Software fallback wrapper now reports least common multiple of requirements
for two encoders.
SimulcastEncoderAdapter queries actual encoder before InitEncode call
and requests alignment for all layers if simulcast is not supported by
any of the encoders.
Bug: chromium:1084702
Change-Id: Iaed8190737125d447036b6c664b863be72556a5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225881
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34466}
CL partially auto-generated with:
git grep -l "\bassert(" | grep "\.[c|h]" | \
xargs sed -i 's/\bassert(/RTC_DCHECK(/g'
And with:
git grep -l "RTC_DCHECK(false)" | \
xargs sed -i 's/RTC_DCHECK(false)/RTC_NOTREACHED()/g'
With some manual changes to include "rtc_base/checks.h" where
needed.
A follow-up CL will remove assert() from Obj-C code as well
and remove the #include of <assert.h>.
The choice to replace with RTC_DCHECK is because assert()
is because RTC_DCHECK has similar behavior as assert()
based on NDEBUG.
This CL also contains manual changes to switch from
basic RTC_DCHECK to other (preferred) versions like
RTC_DCHECK_GT (and similar).
Bug: webrtc:6779
Change-Id: I00bed8886e03d685a2f42324e34aef2c9b7a63b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224846
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34442}
This is needed to facilitate dumping of stats to CSV in tests.
Bug: none
Change-Id: Ic78a4630f70a9238d26161ac89c205903dfc852f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225300
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34435}
When the transport is terminated, if an error has occured, it will
be propagated to the channels.
When such errors can happen at the SCTP level (e.g. out of resources),
RTCError may contain an error code matching the definition at
https://www.iana.org/assignments/sctp-parameters/sctp-parameters.xhtml#sctp-parameters-24
If the m= line is rejected or removed from SDP, an error will again be sent
to the data channels, signaling their unexpected transition to closed.
Bug: webrtc:12904
Change-Id: Iea3d8aba0a57bbedb5d03f0fb6f7aba292e92fe8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223541
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34386}
Relanding after adding to chromium stats whitelist:
https://chromium-review.googlesource.com/c/chromium/src/+/2983329
This solves two problems:
* Echo return loss stats weren't being gathered in Chrome, because they
need to be taken from the audio processor attached to the track
rather than the audio send stream.
* The standardized location is in RTCAudioSourceStats, not
RTCMediaStreamTrackStats. For now, will populate the stats in both
locations.
Bug: webrtc:12770
Change-Id: I3633ee428d07b283b0cc503a84d8fa2e79415dfb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223761
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34367}
This class takes RtpPacketReceived and assembles them into RtpFrameObjects.
Change-Id: Ia9785d069fecccc1d5b81efd257f33c8bd7a778b
Bug: webrtc:7408, webrtc:12579
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222580
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34364}
This reverts commit a27cfbffdf.
Reason for revert: WebRtcBrowserTest.RunsAudioVideoWebRTCCallInTwoTabsGetStatsPromise failing.
Original change's description:
> Fix echo return loss stats and add to RTCAudioSourceStats.
>
> This solves two problems:
> * Echo return loss stats weren't being gathered in Chrome, because they
> need to be taken from the audio processor attached to the track
> rather than the audio send stream.
> * The standardized location is in RTCAudioSourceStats, not
> RTCMediaStreamTrackStats. For now, will populate the stats in both
> locations.
>
> Bug: webrtc:12770
> Change-Id: I47eaf7f2b50b914a1be84156aa831e27497d07e3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223182
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34344}
TBR=deadbeef@webrtc.org,hbos@webrtc.org,hbos@chromium.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I6b2587d762f005adef67c0d5121f1b58c3b76688
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12770
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223068
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#34352}
This solves two problems:
* Echo return loss stats weren't being gathered in Chrome, because they
need to be taken from the audio processor attached to the track
rather than the audio send stream.
* The standardized location is in RTCAudioSourceStats, not
RTCMediaStreamTrackStats. For now, will populate the stats in both
locations.
Bug: webrtc:12770
Change-Id: I47eaf7f2b50b914a1be84156aa831e27497d07e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223182
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34344}
This is a reland of e6ee8fab7e
Original change's description:
> Deprecate microsecond timestamps in RTC event log.
>
> (Microsecond timestamps are only used in the legacy wire-format,
> and the clocks only have microsecond resolution on some platforms.)
>
> Also convert structs on the parsing side to use a Timestamp instead
> of a uint64_t to represent the log time.
>
> Bug: webrtc:11933
> Change-Id: Ide5a0217d99f13f2e243115b163f13e0525648c7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219467
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34097}
Bug: webrtc:11933
Change-Id: I295be966ee96b50719ceb4690dad7e7ce958dbac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221361
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34321}
Also applies sanitizing to prflx candidates, not just local ones.
Also add tests for the port allocator Sanitize function.
Bug: chromium:1218346
Change-Id: Ifbc7843cd6e289c09ca72b6ec610a34bbbf7e04e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222581
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34292}
kron is an owner of UMA metrics for WebRTC in chromium,
see tools/metrics/histograms/histograms_xml/web_rtc/OWNERS
Bug: webrtc:12096
Change-Id: I9804d747fc4e52d2ed2a9d96cc4ed315639210da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221961
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34268}
This is a change from the previous 100Hz frequency.
Also changing the locks slightly in AcmReceiver so that grabbing the
neteq lock right after we've let it go, isn't necessary inside of
AcmReceiver::GetAudio and also to avoid grabbing the neteq lock while
holding the AcmReceiver lock.
Bug: webrtc:12868
Change-Id: If6ee35f3dca20eb5bdbc615123aa099ccecf57c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221371
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34258}
Bilinear is faster but lesser quality, box is best quality. Our code
base has disagreed about which filter to use for quite some time,
causing aliasing bug reports. In an effort to avoid aliasing artifacts
and make our scaling filters more predictable, we're updating all uses
to kFilterBox.
WebRTC already uses kFilterBox everywhere except for these three
places. The main discrepency was between Chromium and WebRTC but that
has already been fixed. This CL fixes the last remaining bilinears.
This brings the WebRTC kFilterBox use count up from 11 to 14 and the
kFilterBilinear use count down from 3 to 0.
Bug: chromium:1212630
Change-Id: I5fe4aa92b9275d65b91ea97925533055d190d317
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221372
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34248}
https://crbug.com/1212611 was fixed. I confirmed that the Chrome
build works with these includes removed.
Bug: None
Change-Id: I68b0b6fed0562c6f6c981ddb326aa628352382fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221200
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34216}
This way we can have custom implementation of RtpTransportControllerSendInterface and pass it properly to Call.
Call relies on RtpTransportControllerSendInterface already so this is natural way to customize RTP related classes.
If there is custom factory present in dependencies it will be used, otherwise default factory will be used.
Intention behind this change is to have ability to have custom QoS with custom parameters.
Bug: webrtc:12778
Change-Id: I5b88957025621ef4bcd63eaa98c218ad213da9c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217769
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@nvidia.com>
Cr-Commit-Position: refs/heads/master@{#34181}
Deprecate CreateDataChannel, and make it a simple wrapper function.
Bug: webrtc:12796
Change-Id: I053d75a264596ba87ca734a29df9241de93a80c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219784
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34130}
Reland of commit a743303211
Previously, RTP header extensions with encryption had been filtered
if the encryption had been activated (not the other way around) which
was likely an unintended logic inversion.
In addition, it ensures that encrypted RTP header extensions are only
negotiated if RTP header extension encryption is turned on. Formerly,
which extensions had been negotiated depended on the order in which
they were inserted, regardless of whether or not header encryption was
actually enabled, leading to no extensions being sent on the wire.
Further changes:
- If RTP header encryption enabled, prefer encrypted extensions over
non-encrypted extensions
- Add most extensions to list of extensions supported for encryption
- Discard encrypted extensions in a session description in case encryption
is not supported for that extension
- Mark FindHeaderExtensionByUri without filter argument as deprecated
Bug: webrtc:11713
Change-Id: I52a5ade1b94bc01d1c2a35cb56023684fcaf9982
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219081
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34129}
This reverts commit 096ad02c02.
Reason for revert: Including a fix for the test issue.
Original change's description:
> Revert "Fix race between enabled() and set_enabled() in VideoTrack."
>
> This reverts commit 5ffefe9d2d.
>
> Reason for revert: Breaks Chromium Android browser tests on fyi bots.
>
> Original change's description:
> > Fix race between enabled() and set_enabled() in VideoTrack.
> >
> > Along the way I introduced VideoSourceBaseGuarded, which is equivalent
> > to VideoSourceBase except that it applies thread checks. I found that
> > it's easy to use VideoSourceBase incorrectly and in fact there appear
> > to be tests that do this.
> >
> > I made the source object const in VideoTrack, as it already was in
> > AudioTrack, and that allowed for making the GetSource() accessors
> > bypass the proxy thread hop and give the caller direct access.
> >
> > Bug: webrtc:12773, b/188139639, webrtc:12780
> > Change-Id: I022175c4239a1306ef54059c131d81411d5124fe
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219160
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Reviewed-by: Andrey Logvin <landrey@webrtc.org>
> > Commit-Queue: Tommi <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#34096}
>
> TBR=mbonadei@webrtc.org,tommi@webrtc.org,landrey@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
>
> Change-Id: I16323d459c76eb6a87cc602a0048f6ee01c81626
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:12773
> Bug: b/188139639
> Bug: webrtc:12780
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219637
> Reviewed-by: Evan Shrubsole <eshr@google.com>
> Commit-Queue: Evan Shrubsole <eshr@google.com>
> Cr-Commit-Position: refs/heads/master@{#34101}
# Not skipping CQ checks because this is a reland.
Bug: webrtc:12773
Bug: b/188139639
Bug: webrtc:12780
Change-Id: Ib35fe15a6c43de8f286d60aff02b19df1ab76925
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219639
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34104}
This is loosely based on the similar implementation in gecko.
