Design of individual block in ExtendedReports packet suggest there is
no point to have more than one block per type.
This CL reduce complexity of having several blocks of the same type in
same report.
BUG=webrtc:5260
Review-Url: https://codereview.webrtc.org/2378113002
Cr-Commit-Position: refs/heads/master@{#14855}
Contains fixes for a non-perfect implementation in https://codereview.webrtc.org/2328433003/
Summary:
Adds WebRTC.Audio.RecordedOnlyZeros UMA stat when recording stops if:
- All level estimates during the audio session were zero, and
- If the audio session was longer than 10 seconds.
Adds four simple methods to the AudioDeviceBuffer (ADB) class to allow the ADM
to update the ADB about when media starts and stops in both directions.
Moves any "critical" parst out frome the timer (based on task queue) and ensures
that it only does trivial logging tasks.
The task queue is now owned by a unique pointer to improve control of when it
starts and stops.
Adds time measurements (for logging) of both total time playing out and total
recording time. Units are in milliseconds.
BUG=webrtc:6592
Review-Url: https://codereview.webrtc.org/2445363003
Cr-Commit-Position: refs/heads/master@{#14854}
The new code is only exercised in tests so far. The H264 profile-level-id
parsing is not complete, but it should be enough for our purposes for
now.
BUG=webrtc:6400,webrtc:6337
Review-Url: https://codereview.webrtc.org/2459633002
Cr-Commit-Position: refs/heads/master@{#14850}
This change will allow for a audio source to report its sampling rate
to the audio mixer. It is needed in order to mix at a lower sampling
rate. Mixing at a lower sampling rate can in many cases lead to big
efficiency improvements, as reported by experiments.
The code affected is all implementations of the Source interface:
AudioReceiveStream and a mock class. The AudioReceiveStream now
queries its underlying voe::Channel object for the needed frequency.
Note that the changes to the mixing algorithm are done in a later CL.
BUG=webrtc:6346
NOTRY=True
TBR=solenberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2448113009
Cr-Commit-Position: refs/heads/master@{#14839}
I have chosen part of 2435603010 changes to compose this change.
According to the discussion we have made in previous change, this CL contains,
1. Source structure to represent a source of a DesktopCapturer.
2. GetSourceList / SelectSource / FocusOnSelectedSource functions in
DesktopCapturer.
3. ScreenCapturer and WindowCapturer forward corresponding functions to the new
DesktopCapturer APIs.
After this change, We can remove WindowCapturer & ScreenCapturer references from
Chromium, and use the new APIs.
BUG=webrtc:6513
Committed: https://crrev.com/9cb0b3b4ac916cdf52d97a63d923dfbe73f0541e
Review-Url: https://codereview.webrtc.org/2452263003
Cr-Original-Commit-Position: refs/heads/master@{#14830}
Cr-Commit-Position: refs/heads/master@{#14832}
Reason for revert:
Build break in Chromium
Original issue's description:
> Add DesktopCapturer GetSourceList SelectSource FocusOnSelectedSource functions
>
> I have chosen part of 2435603010 changes to compose this change.
> According to the discussion we have made in previous change, this CL contains,
> 1. Source structure to represent a source of a DesktopCapturer.
> 2. GetSourceList / SelectSource / FocusOnSelectedSource functions in
> DesktopCapturer.
> 3. ScreenCapturer and WindowCapturer forward corresponding functions to the new
> DesktopCapturer APIs.
>
> After this change, We can remove WindowCapturer & ScreenCapturer references from
> Chromium, and use the new APIs.
>
> BUG=webrtc:6513
>
> Committed: https://crrev.com/9cb0b3b4ac916cdf52d97a63d923dfbe73f0541e
> Cr-Commit-Position: refs/heads/master@{#14830}
TBR=sergeyu@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6513
Review-Url: https://codereview.webrtc.org/2464553002
Cr-Commit-Position: refs/heads/master@{#14831}
I have chosen part of 2435603010 changes to compose this change.
