.webrtc.org/2753823003/
> >
> > BUG=webrtc:7340
> > NOTRY=True
> >
> > Review-Url: https://codereview.webrtc.org/2747863003
> > Cr-Commit-Position: refs/heads/master@{#17466}
> > Committed: 16ab93b952
>
> TBR=kjellander@webrtc.org,henrik.lundin@webrtc.org,kwiberg@webrtc.org,michaelt@webrtc.org,peah@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:7340
>
> Review-Url: https://codereview.webrtc.org/2786363002
> Cr-Commit-Position: refs/heads/master@{#17483}
> Committed: d00aad5eb2TBR=kjellander@webrtc.org,henrik.lundin@webrtc.org,kwiberg@webrtc.org,michaelt@webrtc.org,peah@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7340
NOTRY=True
Review-Url: https://codereview.webrtc.org/2791963003
Cr-Commit-Position: refs/heads/master@{#17584}
Previously it was assumed that exactly one device is connected.
Now adb will get an argument with the device ID taken from the runner
script's stdout.
BUG=webrtc:7229
TBR=kjellander@webrtc.org
NOTRY=true
Review-Url: https://codereview.webrtc.org/2783343003
Cr-Commit-Position: refs/heads/master@{#17580}
This CL corrects the behavior during buffer underruns.
Furthermore, it increases the tolerance for API call jitter, and
removes the minimum value for the comfort noise.
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2787123003
Cr-Commit-Position: refs/heads/master@{#17576}
This CL adds fairly significant changes to the echo removal
functionality, the main ones being.
-More centralized control over the echo removal.
-Updated echo suppression gain behavior.
-Significantly increased usage of the linear adaptive filter.
-New echo removal functionality when the linear filter is not usable.
This CL is chained to the CL https://codereview.webrtc.org/2784023002/
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2782423003
Cr-Commit-Position: refs/heads/master@{#17575}
Adds a SetEncoder call to voe::Channel, so that we can move encoder setup outside of Voice Engine.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2703373006
Cr-Commit-Position: refs/heads/master@{#17572}
Added new callback class that enables routing both the locally recorded
and playout audio data to external module.
A factory method that creates an AudioDeviceModule instance that also
registers the AudioTransportCapture with the device is also added.
BUG=webrtc:7337
R=solenberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2753453002 .
Cr-Commit-Position: refs/heads/master@{#17570}
This CL contains all the changes made to audio_coding while making
audio encoders injectable. Apart from some small changes to
webrtcvoiceengine, nothing here is hooked up to the outside
world. Those changes will be added to a follow-up CL.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2695243005
Cr-Commit-Position: refs/heads/master@{#17569}
Because on gcc, cast to void doesn't silence the warning. See
https://gcc.gnu.org/bugzilla/show_bug.cgi?id=66425
Also add an RTC_ prefix to the macro instead of only defining it if it
wasn't already defined, to ensure that we always get our own version.
BUG=none
Review-Url: https://codereview.webrtc.org/2797983003
Cr-Commit-Position: refs/heads/master@{#17563}
They provide parsing of numbers from char* and std::string into any of
the fundamental integer types. Values are returned
in an rtc::Optional, which is left empty if parsing was unsuccessful.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2696003004
Cr-Commit-Position: refs/heads/master@{#17557}
The root cause is because the offset used in DxgiOutputDuplicator should always
depend on the position of the monitor in the system instead of the offset in the
target frame. Otherwise, once switching between two monitors with different
screen size, the updated region in the context would base on the old monitor,
and cause the copied regions to be out of the source DesktopFrame.
This issue also impacts the SpreadContextChange() function, the updated region
stores in the Context should also only depend on the position of the monitor.
BUG=chromium:706797
Review-Url: https://codereview.webrtc.org/2801433002
Cr-Commit-Position: refs/heads/master@{#17548}
This CL adds major render pipeline changes to the AEC3 code. The reason
for these are that
1) It allows the echo removal unit to receive information about the content
in bands beyond band 0, thereby allowing removal of high-frequency
echoes
2) It allows more controlled handling of the render buffers, allowing proper
buffer behaviour during capture glitches and clock-drift.
Unfortunately, the render pipeline caused a lot of related changes in much
of the rest of the AEC3 files. Most of these are, however, caused by
a change of class name.
Another unfortunate effect of this CL, is that a number of unittest cease to
compile. I chose to temporarily solve that by removing them from the
build using #if/#endif. The reason for that is that those will anyway again
need to be changed in the next review, and doing like this avoids them
having to be reviewed twice.
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2784023002
Cr-Commit-Position: refs/heads/master@{#17547}
Because the new API doesn't use constraints, refactors settings to instead have a
video resolution setting.
BUG=webrtc:7177
Review-Url: https://codereview.webrtc.org/2778163005
Cr-Commit-Position: refs/heads/master@{#17545}
I can't believe it's been this long before I found out we didn't have
them...
BUG=none
Review-Url: https://codereview.webrtc.org/2804673002
Cr-Commit-Position: refs/heads/master@{#17544}
- snake_case -> CapWords
- compulsory docstring added
- style
A followup CL will fix remaining issues as raised by the next version of the WebRTC Python linter (update in progress).