Bug: webrtc:9965
Change-Id: I5203a05e1c34ca6f97bd1b143790f95ff245e340
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219791
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Doudou Kisabaka <doudouk@google.com>
Cr-Commit-Position: refs/heads/master@{#34102}
This reverts commit 5ffefe9d2d.
Reason for revert: Breaks Chromium Android browser tests on fyi bots.
Original change's description:
> Fix race between enabled() and set_enabled() in VideoTrack.
>
> Along the way I introduced VideoSourceBaseGuarded, which is equivalent
> to VideoSourceBase except that it applies thread checks. I found that
> it's easy to use VideoSourceBase incorrectly and in fact there appear
> to be tests that do this.
>
> I made the source object const in VideoTrack, as it already was in
> AudioTrack, and that allowed for making the GetSource() accessors
> bypass the proxy thread hop and give the caller direct access.
>
> Bug: webrtc:12773, b/188139639, webrtc:12780
> Change-Id: I022175c4239a1306ef54059c131d81411d5124fe
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219160
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Andrey Logvin <landrey@webrtc.org>
> Commit-Queue: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34096}
TBR=mbonadei@webrtc.org,tommi@webrtc.org,landrey@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I16323d459c76eb6a87cc602a0048f6ee01c81626
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12773
Bug: b/188139639
Bug: webrtc:12780
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219637
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#34101}
This reverts commit e6ee8fab7e.
Reason for revert: Breaks downstream test
Original change's description:
> Deprecate microsecond timestamps in RTC event log.
>
> (Microsecond timestamps are only used in the legacy wire-format,
> and the clocks only have microsecond resolution on some platforms.)
>
> Also convert structs on the parsing side to use a Timestamp instead
> of a uint64_t to represent the log time.
>
> Bug: webrtc:11933
> Change-Id: Ide5a0217d99f13f2e243115b163f13e0525648c7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219467
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34097}
TBR=terelius@webrtc.org,srte@webrtc.org,crodbro@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I478c9a4a1664b984891c4fcfc78f0ce9a51fe4c0
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11933
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219636
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34100}
This fixes the WebRTC roll in Chrome, which was failing because the
//third_party/nearby library was transively including Chromes //base/check
via WebRTC overrides by including
api/peer_connection_interface.h
which included api/proxy.h
which included rtc_base/event.h
which was overrided in webrtc_overrides.
This override contains an include to base/check.h.
Adding a temporary dependency on event.h fixes this for now. This can be
removed once https://crbug.com/1212611 is fixed.
Bug: None
Change-Id: Id372e0737664d4a94ade0f75ae5fcf21f3db929f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219630
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#34098}
(Microsecond timestamps are only used in the legacy wire-format,
and the clocks only have microsecond resolution on some platforms.)
Also convert structs on the parsing side to use a Timestamp instead
of a uint64_t to represent the log time.
Bug: webrtc:11933
Change-Id: Ide5a0217d99f13f2e243115b163f13e0525648c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219467
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34097}
Along the way I introduced VideoSourceBaseGuarded, which is equivalent
to VideoSourceBase except that it applies thread checks. I found that
it's easy to use VideoSourceBase incorrectly and in fact there appear
to be tests that do this.
I made the source object const in VideoTrack, as it already was in
AudioTrack, and that allowed for making the GetSource() accessors
bypass the proxy thread hop and give the caller direct access.
Bug: webrtc:12773, b/188139639, webrtc:12780
Change-Id: I022175c4239a1306ef54059c131d81411d5124fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219160
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34096}
This change prepares for a later change in Chromium that makes it
stop depending on headers exposed by WebRTC that require inclusion of
api/proxy.h.
No-Try because of lack of infra lack of capacity on macs.
No-Try: True
Bug: webrtc:12787
Change-Id: I628424fe49e873027595b80336be2b821c22245e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219688
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34072}
This also changes the meaning of |estimated_capture_clock_offset| in
|absolute_capture_time_| to become a remote to capturer clock offset.
Bug: chromium:1056230, webrtc:10739
Change-Id: Id658590e027bbe77ae0834ea224e1dc977a305f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219163
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#34067}
This change enables the use of two different adaptation speeds of the
matched filter of the delay estimator of AEC3.
One speed is used when no delay has been found, and one is used after a
reliable delay has been found. The purpose is to use a slower adaptation
speed to reduce the risk of divergence during double-talk without
slowing down the search for the initial delay.
The CL prepares for experimentation by adding field trials for
controlling the two adaptation speeds.
Bug: webrtc:12775
Change-Id: I817a1ab5ded0f78d20de45edcf04c708290173fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219083
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34055}
For implementations where the signaling and worker threads are not
the same thread, this significantly cuts down on Thread::Invoke()s that
would block the signaling thread while waiting for the worker thread.
For Audio and Video Rtp receivers, the following methods now do not
block the signaling thread:
* GetParameters
* SetJitterBufferMinimumDelay
* GetSources
* SetFrameDecryptor / GetFrameDecryptor
* SetDepacketizerToDecoderFrameTransformer
Importantly this change also makes the track() accessor accessible
directly from the application thread (bypassing the proxy) since
for receiver objects, the track object is const.
Other changes:
* Remove RefCountedObject inheritance, use make_ref_counted instead.
* Every member variable in the rtp receiver classes is now RTC_GUARDED
* Stop() now fully clears up worker thread state, and Stop() is
consistently called before destruction. This means that there's one
thread hop instead of at least 4 before (sometimes more), per receiver.
* OnChanged triggered volume for audio tracks is done asynchronously.
* Deleted most of the JitterBufferDelay implementation. Turns out that
it was largely unnecessary overhead and complexity.
It seems that these two classes are copy/pasted to a large extent
so further refactoring would be good in the future, as to not have to
fix each issue twice.
Bug: chromium:1184611
Change-Id: I1ba5c3abbd1b0571f7d12850d64004fd2d83e5e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218605
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34022}
This is one step in getting rid of cricket::MediaType.
Bug: webrtc:12754
Fixes: webrtc:12764
Change-Id: Idee832572bdc4c0e3bfdec6fb31ec0ba9db3e995
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218346
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33994}
Applications should use CreatePeerConnectionOrError instead.
Moved fallback implementations of CreatePeerConnection into the
api/peer_connection_interface.h file, so that we do not have to
declare these methods in the proxy.
Bug: webrtc:12238
Change-Id: I70c56336641c2a108b68446ae41f43409277a584
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217762
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33964}
It doesn't make sense to use negative values or 0 to disable the
feature, so we use an optional int value.
Values bigger than 65535 are clamped down.
Bug: webrtc:12730
Change-Id: I6bd9cd92f7d0a70a78cf5a7c91dca52c28d08ba1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217760
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33954}
This reverts commit 793bac569f.
Reason for revert: rare compilation error fixed
Original change's description:
> Revert "Refactor the PlatformThread API."
>
> This reverts commit c89fdd716c.
>
> Reason for revert: Causes rare compilation error on win-libfuzzer-asan trybot.
> See https://ci.chromium.org/p/chromium/builders/try/win-libfuzzer-asan-rel/713745?
>
> Original change's description:
> > Refactor the PlatformThread API.
> >
> > PlatformThread's API is using old style function pointers, causes
> > casting, is unintuitive and forces artificial call sequences, and
> > is additionally possible to misuse in release mode.
> >
> > Fix this by an API face lift:
> > 1. The class is turned into a handle, which can be empty.
> > 2. The only way of getting a non-empty PlatformThread is by calling
> > SpawnJoinable or SpawnDetached, clearly conveying the semantics to the
> > code reader.
> > 3. Handles can be Finalized, which works differently for joinable and
> > detached threads:
> > a) Handles for detached threads are simply closed where applicable.
> > b) Joinable threads are joined before handles are closed.
> > 4. The destructor finalizes handles. No explicit call is needed.
> >
> > Fixed: webrtc:12727
> > Change-Id: Id00a0464edf4fc9e552b6a1fbb5d2e1280e88811
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215075
> > Commit-Queue: Markus Handell <handellm@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Reviewed-by: Tommi <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33923}
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> TBR=handellm@webrtc.org
>
> Bug: webrtc:12727
> Change-Id: Ic0146be8866f6dd3ad9c364fb8646650b8e07419
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217583
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Reviewed-by: Markus Handell <handellm@webrtc.org>
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33936}
# Not skipping CQ checks because this is a reland.
Bug: webrtc:12727
Change-Id: Ifd6f44eac72fed84474277a1be03eb84d2f4376e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217881
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33950}
This reverts commit c89fdd716c.
Reason for revert: Causes rare compilation error on win-libfuzzer-asan trybot.
See https://ci.chromium.org/p/chromium/builders/try/win-libfuzzer-asan-rel/713745?
Original change's description:
> Refactor the PlatformThread API.
>
> PlatformThread's API is using old style function pointers, causes
> casting, is unintuitive and forces artificial call sequences, and
> is additionally possible to misuse in release mode.
>
> Fix this by an API face lift:
> 1. The class is turned into a handle, which can be empty.
> 2. The only way of getting a non-empty PlatformThread is by calling
> SpawnJoinable or SpawnDetached, clearly conveying the semantics to the
> code reader.
> 3. Handles can be Finalized, which works differently for joinable and
> detached threads:
> a) Handles for detached threads are simply closed where applicable.
> b) Joinable threads are joined before handles are closed.