According to the discussion we have made in previous change, this CL contains,
1. Source structure to represent a source of a DesktopCapturer.
2. GetSourceList / SelectSource / FocusOnSelectedSource functions in
DesktopCapturer.
3. ScreenCapturer and WindowCapturer forward corresponding functions to the new
DesktopCapturer APIs.
After this change, We can remove WindowCapturer & ScreenCapturer references from
Chromium, and use the new APIs.
BUG=webrtc:6513
Review-Url: https://codereview.webrtc.org/2452263003
Cr-Commit-Position: refs/heads/master@{#14830}
renamed kName to kUri and make it more const.
remove IsSupportedBy to reduce header dependency.
BUG=webrtc:1994
Review-Url: https://codereview.webrtc.org/2457783005
Cr-Commit-Position: refs/heads/master@{#14825}
This algorithm calculates an estimate of the Pearson product-moment correlation coefficient between the power of 10ms audio buffers taken from the render and capture sides, for various different delay values.
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2419563003
Cr-Commit-Position: refs/heads/master@{#14824}
Reason for revert:
Speculative revert. This CL is a plausible cause for breakages in internal projects.
Original issue's description:
> Removes all uses of FileWrapper in audio_device.
>
> BUG=webrtc:6463
>
> Committed: https://crrev.com/04055e95bf97d106053d90bcc9e974eb4ad175e6
> Cr-Commit-Position: refs/heads/master@{#14811}
TBR=sprang@webrtc.org,henrika@webrtc.org,palmkvist@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6463
Review-Url: https://codereview.webrtc.org/2459873002
Cr-Commit-Position: refs/heads/master@{#14822}
Introduce a convention on categorizing GN targets:
1. Production code
2. Tests
3. Examples
4. Tools
The first two have targets spread out all over the tree,
while the latter are isolated to examples/ and tools/ directories.
Another new convention: Each directory's BUILD.gn file shall contain
a target named similar to the directory name. This target shall
contain the 'most common' production code, i.e. so that most
consumers of the directory can depend on only the directory
(which implicitly means that target in GN).
//webrtc:webrtc_tests is changed to depend on all WebRTC tests.
From now on, it's necessary to add new test targets to this dependency
tree in order to get them compiled.
Two new group targets are created:
//webrtc/modules/audio_coding:audio_coding_tests
//webrtc/modules/audio_processing:audio_processing_tests
to reduce the long list of tests in //webrtc:webrtc_tests.
Visibility on //webrtc:webrtc and //webrtc:webrtc_tests is restricted
to the root target, to avoid circular dependencies due to the monolithic
property of these targets (a problem we've had in the past).
The 'root' target at the top level is renamed to 'default', which means GN will
build this target instead of _all_ generated targets
(see https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/faq.md#Can-I-control-what-targets-are-built-by-default).
This target now depends on everything we want to build, meaning all targets now
explicitly needs to be wired up from the root target in order to get build.
Having this, the number of compiled objects on Android is decreased from 8855 to 6276. It also gives us better control over our build.
BUG=webrtc:6440
TESTED=git cl try --clobber
NOTRY=True
Review-Url: https://codereview.webrtc.org/2441383002
Cr-Commit-Position: refs/heads/master@{#14821}
This CL does not contain the actual algorithm, but only creates an empty processing component and connects the right signals to it. The algorithm will be added in a follow-up CL.
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2405403003
Cr-Commit-Position: refs/heads/master@{#14820}
The test program audioproc_f does not call the stream_analog_level
method. This should be done do
1) Ensure that proper log output is produced when reproducing a call.
2) Ensure that this method is properly tested.
3) Ensure that the correct side-effects are triggered (this method
is not const).
BUG=webrtc:6564
Review-Url: https://codereview.webrtc.org/2449043008
Cr-Commit-Position: refs/heads/master@{#14817}
Reason for revert:
This CL breaks internal dependencies.
Original issue's description:
> New statistics interface for APM
>
> This adds functions to enable and retrieve statistics from APM. These functions are not yet called, which will happen in a follow-up CL.