BUG=webrtc:7218
NOTRY=True
Review-Url: https://codereview.webrtc.org/2793903006
Cr-Commit-Position: refs/heads/master@{#17543}
This class has been deprecated for a long time and has been replaced by
Camera1Capturer.
BUG=webrtc:7440
Review-Url: https://codereview.webrtc.org/2789183004
Cr-Commit-Position: refs/heads/master@{#17538}
This provides a better default for audio-only tests.
BUG=None
Review-Url: https://codereview.webrtc.org/2794193003
Cr-Commit-Position: refs/heads/master@{#17536}
This will eventually implement webrtc::RtpTransportInterface from api/ortc.
It needs to live in the pc build target until the pc <- ortc dependency is inverted.
BUG=webrtc:7013
Review-Url: https://codereview.webrtc.org/2792223002
Cr-Commit-Position: refs/heads/master@{#17534}
Reason for revert:
Breaks Chrome FYI Android bots.
See:
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28L%20Nexus9%29/builds/20418https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28L%20Nexus6%29/builds/14724https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28L%20Nexus5%29/builds/20133https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28K%20Nexus5%29/builds/15111
Original issue's description:
> Deliver video frames on Android, on the decode thread.
>
> VideoCoding
> * Adding a method for polling for frames on Android only until the capture implementation takes care of this (longer term plan).
>
> CodecDatabase
> * Add an accessor for the current decoder
> * Use std::unique_ptr<> for ownership.
> * Remove "Release()" and "ReleaseDecoder()". Instead just delete.
> * Remove |friend| relationship between CodecDatabase and VCMGenericDecoder.
>
> VCMDecodedFrameCallback
> * DCHECKs for thread correctness.
> * Remove |lock_| now that a threading model has been established and verified.
>
> VCMGenericDecoder
> * All methods now have thread checks.
> * Variable access associated with thread checkers.
>
> VideoReceiver
> * Added two notification methods, DecoderThreadStarting() and DecoderThreadStopped()
> * Allows us to establish a period when the decoder thread is not running and it is safe to modify variables such as callbacks, that are only read when the decoder thread is running.
> * Allows us to DCHECK thread guarantees.
> * Allows synchronizing callbacks from the module process thread and have them only active while the decoder thread is running.
> * The above, allows us to establish two modes for the thread, single-threaded-mutable and multi-threaded-const.
> * Using that knowledge, we can remove |receive_crit_| as well as locking for a number of member variables.
>
> MediaCodecVideoDecoder
> * Removed frame polling code from this class, since this is now done from the root thread function in VideoReceiveStream.
>
> VideoReceiveStream
> * On Android: Polls for decoded frames every 10ms (same interval as previously in MediaCodecVideoDecoder)
> * [Un]Registers the |video_receiver_| with the module thread only around the time the decoder thread is started/stopped.
> * Notifies the receiver of start/stop events of the decoder thread.
> * Changed the decoder thread to use the new PlatformThread callback type.
>
> BUG=webrtc:7361, 695438
>
> Review-Url: https://codereview.webrtc.org/2764573002
> Cr-Commit-Position: refs/heads/master@{#17527}
> Committed: e3aa88bbd5TBR=sakal@webrtc.org,mflodman@webrtc.org,stefan@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7361, 695438
Review-Url: https://codereview.webrtc.org/2792033003
Cr-Commit-Position: refs/heads/master@{#17530}
The stats will be used in an upcoming cl to track the rate of adapt changes.
BUG=webrtc:6634
Review-Url: https://codereview.webrtc.org/2775173004
Cr-Commit-Position: refs/heads/master@{#17528}
VideoCoding
* Adding a method for polling for frames on Android only until the capture implementation takes care of this (longer term plan).
CodecDatabase
* Add an accessor for the current decoder
* Use std::unique_ptr<> for ownership.
* Remove "Release()" and "ReleaseDecoder()". Instead just delete.
* Remove |friend| relationship between CodecDatabase and VCMGenericDecoder.
VCMDecodedFrameCallback
* DCHECKs for thread correctness.
* Remove |lock_| now that a threading model has been established and verified.
VCMGenericDecoder
* All methods now have thread checks.
* Variable access associated with thread checkers.
VideoReceiver
* Added two notification methods, DecoderThreadStarting() and DecoderThreadStopped()
* Allows us to establish a period when the decoder thread is not running and it is safe to modify variables such as callbacks, that are only read when the decoder thread is running.
* Allows us to DCHECK thread guarantees.
* Allows synchronizing callbacks from the module process thread and have them only active while the decoder thread is running.
* The above, allows us to establish two modes for the thread, single-threaded-mutable and multi-threaded-const.
* Using that knowledge, we can remove |receive_crit_| as well as locking for a number of member variables.
MediaCodecVideoDecoder
* Removed frame polling code from this class, since this is now done from the root thread function in VideoReceiveStream.
VideoReceiveStream
* On Android: Polls for decoded frames every 10ms (same interval as previously in MediaCodecVideoDecoder)
* [Un]Registers the |video_receiver_| with the module thread only around the time the decoder thread is started/stopped.
* Notifies the receiver of start/stop events of the decoder thread.
* Changed the decoder thread to use the new PlatformThread callback type.
BUG=webrtc:7361, 695438
Review-Url: https://codereview.webrtc.org/2764573002
Cr-Commit-Position: refs/heads/master@{#17527}