> 4. The destructor finalizes handles. No explicit call is needed.
>
> Fixed: webrtc:12727
> Change-Id: Id00a0464edf4fc9e552b6a1fbb5d2e1280e88811
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215075
> Commit-Queue: Markus Handell <handellm@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33923}
# Not skipping CQ checks because original CL landed > 1 day ago.
TBR=handellm@webrtc.org
Bug: webrtc:12727
Change-Id: Ic0146be8866f6dd3ad9c364fb8646650b8e07419
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217583
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33936}
PlatformThread's API is using old style function pointers, causes
casting, is unintuitive and forces artificial call sequences, and
is additionally possible to misuse in release mode.
Fix this by an API face lift:
1. The class is turned into a handle, which can be empty.
2. The only way of getting a non-empty PlatformThread is by calling
SpawnJoinable or SpawnDetached, clearly conveying the semantics to the
code reader.
3. Handles can be Finalized, which works differently for joinable and
detached threads:
a) Handles for detached threads are simply closed where applicable.
b) Joinable threads are joined before handles are closed.
4. The destructor finalizes handles. No explicit call is needed.
Fixed: webrtc:12727
Change-Id: Id00a0464edf4fc9e552b6a1fbb5d2e1280e88811
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215075
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33923}
Before this CL, timestamps of received packets were rounded
to the nearest millisecond and stored as int64_t. Due to the
rounding it sometimes happened that timestamps later in the
pipeline that are not rounded seem to occur even before the
video frame was received.
Change-Id: I92d8f3540b23baae2d4a1dc6a7cb3f58bcdaad18
Bug: webrtc:12722
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216398
Reviewed-by: Chen Xing <chxg@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33916}
QueryCodecSupport() is needed in the coming work with extending
the MediaCapabilities API to WebRTC. In the API, a user can query
whether a specified media configuration is supported, power
efficient, and smooth. QueryCodecSupport() will be used to determine
if the configuration is supported and power efficient.
Bug: chromium:1187565
Change-Id: Ib1d93433a180b433f0bf60d6e871d03cebc4c0a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215922
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33843}
SdpVideoFormat is used to configure video encoder and decoders.
This CL adds support for comparing two SdpVideoFormat objects
to determine if they specify the same video codec.
This functionality previously only existed in media/base/codec.h
which made the code sensitive to circular dependencies. Once
downstream projects stop using cricket::IsSameCodec, this code
can be removed.
Bug: chromium:1187565
Change-Id: I242069aa6af07917637384c80ee4820887defc7d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213427
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33794}
On a 32bit system, this reduces the allocation size of the flag
down from 12 bytes to 8, and removes the need for a vtable (the extra
4 bytes are the vtable pointer).
The downside is that this change makes the binary layout of the
flag, less compatible with RefCountedObject<> based reference counting
objects and thus we don't immediately get the benefits of identical
COMDAT folding and subsequently there's a slight binary size increase.
With wider use, the binary size benefits will come.
Bug: none
Change-Id: I04129771790a3258d6accaf0ab1258b7a798a55e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215681
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33793}
The ERLE is used to estimate residual echo for echo suppression. The
ERLE is reduced during far-end offset to avoid echo leakage. When there
is a strong near-end present this can cause unnecessary transparency loss.
This change adds an ERLE estimation that does not compensate for onsets and
uses it for residual echo estimation when the suppressor considers the near-end to be dominant.
Bug: webrtc:12686
Change-Id: Ida78eeacf1f95c6e62403f86ba3f2ff055898a84
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215323
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33786}
This is a refactor to simplify a follow-up CL of adding
SdpVideoFormat::IsSameCodec.
The original files media/base/h264_profile_level_id.* and
media/base/vp9_profile.h must be kept until downstream projects
stop using them.
Bug: chroimium:1187565
Change-Id: Ib39eca095a3d61939a914d9bffaf4b891ddd222f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215236
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33782}
Since there is only a single type of DataChannel now, the enum was only used
when data channels were disabled at the PC API. That option has been
deprecated 4 years ago, it's now time to remove it.
Bug: webrtc:6625
Change-Id: I9e4ada1756da186e9639dd0fbf0249c55ea0b6c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215661
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33778}
SequenceChecker needs to be prefixed with & in RTC_DCHECK_RUN_ON;
all examples except the first one were showing this.
Bug: none
Change-Id: I90468689675319f9df67eb04a5d4cc0767ffb7a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215582
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33771}
This denies the ability to request RTP data channels to callers.
Later CLs will rip out the actual code for creating these channels.
Bug: chromium:928706
Change-Id: Ibb54197f192f567984a348f1539c26be120903f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177901
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33740}
This reverts commit a743303211.
Reason for revert: Breaks downstream tests that attempt to call FindHeaderExtensionByUri with 2 arguments. Could you keep the old 2-argument method declaration and just forward the call to the new 3-argument method with a suitable no-op filter?
Original change's description:
> Fix RTP header extension encryption
>
> Previously, RTP header extensions with encryption had been filtered
> if the encryption had been activated (not the other way around) which
> was likely an unintended logic inversion.
>
> In addition, it ensures that encrypted RTP header extensions are only
> negotiated if RTP header extension encryption is turned on. Formerly,
> which extensions had been negotiated depended on the order in which
> they were inserted, regardless of whether or not header encryption was
> actually enabled, leading to no extensions being sent on the wire.
>
> Further changes:
>
> - If RTP header encryption enabled, prefer encrypted extensions over
> non-encrypted extensions
> - Add most extensions to list of extensions supported for encryption
> - Discard encrypted extensions in a session description in case encryption
> is not supported for that extension
>
> Note that this depends on https://github.com/cisco/libsrtp/pull/491 to get
> into libwebrtc (cherry-pick or bump libsrtp version). Otherwise, two-byte
> header extensions will prevent any RTP packets being sent/received.
>
> Bug: webrtc:11713
> Change-Id: Ia0779453d342fa11e06996d9bc2d3c826f3466d3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177980
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Taylor <deadbeef@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33723}
TBR=deadbeef@webrtc.org,terelius@webrtc.org,hta@webrtc.org,lennart.grahl@gmail.com
Change-Id: I7df6b0fa611c6496dccdfb09a65ff33ae4a52b26
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11713
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215222
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33727}
Previously, RTP header extensions with encryption had been filtered
if the encryption had been activated (not the other way around) which
was likely an unintended logic inversion.
In addition, it ensures that encrypted RTP header extensions are only
negotiated if RTP header extension encryption is turned on. Formerly,
which extensions had been negotiated depended on the order in which
they were inserted, regardless of whether or not header encryption was
actually enabled, leading to no extensions being sent on the wire.
Further changes:
- If RTP header encryption enabled, prefer encrypted extensions over
non-encrypted extensions
- Add most extensions to list of extensions supported for encryption
- Discard encrypted extensions in a session description in case encryption
is not supported for that extension
Note that this depends on https://github.com/cisco/libsrtp/pull/491 to get
into libwebrtc (cherry-pick or bump libsrtp version). Otherwise, two-byte
header extensions will prevent any RTP packets being sent/received.
Bug: webrtc:11713
Change-Id: Ia0779453d342fa11e06996d9bc2d3c826f3466d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177980
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33723}
Since we want most users to use the PeerConnection API, this is the
part that we should document.
If we want people to use other APIs, we can add to the file.
Bug: webrtc:12674
Change-Id: Icf14f218cf51c640e6f846f10b49dff84106dc21
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215066
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33707}
This is the result of compiling Chromium with
Wtautological-unsigned-zero-compare. For more details, see:
https://chromium-review.googlesource.com/c/chromium/src/+/2802412
Change-Id: I05cec6ae5738036a56beadeaa1dde5189edf0137
Bug: chromium:1195670
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213783
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33689}
This change adds support for emitting encoded frames
for recording when the decoder can't easily read out
encoded width and height as is the case for AV1 streams,
in which case the information is buried in OBUs. Downstream
project relies on resolution information being present for key
frames. With the change, VideoReceiveStream2 infers the
resolution from decoded frames, and supplies it in the
RecordableEncodedFrames.
Bug: chromium:1191972
Change-Id: I07beda6526206c80a732976e8e19d3581489b8fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214126
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33662}
This reverts commit 5a40b37105.
Reason for revert: Fixed the bug and ran layout tests.
Original change's description:
> Revert "Use the new DNS resolver API in PeerConnection"
>
> This reverts commit acf8ccb3c9.
>
> Reason for revert: Speculative revert for https://ci.chromium.org/ui/p/chromium/builders/try/win10_chromium_x64_rel_ng/b8851745102358680592/overview.
>
> Original change's description:
> > Use the new DNS resolver API in PeerConnection
> >
> > Bug: webrtc:12598
> > Change-Id: I5a14058e7f28c993ed927749df7357c715ba83fb
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212961
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33561}
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> TBR=hta@webrtc.org
>
> Bug: webrtc:12598
> Change-Id: Idc9853cb569849c49052f9cbd865614710fff979
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213188
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33591}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:12598
Change-Id: Ief7867f2f23de66504877cdab1b23a11df2d5de4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214120
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33647}
Before, the calls went through the signaling thread and
blocked while the operation completed on the worker.
Bug: webrtc:12601
Change-Id: I58991fa98a55d0fa9304a68bd85bb269f1f123d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212619
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33615}
Packets, chunks, parameters and error causes - the SCTP entities
that are sent on the wire - are buffers with fields that are stored
in big endian and that generally consist of a fixed header size, and
a variable sized part, that can e.g. be encoded sub-fields or
serialized strings.
The BoundedByteReader and BoundedByteWriter utilities make it easy
to read those fields with as much aid from the compiler as possible,
by having compile-time assertions that fields are not accessed
outside the buffer's span.