>
> BUG=webrtc:6525
>
> Committed: https://crrev.com/8b8d3e4c30e8ea3846b58dfd36d1fd35a7799df4
> Cr-Commit-Position: refs/heads/master@{#14810}
TBR=peah@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2456333002
Cr-Commit-Position: refs/heads/master@{#14814}
aecdump recordings in an efficient manner, it is
important to be able to use a standardized analysis
script. For this to be feasible, data log points should
be present.
This CL adds those logpoints as well as the framework
needed to for those to work.
BUG=webrtc:6564
Review-Url: https://codereview.webrtc.org/2457783003
Cr-Commit-Position: refs/heads/master@{#14812}
This adds functions to enable and retrieve statistics from APM. These functions are not yet called, which will happen in a follow-up CL.
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2433153003
Cr-Commit-Position: refs/heads/master@{#14810}
Replaced with a size() method, returning the corresponding attribute
(_length) of the underlying EncodedImage.
BUG=None
Review-Url: https://codereview.webrtc.org/2444193010
Cr-Commit-Position: refs/heads/master@{#14809}
Fixing compile error for non Intel platforms
when using C implemented fft functions
(for example, optimizations are not implemented for mips64el
and C functions must be used)
Adding bypass of presubmit to avoid code style and header
errors caused by the fact that files with legacy code are
being renamed.
NOPRESUBMIT=true
BUG=webrtc:6595
TEST=gn gen out-gn/mips64-android-webrtc --args="is_debug=false target_os=\"android\" target_cpu=\"mips64el\""
ninja -C out-gn/mips64-android-webrtc audio_processing
Review-Url: https://codereview.webrtc.org/2442773002
Cr-Commit-Position: refs/heads/master@{#14808}
This is a trivial change, I just cutted and pasted part of the code in
screen_capturer_unittest.cc to screen_capturer_integration_test.cc, removed
DISABLED_ prefixes, and updated build file.
BUG=webrtc:6366
Review-Url: https://codereview.webrtc.org/2444583002
Cr-Commit-Position: refs/heads/master@{#14806}
This helps a lot to avoid reducing the bitrate too quickly when there's a short period of very few packets delivered, followed by the rate resuming at the regular rate. It specifically avoids the BWE going down to super low values as a response delay spikes.
BUG=webrtc:6566
R=terelius@webrtc.org
Review URL: https://codereview.webrtc.org/2422063002 .
Cr-Commit-Position: refs/heads/master@{#14802}
Enabled the plugin and cleaned up all issues it found, mainly virtual
destructors not being marked as override.
BUG=webrtc:163
Review-Url: https://codereview.webrtc.org/2436503004
Cr-Commit-Position: refs/heads/master@{#14793}
- Update MockAudioProcessing to current APM interface.
- Replace calls to VoEAudioProcessing::Start/StopAecDump with direct calls on APM.
- Add AudioProcessing* in WVoE, get it from VoE, so we can call directly on APM.
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/2446143002
Cr-Commit-Position: refs/heads/master@{#14786}
Include CVO in key frame.
Include CVO in delta frame when rotation changes.
Include CVO when it is non zero to support current receiver implementation.
BUG=webrtc:6600
Review-Url: https://codereview.webrtc.org/2452583002
Cr-Commit-Position: refs/heads/master@{#14784}
Reason for revert:
Internal project has been fixed
Original issue's description:
> Revert of Move bitstream parser to more appropriate directory. (patchset #4 id:60001 of https://codereview.webrtc.org/2370853005/ )
>
> Reason for revert:
> Breaks internal project
>
> Original issue's description:
> > Move current bitstream parser to more appropriate directory.
> >
> > This CL groups together the code that has to do with parsing H264 bitstreams.
> > This code logically belongs together, and having it in the same directory not
> > only simplifies things from a project structure perspective, but also makes it
> > easier to refactor out common parts incrementally.