There are some byte reading functionality already in modules/rtp_rtcp,
but that module would be a bit unfortunate to depend on, and doesn't
have the compile time bounds checking that is the biggest feature of
this abstraction of an rtc::ArrayView.
Bug: webrtc:12614
Change-Id: I9fc641aff22221018dda9add4e2c44853c0f64f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212967
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33597}
This API should allow existing factories to be used unmodified, but
offers a new API that documents ownership better and does not use
sigslot.
Bug: webrtc:12598
Change-Id: I0f68371059cd4a18ab07b87fc0e7526dcc0ac669
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212609
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33553}
Deleting obsolete stats. Spec: https://www.w3.org/TR/webrtc-stats/
1. RTCInbound/OutboundRtpStats.isRemote: No longer useful with remote stream stats
2. RTCIceCandidateStats.deleted: This field was obsoleted because if the ICE candidate is deleted it no longer appears in getStats()
I also marked as many other obsoleted stats possible according to spec. I am not as confident to delete them but feel free to comment to let me know if anything is off / can be deleted.
Bug: webrtc:12583
Change-Id: I688d0076270f85caa86256349753e5f0e0a44931
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211781
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33549}
Changes:
- adding the `RTCRemoteOutboundRtpStreamStats` dictionary (see [1])
- collection of remote outbound stats (only for audio streams)
- adding `remote_id` to the inbound stats and set with the ID of the
corresponding remote outbound stats only if the latter are available
- unit tests
[1] https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats
Tested: verified from chrome://webrtc-internals during an appr.tc call
Bug: webrtc:12529
Change-Id: Ide91dc04a3c387ba439618a9c6b64a95994a1940
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211042
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33545}
This avoids falling back on the VideoFrameBuffer::CropAndScale default
implementation which performs ToI420. This has two major benefits:
1. We save CPU by not converting to I420 for NV12 frames.
2. We make is possible for simulcast encoders to use Scale() and be
able to trust that the scaled simulcast layers have the same pixel
format as the top layer, which is required by libvpx.
In order to invoke NV12Buffer::CropAndScaleFrom() without introducing a
circular dependency, nv12_buffer.[h/cc] is moved to the "video_frame"
build target.
Bug: webrtc:12595, webrtc:12469
Change-Id: I81aac5c6b3e81c49f32a7be6dc2640e6b40f7692
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212643
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33521}
0 is not valid per https://tools.ietf.org/html/rfc5245#section-15.1 which says
<component-id>: is a positive integer between 1 and 256
This is part of the RTCIceTransport extension API which is probably going away.
BUG=chromium:1044875
Change-Id: I56d8dec79d3191e084f4a25a2c0a4d0b67afde74
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212642
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#33515}
This reverts commit 31c5c9da35.
Reason for revert: made QP parser thread-safe https://webrtc.googlesource.com/src/+/0e42cf703bd111fde235d06d08b02d3a7b02c727
Original change's description:
> Revert "Reland "Enable quality scaling when allowed""
>
> This reverts commit 0021fe7793.
>
> Reason for revert: Broken on linux_tsan bot: https://ci.chromium.org/ui/p/webrtc/builders/ci/Linux%20Tsan%20v2/25329/overview
>
> Original change's description:
> > Reland "Enable quality scaling when allowed"
> >
> > This reverts commit eb449a979b.
> >
> > Reason for revert: Added QP parsing in https://webrtc.googlesource.com/src/+/8639673f0c098efc294a7593fa3bd98e28ab7508
> >
> > Original change's description:
> > Before this CL quality scaling was conditioned on scaling settings
> > provided by encoder. That should not be a requirement since encoder
> > may not be aware of quality scaling which is a WebRTC feature. In M90
> > chromium HW encoders do not provide scaling settings (chromium:1179020).
> > The default scaling settings provided by these encoders are not correct
> > (b/181537172).
> >
> > This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
> > is set to true in singlecast with normal video feed (not screen sharing)
> > mode. If quality scaling is allowed it is enabled no matter whether
> > scaling settings are present in encoder info or not. Setting from
> > QualityScalingExperiment are used in case if not provided by encoder.
> >
> > Bug: chromium:1179020
> > Bug: webrtc:12511
> > Change-Id: I97911fde9005ec25028a640a3f007d12f2bbc2e5
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211349
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> > Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33438}
>
> TBR=brandtr@webrtc.org,ilnik@webrtc.org,ssilkin@webrtc.org,rubber-stamper@appspot.gserviceaccount.com
>
> Change-Id: Id7633a1e98f95762e81487887f83a0c35f89195c
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:1179020
> Bug: webrtc:12511
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211352
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33439}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:1179020
Bug: webrtc:12511
Change-Id: I3a31e1c1fdf7da93226f8c1e0a96b43fe0b786df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212026
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33481}
Use PendingTaskSafetyFlag for safe Stop. Followup to
https://webrtc-review.googlesource.com/c/src/+/209181.
Also fix rtc::scoped_refptr to work with RTC_PT_GUARDED_BY.
Bug: webrtc:12339
Change-Id: Ic0e3ecb17049f1a0e6af887ba5f97a5b48a32d98
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211351
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33447}
This reverts commit 0021fe7793.
Reason for revert: Broken on linux_tsan bot: https://ci.chromium.org/ui/p/webrtc/builders/ci/Linux%20Tsan%20v2/25329/overview
Original change's description:
> Reland "Enable quality scaling when allowed"
>
> This reverts commit eb449a979b.
>
> Reason for revert: Added QP parsing in https://webrtc.googlesource.com/src/+/8639673f0c098efc294a7593fa3bd98e28ab7508
>
> Original change's description:
> Before this CL quality scaling was conditioned on scaling settings
> provided by encoder. That should not be a requirement since encoder
> may not be aware of quality scaling which is a WebRTC feature. In M90
> chromium HW encoders do not provide scaling settings (chromium:1179020).
> The default scaling settings provided by these encoders are not correct
> (b/181537172).
>
> This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
> is set to true in singlecast with normal video feed (not screen sharing)
> mode. If quality scaling is allowed it is enabled no matter whether
> scaling settings are present in encoder info or not. Setting from
> QualityScalingExperiment are used in case if not provided by encoder.
>
> Bug: chromium:1179020
> Bug: webrtc:12511
> Change-Id: I97911fde9005ec25028a640a3f007d12f2bbc2e5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211349
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33438}
TBR=brandtr@webrtc.org,ilnik@webrtc.org,ssilkin@webrtc.org,rubber-stamper@appspot.gserviceaccount.com
Change-Id: Id7633a1e98f95762e81487887f83a0c35f89195c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1179020
Bug: webrtc:12511
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211352
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33439}
This reverts commit eb449a979b.
Reason for revert: Added QP parsing in https://webrtc.googlesource.com/src/+/8639673f0c098efc294a7593fa3bd98e28ab7508
Original change's description:
Before this CL quality scaling was conditioned on scaling settings
provided by encoder. That should not be a requirement since encoder
may not be aware of quality scaling which is a WebRTC feature. In M90
chromium HW encoders do not provide scaling settings (chromium:1179020).
The default scaling settings provided by these encoders are not correct
(b/181537172).
This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
is set to true in singlecast with normal video feed (not screen sharing)
mode. If quality scaling is allowed it is enabled no matter whether
scaling settings are present in encoder info or not. Setting from
QualityScalingExperiment are used in case if not provided by encoder.
Bug: chromium:1179020
Bug: webrtc:12511
Change-Id: I97911fde9005ec25028a640a3f007d12f2bbc2e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211349
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33438}
Spec: https://www.w3.org/TR/webrtc-stats/#receivedrtpstats-dict*
According to the spec, |RTCReceivedRtpStreamStats| is the base class for |RTCInboundRtpStreamStats| and |RTCRemoteInboundRtpStreamStats|. This structure isn't visible in JavaScript but it's important to bring it up to spec for the C++ part. This CL adds the barebone |RTCReceivedRtpStreamStats| with a bunch of TODOs for later migrations.
This commit makes the minimum |RTCReceivedRtpStreamStats| and rebase |RTCInboundRtpStreamStats| and |RTCRemoteInboundRtpStreamStats| to use the new class as the parent class.
This commit also moves |jitter| and |packets_lost| to |RTCReceivedRtpStreamStats|, from |RTCInboundRtpStreamStats| and |RTCRemoteInboundRtpStreamStats|. Moving these two first because they are the two that exist in both subclasses for now.
Bug: webrtc:12532
Change-Id: I0ec74fd241f16c1e1a6498b6baa621ca0489f279
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210340
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33435}
Spec: https://www.w3.org/TR/webrtc-stats/#dom-rtcvideosourcestats-frames
Wiring up the "frames" stats with the cumulative fps counter on the video source.
Tests:
./out/Default/peerconnection_unittests
./out/Default/video_engine_tests
Bug: webrtc:12499
Change-Id: I4103f56ed04cb464f5f7e70fbf2d77c25a867a68
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208782
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33404}
This reverts commit 83be84bb74.
Reason for revert: Suspect of crbug.com/1185276
Original change's description:
> Reland "Enable quality scaling when allowed"
>
> This reverts commit 609b524dd3.
>
> Reason for revert: Disable QualityScalingAllowed_QualityScalingEnabled on iOS.
>
> Original change's description:
> Before this CL quality scaling was conditioned on scaling settings
> provided by encoder. That should not be a requirement since encoder
> may not be aware of quality scaling which is a WebRTC feature. In M90
> chromium HW encoders do not provide scaling settings (chromium:1179020).
> The default scaling settings provided by these encoders are not correct
> (b/181537172).
>
> This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
> is set to true in singlecast with normal video feed (not screen sharing)
> mode. If quality scaling is allowed it is enabled no matter whether
> scaling settings are present in encoder info or not. Setting from
> QualityScalingExperiment are used in case if not provided by encoder.
>
> Bug: chromium:1179020
> Bug: webrtc:12511
> Change-Id: Ia0923e5a62acdfdeb06f9aad5d670be8a0f8d746
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209643
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33385}
Bug: chromium:1179020
Bug: webrtc:12511
Change-Id: I7004014c5936176f8c125aeb55da91ce095b266e
TBR: ssilkin@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209708
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33393}
This reverts commit 609b524dd3.
Reason for revert: Disable QualityScalingAllowed_QualityScalingEnabled on iOS.
Original change's description:
Before this CL quality scaling was conditioned on scaling settings
provided by encoder. That should not be a requirement since encoder
may not be aware of quality scaling which is a WebRTC feature. In M90
chromium HW encoders do not provide scaling settings (chromium:1179020).
The default scaling settings provided by these encoders are not correct
(b/181537172).
This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
is set to true in singlecast with normal video feed (not screen sharing)
mode. If quality scaling is allowed it is enabled no matter whether
scaling settings are present in encoder info or not. Setting from
QualityScalingExperiment are used in case if not provided by encoder.
Bug: chromium:1179020
Bug: webrtc:12511
Change-Id: Ia0923e5a62acdfdeb06f9aad5d670be8a0f8d746
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209643
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33385}
This reverts commit 752cbaba90.
Reason for revert: The test VideoStreamEncoderTest.QualityScalingAllowed_QualityScalingEnabled seems to fail on iOS.
Original change's description:
> Enable quality scaling when allowed
>
> Before this CL quality scaling was conditioned on scaling settings
> provided by encoder. That should not be a requirement since encoder
> may not be aware of quality scaling which is a WebRTC feature. In M90
> chromium HW encoders do not provide scaling settings (chromium:1179020).
> The default scaling settings provided by these encoders are not correct
> (b/181537172).
>
> This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
> is set to true in singlecast with normal video feed (not screen sharing)
> mode. If quality scaling is allowed it is enabled no matter whether
> scaling settings are present in encoder info or not. Setting from
> QualityScalingExperiment are used in case if not provided by encoder.
>
> Bug: chromium:1179020, webrtc:12511
> Change-Id: I83d55319ce4b9f4fb143187ced94a16a778b4de3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209184
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33373}
Bug: chromium:1179020
Bug: webrtc:12511
Change-Id: Icabf2d9a034d359f79491f2c37f1044f17a7445d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209641
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33381}
Before this CL quality scaling was conditioned on scaling settings
provided by encoder. That should not be a requirement since encoder
may not be aware of quality scaling which is a WebRTC feature. In M90
chromium HW encoders do not provide scaling settings (chromium:1179020).
The default scaling settings provided by these encoders are not correct
(b/181537172).
This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
is set to true in singlecast with normal video feed (not screen sharing)
mode. If quality scaling is allowed it is enabled no matter whether
scaling settings are present in encoder info or not. Setting from
QualityScalingExperiment are used in case if not provided by encoder.
Bug: chromium:1179020, webrtc:12511
Change-Id: I83d55319ce4b9f4fb143187ced94a16a778b4de3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209184
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33373}
The new verification makes verification a function on a message.
It also stores the password used in the request message, so that
it is easily accessible when verifying the response.
Bug: chromium:1177125
Change-Id: I505df4b54214643a28a6b292c4e2262b9d97b097
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209060
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33366}
Tests:
./out/Default/peerconnection_unittests
Manually tested with Chromium to see the data populated
Spec: https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict*
Bug: webrtc:12506
Change-Id: I60ef8061fb31deab06ca5f115246ceb5a8cdc5ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208960
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33361}
This is a reland of a37f2bd942
Original change's description:
> Rename SIGNALING and WORKER to PRIMARY and SECONDARY
>
> This makes the proxy macros less confusing when the secondary thread
> is sometimes the worker thread, sometimes the networking thread.
>
> Bug: none
> Change-Id: I1a8cebb82d09be44fe40e80c861bcfb47b9928e8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208763
> Reviewed-by: Tommi <tommi@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33346}
Bug: none
Change-Id: If46a6679ac0fc947797dd7be87626ef7702faca2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208845
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33349}
This reverts commit a37f2bd942.
Reason for revert: Breaks compile step (e.g. https://ci.chromium.org/ui/p/webrtc/builders/ci/Android64%20Builder%20x64%20(dbg)/19773/overview).
Original change's description:
> Rename SIGNALING and WORKER to PRIMARY and SECONDARY
>
> This makes the proxy macros less confusing when the secondary thread
> is sometimes the worker thread, sometimes the networking thread.
>
> Bug: none
> Change-Id: I1a8cebb82d09be44fe40e80c861bcfb47b9928e8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208763
> Reviewed-by: Tommi <tommi@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33346}
TBR=hta@webrtc.org
Bug: none
Change-Id: I2014faab3392f445f56edd9e833d00000ebc5ca3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208840
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33347}
This makes the proxy macros less confusing when the secondary thread
is sometimes the worker thread, sometimes the networking thread.
Bug: none
Change-Id: I1a8cebb82d09be44fe40e80c861bcfb47b9928e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208763
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33346}
This will allow us to optimize the internal buffers of
webrtc::VideoFrame for the resolution(s) that we actually want to
encode.
Bug: webrtc:12469, chromium:1157072
Change-Id: If378b52b5e35aa9a9800c1f7dfe189437ce43253
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208540
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33342}
Trying to take my first stab at contributing to WebRTC and I chose to populate jitter stats for video RTP streams. Please yell at me if this isn't something I'm not supposed to pick up. Appreciate a review, thanks!
Bug: webrtc:12487
Change-Id: Ifda985e9e20b1d87e4a7268f34ef2e45b1cbefa3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208360
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33325}
Currently the echo canceller reference signal is high-pass filtered to
avoid the need of modeling the capture-side high-pass filter as part of
the echo path.
This can lead to the lowest frequency bins of the linear filter
diverging as there is little low-frequency content available for
training. Over time the filter can output an increasing amount of
low-frequency power, which in turn affects the filter's ability to
adapt properly.
Disabling the high-pass filtering of the echo canceller reference solves
this issue, resulting in improved filter convergence.
Bug: webrtc:12265
Change-Id: Ic526a4b1b73e1808cfcd96a8cdee801b96a27671
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208288
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33322}
This remove webrtc-specific macro that has no reason to be webrtc specific
ABSL_DEPRECATED takes a message parameter encouraging to write text how class or function is deprecated.
Bug: webrtc:12484
Change-Id: I89f1398f91dacadc37f7db469dcd985e3724e444
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208282
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33314}
This make it easier to create parameters from a single endpoint ptr.
Bug: None
Change-Id: Id64757353505a21c7731655e1b7a3178fa2e5ef8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207425
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33263}
Audio interruption metric is not implemented for codecs doing their own PLC.
R=ivoc@webrtc.org, jakobi@webrtc.org
Bug: b/177523033 webrtc:12456
Change-Id: I0aca6fa5c0ff617e76ee1e4ed8d95703c7097223
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206561
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Pablo Barrera González <barrerap@google.com>
Cr-Commit-Position: refs/heads/master@{#33229}
This reverts commit 6b143c1c06.
Reason for revert:
Relanding with updated expectations for SctpTransport::Information
based on TransceiverStateSurfacer in Chromium.
Original change's description:
> Revert "Fix unsynchronized access to mid_to_transport_ in JsepTransportController"
>
> This reverts commit 6cd4058504.
>
> Reason for revert: Breaks WebRTC Chromium FYI Bots
>
> First failure:
> https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Android%20Tests%20%28dbg%29%20%28L%20Nexus5%29/1925
>
> Failed tests:
> WebRtcDataBrowserTest.CallWithSctpDataAndMedia
> WebRtcDataBrowserTest.CallWithSctpDataOnly
>
> Original change's description:
> > Fix unsynchronized access to mid_to_transport_ in JsepTransportController
> >
> > * Added several thread checks to JTC to help with programmer errors.
> > * Avoid a few Invokes() to the network thread here and there such
> > as for fetching sctp transport name for getStats(). The transport
> > name is now cached when it changes on the network thread.
> > * JsepTransportController instances now get deleted on the network
> > thread rather than on the signaling thread + issuing an Invoke()
> > in the dtor.
> > * Moved some thread hops from JTC over to PC which is where the problem
> > exists and also (imho) makes it easier to see where hops happen in
> > the PC code.
> > * The sctp transport is now started asynchronously when we push down the
> > media description.
> > * PeerConnection proxy calls GetSctpTransport directly on the network
> > thread instead of to the signaling thread + blocking on the network
> > thread.
> > * The above changes simplified things for webrtc::SctpTransport which
> > allowed for removing locking from that class and delete some code.
> >
> > Bug: webrtc:9987, webrtc:12445
> > Change-Id: Ic89a9426e314e1b93c81751d4f732f05fa448fbc
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205620
> > Commit-Queue: Tommi <tommi@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33191}
>
> TBR=tommi@webrtc.org,hta@webrtc.org
>
> Change-Id: I7b2913d5133807589461105cf07eff3e9bb7157e
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9987
> Bug: webrtc:12445
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206466
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33204}
TBR=tommi@webrtc.org,hta@webrtc.org,guidou@webrtc.org
# Not skipping CQ checks because this is a reland.
Bug: webrtc:9987
Bug: webrtc:12445
Change-Id: Icb205cbac493ed3b881d71ea3af4fb9018701bf4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206560
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33219}
measured in the connectionstatechange event to connected which usually
happens once per connection.