> > An added benefit is that this simplifies modular compilation, where for example
> > one would like a build of WebRTC without the H264 codec-specific parts.
> >
> > BUG=webrtc:6338
> >
> > Committed: https://crrev.com/cc6817e9ce4a5ffc73efb660cf0368afbc7d9a4f
> > Cr-Commit-Position: refs/heads/master@{#14684}
>
> TBR=magjed@webrtc.org,stefan@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6338
>
> Committed: https://crrev.com/f04f14e772f803de39f8a6128e5157127cd35103
> Cr-Commit-Position: refs/heads/master@{#14685}
TBR=magjed@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6338
Review-Url: https://codereview.webrtc.org/2434043002
Cr-Commit-Position: refs/heads/master@{#14783}
This refactoring allows runtime checks that functions that access
codec specific information are using the correct union member.
The API also allows replacing the union with another implementation
without changes at calling sites.
BUG=webrtc:6603
Review-Url: https://codereview.webrtc.org/2001533003
Cr-Commit-Position: refs/heads/master@{#14775}
Rename kFullSendSideEstimator -> kSendSideEstimator and add new class SendSideBweSender (controlled by kSendSideEstimator) that actually uses the send side BWE.
Move the mock to logging/rtc_event_log/mock.
Allow congestion_controller, remote_bitrate_estimator and audio to depend on loggging/rtc_event_log
BUG=webrtc:6526
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2431093003
Cr-Commit-Position: refs/heads/master@{#14772}
Several subcomponents inside APM copy render audio from
the render side to the capture side using the same
pattern. Currently this is done independently for the
submodules.
This CL moves the the AGC functionality for this into
APM.
BUG=webrtc:5298, webrtc:6540
Review-Url: https://codereview.webrtc.org/2444283002
Cr-Commit-Position: refs/heads/master@{#14770}
the render side to the capture side using the same
pattern. Currently this is done independently for the
submodules.
This CL moves the the AECM functionality for this into
APM.
BUG=webrtc:5298, webrtc:6540
Review-Url: https://codereview.webrtc.org/2444793005
Cr-Commit-Position: refs/heads/master@{#14768}
Remove functions to enumerate all extensions,
Remove concept of the inactive extension.
Decision if extension should be included into rtp header is done by rtp_sender
GetTotalLengthInBytes now calculates all extension, included or not.
That is used only for calculating how much space to reserve for fec.
Since extension might suddenly be included in the next packet (which still might belong to same fec group), it is safer to calculate all registered extension.
BUG=webrtc:5565, webrtc:1994
Review-Url: https://codereview.webrtc.org/2431253003
Cr-Commit-Position: refs/heads/master@{#14763}
The API has changed for the slice config of SSpatialLayerConfig as of
OpenH264 v1.6. Update H264EncoderImpl with an ifdef that uses the
correct API depending on what version of OpenH264 is being used.
BUG=webrtc:6583
Review-Url: https://codereview.webrtc.org/2440113002
Cr-Commit-Position: refs/heads/master@{#14762}
XGetImage() may return NULL and XServerPixelBuffer wasn't handling this
case properly.
BUG=649487
Review-Url: https://codereview.webrtc.org/2446733003
Cr-Commit-Position: refs/heads/master@{#14754}
It turns out that that audio network adaptor can always be created with knowledge of receiver frame length range. There is no need to keep some infrastructure that is used for runtime setting of receiver frame length ranges.
BUG=webrtc:6303
Review-Url: https://codereview.webrtc.org/2429503002
Cr-Commit-Position: refs/heads/master@{#14748}
PacketList is now list<Packet> instead of list<Packet*>.
Splicing the lists in NetEqImpl::InsertPacketInternal instead of
moving packets. Avoid moving the packet when doing Rfc3389Cng.
Removed PacketBuffer::DeleteFirstPacket and DeleteAllPackets.
BUG=chromium:657300
Review-Url: https://codereview.webrtc.org/2425223002
Cr-Commit-Position: refs/heads/master@{#14747}