BUG=webrtc:12383
Change-Id: Ida136c44bfe65e922627390747f8bee384603715
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204301
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Justin Uberti <juberti@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#33207}
This reverts commit 6cd4058504.
Reason for revert: Breaks WebRTC Chromium FYI Bots
First failure:
https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Android%20Tests%20%28dbg%29%20%28L%20Nexus5%29/1925
Failed tests:
WebRtcDataBrowserTest.CallWithSctpDataAndMedia
WebRtcDataBrowserTest.CallWithSctpDataOnly
Original change's description:
> Fix unsynchronized access to mid_to_transport_ in JsepTransportController
>
> * Added several thread checks to JTC to help with programmer errors.
> * Avoid a few Invokes() to the network thread here and there such
> as for fetching sctp transport name for getStats(). The transport
> name is now cached when it changes on the network thread.
> * JsepTransportController instances now get deleted on the network
> thread rather than on the signaling thread + issuing an Invoke()
> in the dtor.
> * Moved some thread hops from JTC over to PC which is where the problem
> exists and also (imho) makes it easier to see where hops happen in
> the PC code.
> * The sctp transport is now started asynchronously when we push down the
> media description.
> * PeerConnection proxy calls GetSctpTransport directly on the network
> thread instead of to the signaling thread + blocking on the network
> thread.
> * The above changes simplified things for webrtc::SctpTransport which
> allowed for removing locking from that class and delete some code.
>
> Bug: webrtc:9987, webrtc:12445
> Change-Id: Ic89a9426e314e1b93c81751d4f732f05fa448fbc
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205620
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33191}
TBR=tommi@webrtc.org,hta@webrtc.org
Change-Id: I7b2913d5133807589461105cf07eff3e9bb7157e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9987
Bug: webrtc:12445
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206466
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33204}
* Added several thread checks to JTC to help with programmer errors.
* Avoid a few Invokes() to the network thread here and there such
as for fetching sctp transport name for getStats(). The transport
name is now cached when it changes on the network thread.
* JsepTransportController instances now get deleted on the network
thread rather than on the signaling thread + issuing an Invoke()
in the dtor.
* Moved some thread hops from JTC over to PC which is where the problem
exists and also (imho) makes it easier to see where hops happen in
the PC code.
* The sctp transport is now started asynchronously when we push down the
media description.
* PeerConnection proxy calls GetSctpTransport directly on the network
thread instead of to the signaling thread + blocking on the network
thread.
* The above changes simplified things for webrtc::SctpTransport which
allowed for removing locking from that class and delete some code.
Bug: webrtc:9987, webrtc:12445
Change-Id: Ic89a9426e314e1b93c81751d4f732f05fa448fbc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205620
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33191}
We need to be able build chromium with rtc_include_tests = true. It
reveals a lot of targets that are not compatible with chromium but
aren't marked so.
`rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated.
Bug: webrtc:12404
Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33124}
adds metrics for bundle usage, differentiating between
* BUNDLE is not negotiated and there is only a datachannel,
* BUNDLE is not negotiated and there is at most one m-line per media type,
for unified-plan
* BUNDLE is not negotiated and there are multiple m-lines per media type,
* BUNDLE is negotiated and there is only a datachannel,
* BUNDLE is negotiated but there is at most one m-line per media type,
* BUNDLE is negotiated and there are multiple m-lines per media type,
and for plan-b
* BUNDLE is negotiated
* BUNDLE is not negotiated
BUG=webrtc:12383
Change-Id: I41afc4b08fd97239f3b16a8638d9753c69b46d22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202254
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33090}
As an alternative to attaching custom array of bytes.
Bug: b/178094662
Change-Id: I92dcbf04998d8206091125febc520ebfcc4bcebf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203264
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33069}
The extmap-allow-mixed SDP attribute signals that one- and two-byte RTP
header extensions can be mixed. In practice, this also means that WebRTC
will support two-byte RTP header extensions when this is signaled by
both peers.
Bug: webrtc:9985
Change-Id: I80a3f97bab162c7d9a5acf2cae07b977641c039d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197943
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33036}
This is a reland of 69241a93fb
Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which
affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5.
The original CL didn't attach the definition of the macro
NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have
to be related to //rtc_base anymore but to //rtc_base:threading).
Original change's description:
> Refactor rtc_base build targets.
>
> The "//rtc_base:rtc_base" build target has historically been one of the
> biggest targets in the WebRTC build. Big targets are the main source of
> circular dependencies and non-API types leakage.
>
> This CL is a step forward into splitting "//rtc_base:rtc_base" into
> smaller targets (as originally started in 2018).
>
> The only non-automated changes are (like re-wiring the build system):
> * The creation of //rtc_base/async_resolver.{h,cc} which allows to
> break a circular dependency (is has been extracted from
> //rtc_base/net_helpers.{h,cc}).
> * The creation of //rtc_base/internal/default_socket_server.{h,cc} to
> break another circular dependency.
>
> Bug: webrtc:9987
> Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32941}
Bug: webrtc:9987
Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33001}
This reverts commit 69241a93fb.
Reason for revert: Breaks WebRTC roll into Chromium.
Original change's description:
> Refactor rtc_base build targets.
>
> The "//rtc_base:rtc_base" build target has historically been one of the
> biggest targets in the WebRTC build. Big targets are the main source of
> circular dependencies and non-API types leakage.
>
> This CL is a step forward into splitting "//rtc_base:rtc_base" into
> smaller targets (as originally started in 2018).
>
> The only non-automated changes are (like re-wiring the build system):
> * The creation of //rtc_base/async_resolver.{h,cc} which allows to
> break a circular dependency (is has been extracted from
> //rtc_base/net_helpers.{h,cc}).
> * The creation of //rtc_base/internal/default_socket_server.{h,cc} to
> break another circular dependency.
>
> Bug: webrtc:9987
> Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32941}
TBR=mbonadei@webrtc.org,hta@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
No-Try: True
Bug: webrtc:9987
Change-Id: I1e36ad64cc60092f38d6886153a94f1a58339256
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201840
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32986}
This adds a new way to poll decoder metadata.
A default implementation still delegates to the old methods.
Root call site is updates to not use the olds methods.
Follow-ups will dismantle usage of the olds methods in wrappers.
Bug: webrtc:12271
Change-Id: Id0fa6863c96ff9e3b849da452d6540e7c5da4512
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196520
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32976}
This is a reland of 444e04be69
Reason for reland: resolved the breaks from downstream project
Original change's description:
> ChannelStatistics used for RTP stats in VoipStatistics.
>
> - Added local and remote RTP statistics query API.
> - Change includes simplifying remote SSRC change handling
> via received RTP and RTCP packets.
>
> Bug: webrtc:11989
> Change-Id: Ia3ee62c1191baaedc67e033ea3c661d8c9301abc
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/199060
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Tim Na <natim@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32954}
Bug: webrtc:11989
Change-Id: I88620a9f1c037b512821cac9d556905149666870
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201481
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32966}
- Added local and remote RTP statistics query API.
- Change includes simplifying remote SSRC change handling
via received RTP and RTCP packets.
Bug: webrtc:11989
Change-Id: Ia3ee62c1191baaedc67e033ea3c661d8c9301abc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/199060
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32954}
The "//rtc_base:rtc_base" build target has historically been one of the
biggest targets in the WebRTC build. Big targets are the main source of
circular dependencies and non-API types leakage.
This CL is a step forward into splitting "//rtc_base:rtc_base" into
smaller targets (as originally started in 2018).
The only non-automated changes are (like re-wiring the build system):
* The creation of //rtc_base/async_resolver.{h,cc} which allows to
break a circular dependency (is has been extracted from
//rtc_base/net_helpers.{h,cc}).
* The creation of //rtc_base/internal/default_socket_server.{h,cc} to
break another circular dependency.
Bug: webrtc:9987
Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32941}
This is just general cleanup.
The assumed behavior is late decoding, and this function is not used to make any decision (except in the deprecated jitter buffer).
Bug: webrtc:12271
Change-Id: Ifb48186d55903f068f25e44c5f73e7a724f6f456
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/200804
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32940}
It turned out that the negotiated rtp header extensions are not fully known in WebRtcVideoChannel::AddSendStream.
The cl also remove the unnecessary factory for creating VideoStreamEncoder.
Bug: webrtc:12000
Change-Id: If994c8deb69f3ce4212896d3ad757dac94c6e09f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198840
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32916}
This should allow us to remove some SDP parsing in Chromium.
Bug: webrtc:12215
Change-Id: Ib85593d1c9226b29f2ec18617f945c76eca3b2d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197806
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32840}
Adds ability to specify desired frame size separate from actual clip
resolution, as well as clip and desired fps.
This allows e.g. reading an HD clip but running benchmarks in VGA, and
to specify e.g. 60fps for the clip but 30for encoding where frame
dropping kicks in so that motion is actually correct rather than just
plaing the clip slowly.
Bug: webrtc:12229
Change-Id: I4ad4fcc335611a449dc2723ffafbec6731e89f55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195324
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32839}
Two audio channels going into the AudioSource::Sink can either be
down-mixed to mono or encoded as stereo. This change enables WebRTC
users (such as Chromium) to query the number of audio channels actually
encoded. That information can in turn be used to tailor the audio
processing to the number of channels actually encoded.
This change fixes webrtc:8133 from a WebRTC perspective and will be
followed up with the necessary Chromium changes.
Bug: webrtc:8133
Change-Id: I8e8a08292002919784c05a5aacb21707918809c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197426
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32836}
- This CL also affects some return type handling in Android Voip demo
app due to changes in return type handling.
Bug: webrtc:12193
Change-Id: Id76faf7c871476ed1f2d08fb587211ae234ae8d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196625
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32821}
Now that RtpVp9RefFinder sets an additional reference on the frame instead of marking it as inter_layer_predicted it is no longer used.
Bug: webrtc:12206
Change-Id: I10e0930336eafc32dc86feb2f690cb131e55be2c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196740
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32814}
After the refined filter has been determined to perform better than
the coarse filter, and the coefficients of the coarse filters are
overwritten by the ones from the refined filter, at least 100 ms have
to pass before the adaptation of the refined filter is allowed to speed
up due to good coarse filter performance.
This change solves the vicious circle described in webrtc:12265, where
the coarse and refined filters can diverge over time.
This feature can be disabled remotely via a kill-switch. When disabled
the AEC output is bit-exact to before the change.
Bug: webrtc:12265,chromium:1155477
Change-Id: Iacd6e325e987dd8a475bb3e8163fee714c65b20a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196501
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32801}
This will soon become a compile-time error. Fix class hierarchies that
wrap StrictMock in a NiceMock or vice-versa by removing redundant
wrappings and removing inheritance from Nice/StrictMock and fixing the
call sites as appropriate.
Bug: b/173702213
Change-Id: Ic90b1f270c180f7308f40e52e358a8f6a6baad86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196461
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32783}
As requested on bugs.webrtc.org/12096#c2, this CL adds a Chromium
metric OWNERS in order to always have their review when WebRTC's UMA
metrics are updated.
Bug: webrtc:12096
Change-Id: Icd9ab7dda5f7a4ba6ac078f667c1fd39f3314123
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191702
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32728}
CL that should have been part of CL 195541
Bug: webrtc:12238
Change-Id: I3ab7a7a5f0d0bfdbc00904a01444acda02d49e90
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195543
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32712}
Needed in order to return different codes for different failures
in initialization.
Sideswipe: Check TURN URL hostnames for illegal characters.
Bug: webrtc:12238
Change-Id: I1af3a37b9654b83b268304f7356049f9f3786b7a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195541
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32710}
Instead of flushing all packets, it makes sense to flush down to the target level instead. This CL also initiates a flush when the packet buffer is a multiple of the target level, instead of waiting until it is completely full.
Bug: webrtc:12201
Change-Id: I8775147624536824eb88752f6e8ffe57ec6199cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193941
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32701}
This CL does not aim at cleaning up simulcast/SVC configuration, just to make it possible to set the scalability mode for AV1. Implementing a codec agnostic SVC/simulcast API is a (big) project on its own.
Change-Id: Ia88df31eb1111713e5f8832e95c8db44f92887ca
BUG: webrtc:11607
Change-Id: Ia88df31eb1111713e5f8832e95c8db44f92887ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/192541
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32631}
This can be used to test ICE behavior.
Bug: chromium:1024965
Change-Id: Ie4ba9cd5c3cf3c2f71bab3637f925263dbc6296e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193701
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32625}
This will cause encoding of a STUN message with an over-long
byte string attribute to fail.
Bug: chromium:1144646
Change-Id: I265174577376ce01439835c03f2d46700842d211
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191322
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Justin Uberti <juberti@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32603}
Starting from Android N, mobile app may not be able to access
microphone while in background where it fails the call.
In order to mitigate the issue, delay the ADM initialization
as late as possible.
Bug: webrtc:12120
Change-Id: I0fbf0300299b6c53413dfaaf88f748edc0a06bc1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191100
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32598}
And use it in a few places that were using RTC_CHECK(false) or FATAL()
to do the exact same job. There should be no change in behavior.
Bug: none
Change-Id: I36d5e6bcf35fd41534e08a8c879fa0811b4f1967
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191963
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32567}
- Introduced IngressStatistics to cover total_duration which
comes from AudioLevel.
Bug: webrtc:11989
Change-Id: Iba52d3722b5fe6286b048ab5690e32a4f75e972a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190940
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32538}
Avoid excessive echo suppression in frequencies above 2 kHz when
there is a dominant nearend. Calls with clock drift will not be affected
by this change as they tend to have less accurate linear filters.
Bug: webrtc:11985
Change-Id: Iddc628da5e2ba572c1b47acd87dd3be35260dca1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188580
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32533}
Calculate quality metrics for dropped frames by comparing original
frame against last decoded one.
This feature makes comparison of encoders which do/don't drop frames
more fair.
The feature is controlled by analyze_quality_of_dropped_frames flag
and is disabled by default.
Bug: none
Change-Id: Ifab8df92d0b76e743ff3193c05d7c8dbd14921c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190660
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32518}
In the old jitter buffer the two VCMVideoProtection modes |kProtectionNone| and |kProtectionFEC| could be set on the jitter buffer for it to not wait for NACK and instead generate incomplete frames. This has not been possible for a long time.
Bug: webrtc:9378, webrtc:7408
Change-Id: I0a2d3ec34d721126c1128306d5fad88314f8d59f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190680
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32513}
"warning: control reaches end of non-void function [-Wreturn-type]"
Reported by gcc (8.3)
In all the reported cases, the end of function is never actually
reached. Add RTC_CHECK(false) to ensure the compiler is aware that
this path is a dead-end.
Bug: webrtc:12008
Change-Id: I7f816fde3d1897ed2774057c7e05da66e1895e60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189784
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Fabien VALLÉE <fabien.vallee@netgem.com>
Cr-Commit-Position: refs/heads/master@{#32503}
This feature is active if and only if the RTP header extension
playout-delay is used with min playout delay=0 and max playout delay>0.
In this case, a maximum composition delay will be calculated and attached
to the video frame as a signal to use the low-latency renderer algorithm,
which is landed in a separate CL in Chromium.
The maximum composition delay is specified in number of frames and is
calculated based on the max playout delay.
The feature can be completetly disabled by specifying the field trial
WebRTC-LowLatencyRenderer/enabled:false/
Bug: chromium:1138888
Change-Id: I05f461982d0632bd6e09e5d7ec1a8985dccdc61b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190141
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32493}
This allows making more members (including IsUnifiedPlan) const in a future CL.
Also revises the test for ReportUsageHistogram to use a configuration member
variable rather than a hook function in PeerConnectionFactory.
Bug: webrtc:12079
Change-Id: I6f1af7d6164c8a0d8466f76378a925d72d57d685
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190280
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32485}
This is a reland of 239f92ecf7
Original change's description:
> introduce an unsupported content description type
>
> This carries around unsupported content descriptions
> (i.e. things where webrtc does not understand the media type
> or protocol) in a special data type so that a rejected content or
> mediasection is added to the answer SDP.
>
> BUG=webrtc:3513
>
> Change-Id: Ifc4168eae11e899f2504649de5e1eecb6801a9fb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179082
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> Cr-Commit-Position: refs/heads/master@{#32410}
Bug: webrtc:3513
Change-Id: I48e338100f829f1df5b8165217c89b5ef860fe79
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188820
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32457}
This reverts commit 239f92ecf7.
Reason for revert: Breaks downstream projects.
Original change's description:
> introduce an unsupported content description type
>
> This carries around unsupported content descriptions
> (i.e. things where webrtc does not understand the media type
> or protocol) in a special data type so that a rejected content or
> mediasection is added to the answer SDP.
>
> BUG=webrtc:3513
>
> Change-Id: Ifc4168eae11e899f2504649de5e1eecb6801a9fb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179082
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> Cr-Commit-Position: refs/heads/master@{#32410}
TBR=kthelgason@webrtc.org,hta@webrtc.org,philipp.hancke@googlemail.com
Change-Id: I055fe001fe2757d79be7c304eccc43a8e3104f69
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:3513
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188581
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32411}
This carries around unsupported content descriptions
(i.e. things where webrtc does not understand the media type
or protocol) in a special data type so that a rejected content or
mediasection is added to the answer SDP.
BUG=webrtc:3513
Change-Id: Ifc4168eae11e899f2504649de5e1eecb6801a9fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179082
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#32410}
This CL also puts the arguments in a struct to allow for easier future additions.
Bug: webrtc:11005
Change-Id: I47bf664e7106b724eb1fc42299c42bbf022393ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188385
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32409}
This CL also introduces a helper function to perform the config conversion,
which eliminates duplicate code.
Bug: webrtc:10337
Change-Id: I162288f90ebac8f2f345356ec25f0805df698c67
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188121
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32399}
No changes after the fix in the downstream project.
Bug: chromium:1132299
Change-Id: I7cb490d9e8ead7aea758f9542a84268fd4160016
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187497
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32390}
This sincludes the SimulcastEncoderAdapter and the
VideoEncoderSoftwareFallbackWrapper. This avoids converting
the frame when that is not needed.
Bug: webrtc:11976
Change-Id: I686725ecfb79c3b8d87d587a907da1602483bfe8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187343
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#32389}
This file is being accessed from Chrome. Moving it lessens the
dependency of Chrome on files in the pc/ directory, and allows
easier refactoring of pc/.
Bug: webrtc:11967
Change-Id: Iccd568f84e9cf4086e37c58db1b4cba6c376f413
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187489
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32378}
(Reland with no changes after the fix to the downstream project)
This can be overriden for kNative frame types to perform scaling efficiently.
Default implementations for existing buffer types require actual
buffer implementation, thus this CL also merges "video_frame"
with "video_frame_I420" build targets.
Originally Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186303
(Landing with TBR as it's unchaged reland of already approved CL)
TBR=nisse@webrtc.org,sakal@webrtc.org
Bug: webrtc:11976, chromium:1132299
Change-Id: Ia23f7d3e474bd9cdc177104cc5c6d772f04b210f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187345
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32362}
Adds a field to EncoderInfo called preferred_pixel_formats which a
software encoder populates with the pixel formats it supports. When a
kNative frame is received for encoding, the VideoStreamEncoder will
first try to get a frame that is accessible by the software encoder in
that pixel format from the kNative frame. If this fails it will fallback
to converting the frame using ToI420.
This minimizes the number of conversions made in the case that the
encoder supports the pixel format of the native buffer or where
conversion can be accelerated. For example, in Chromium, the capturer can
emit an NV12 frame, which can be consumed by libvpx which supports NV12.
Testing: Tested in Chrome with media::VideoFrame adapters.
Bug: webrtc:11977
Change-Id: I9becc4100136b0c0128f4fa06dedf9ee4dc62f37
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187121
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#32353}
Chrome and other platforms will need access to this class when sending
biplanar buffers to webrtc.
Bug: chromium:1134165
Change-Id: Ia787ab02cb9f302670d6a81e8d4963e7d8fca468
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187181
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#32348}
VideoBitrateAllocation is instead reported through the EncoderSink.
Enable VideoBitrateAllocation reporting from WebRtcVideoChannel::AddSendStream in preparation for
using the extension RtpVideoLayersAllocationExtension instead of RTCP XR.
Bug: webrtc:12000
Change-Id: I5ea8e4f237a1c4e84a89cbfd97ac4353d4c2984f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186940
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32347}
Original description
Move reporting of target bitrate to just after the encoder has been
updated. Originall submitted as refs/heads/master@{#32275}
Patch 1 contains the original cl
,patch 2 the fix to send rtcp even if BWE does not change.
Bug: webrtc:12000
Change-Id: I16766e08229fe1f6f65f449e0e074bed03338693
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186948
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32340}
This reverts commit 39a31afb77.
Reason for revert: Will cause RTCP Target bitrate messages to not be sent unless BWE changes.
Original change's description:
> Refactor reporting of VideoBitrateAllocation
>
> Move reporting of target bitrate to just after the encoder has been
> updated.
>
> Bug: webrtc:12000
> Change-Id: I3e7c5bd44c2f64e5f7e32d6451861b80e0b779ca
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186041
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32275}
TBR=sprang@webrtc.org,perkj@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:12000
Change-Id: Icf21e6ae28dc17c61b9243c037ffef9b623894ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186945
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32337}
During muted state NetEq shortcircuits a large part of the internals to
quickly return a buffer filled with zeros. It can be beneficial for the
controller to be aware that it is in muted state.
Bug: webrtc:11005
Change-Id: I5fe24b4a3704d953cbd68b5a24bbb7ef58b30be0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186760
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32330}
The extension is suggested to be used for signaling per target bitrate, resolution
and frame rate to a SFU to allow a SFU to know what video layers a client is currently targeting.
It is hoped to replace the current Target bitrate RTCP XR message currently used only for screen share.
Bug: webrtc:12000
Change-Id: Id7b55e7ddaf6304e31839fd0482b096e1dbe8925
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185980
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32313}
This needs to be done still for kNative frames, but all other frame types
can be passed in.
I have checked all VideoEncoder implementations in Chromium and confirmed they either convert the frame to their preferred pixel format, or just
forward the frame to a delegate encoder.
Tested:
- video_loopback with NV12 generated frames for VP9, the only
codec supporting NV12, as well as VP8 which only accepts I420 frames.
- internal_tests tryrun
Bug: webrtc:11976,webrtc:11635
Change-Id: If39a815fb0c5636fceb1040c8946c3db2fb350a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185803
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32306}
This is a reland of 2978abb88c
Original change's description:
> Reduce the amount of howling reduction in AEC3
>
> This CL backs off the howling protection functionality in AEC3.
> The effect is increased transparency in some cases. No negative effects
> have been identified in the hands-on testing.
>
>
> A kill-switch is added that can be used to turn off the functionality.
>
> Bug: b/150764764
> Change-Id: I604c569c76f911799556a60bc8fd2fb43bbfe196
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186082
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32258}
Bug: b/150764764,chromium:1134939
Change-Id: I5eea60b35e6d09003ec2fee3865513df8bdd5823
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186260
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32304}
This adds a config parameter for reverb modelling in nonlinear mode.
The killswitch is preserved, and will have the same effect before and after this CL.
Tested: Bitexact on a large number of aecdumps both with and without WebRTC-Aec3NonlinearModeReverbKillSwitch.
Bug: webrtc:8671
Change-Id: I7b2fa92307af5316dbe895392367abb8d15f06bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183541
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32298}
API to injecting a heavy audio processing operation into WebRTC audio capture pipeline
Bug: webrtc:12003
Change-Id: I9f6f58f468bd84efd0a9d53d703db6229a03959e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165788
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32291}
This reverts commit 18515e2793.
Reason for revert: Continuation of previous revert https://webrtc-review.googlesource.com/c/src/+/186304
Original change's description:
> Reland "Reduce the amount of howling reduction in AEC3"
>
> This is a reland of 2978abb88c
>
> Original change's description:
> > Reduce the amount of howling reduction in AEC3
> >
> > This CL backs off the howling protection functionality in AEC3.
> > The effect is increased transparency in some cases. No negative effects
> > have been identified in the hands-on testing.
> >
> >
> > A kill-switch is added that can be used to turn off the functionality.
> >
> > Bug: b/150764764
> > Change-Id: I604c569c76f911799556a60bc8fd2fb43bbfe196
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186082
> > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> > Commit-Queue: Per Åhgren <peah@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#32258}
>
> Bug: b/150764764
> Change-Id: I277f24eb3288ad0307e7463bad9aea6436cfe879
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186263
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32274}
TBR=gustaf@webrtc.org,peah@webrtc.org
Change-Id: I0596855e3bf910878aebcee90e03345d5db1105d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: b/150764764
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186269
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32276}
Move reporting of target bitrate to just after the encoder has been
updated.
Bug: webrtc:12000
Change-Id: I3e7c5bd44c2f64e5f7e32d6451861b80e0b779ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186041
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32275}
This is a reland of 2978abb88c
Original change's description:
> Reduce the amount of howling reduction in AEC3
>
> This CL backs off the howling protection functionality in AEC3.
> The effect is increased transparency in some cases. No negative effects
> have been identified in the hands-on testing.
>
>
> A kill-switch is added that can be used to turn off the functionality.
>
> Bug: b/150764764
> Change-Id: I604c569c76f911799556a60bc8fd2fb43bbfe196
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186082
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32258}
Bug: b/150764764
Change-Id: I277f24eb3288ad0307e7463bad9aea6436cfe879
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186263
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32274}
This reverts commit 2978abb88c.
Reason for revert: Breaks ApmTest.Process.
After trying to enable AVX2, we increased the amount of error
we tolerate (kFloatNear) and this CL introduced a regression which
makes the test fail after we reverted the enabling of AVX2 (restoring
the old tolerance).
With this CL:
../../modules/audio_processing/audio_processing_unittest.cc:1779: Failure
The difference between test->rms_dbfs_average() and rms_dbfs_average is 0.00142669677734375, which exceeds kFloatNear, where
test->rms_dbfs_average() evaluates to 52.907142639160156,
rms_dbfs_average evaluates to 52.905715942382812, and
kFloatNear evaluates to 0.00050000000000000001.
[ FAILED ] ApmTest.Process (5347 ms)
[----------] 1 test from ApmTest (5348 ms total)
[----------] Global test environment tear-down
[==========] 1 test from 1 test suite ran. (5350 ms total)
[ PASSED ] 0 tests.
[ FAILED ] 1 test, listed below:
[ FAILED ] ApmTest.Process
After reverting it:
[ OK ] ApmTest.Process (5345 ms)
[----------] 1 test from ApmTest (5347 ms total)
[----------] Global test environment tear-down
[==========] 1 test from 1 test suite ran. (5350 ms total)
[ PASSED ] 1 test.
Original change's description:
> Reduce the amount of howling reduction in AEC3
>
> This CL backs off the howling protection functionality in AEC3.
> The effect is increased transparency in some cases. No negative effects
> have been identified in the hands-on testing.
>
>
> A kill-switch is added that can be used to turn off the functionality.
>
> Bug: b/150764764
> Change-Id: I604c569c76f911799556a60bc8fd2fb43bbfe196
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186082
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32258}
TBR=gustaf@webrtc.org,peah@webrtc.org
Change-Id: I4723c5cd66e3046851089157ec586afab55c5ce8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: b/150764764
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186122
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32261}
This CL backs off the howling protection functionality in AEC3.
The effect is increased transparency in some cases. No negative effects
have been identified in the hands-on testing.
A kill-switch is added that can be used to turn off the functionality.
Bug: b/150764764
Change-Id: I604c569c76f911799556a60bc8fd2fb43bbfe196
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186082
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32258}
The former was unused, the latter is replaced with the explicit C++11
deletions. The related RTC_DISALLOW_COPY_AND_ASSIGN is left for now,
it is used in a lot more places.
Bug: None
Change-Id: I49503e7f2b9ff43c6285f8695833479bbc18c380
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185500
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32224}
NV12 frames can be encoded by libvpx now, and this change allows for
encoding of them with VP9.
VP9 encode/decode tests now run with NV12 as well as I420.
Manually tested using video loopback with VP9 and NV12 generated frames.
out/Default/video_loopback.app/Contents/MacOS/video_loopback --clip=GeneratorNV12 --codec="VP9"
Bug: webrtc:11635, webrtc:11974
Change-Id: Ifc5cbf77d2a27821cd5560c253d5d447c7a7cf53
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185123
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#32220}
