Previously, windows captured by WgcCapturerWin may have been occluded
by other windows with no functionality to bring them to the top of the
z order. This change implements FocusOnSelectedSource for
WgcCapturerWin to afford consumers this functionality, and match the
experience offered by current capturers.
Bug: webrtc:12664
Change-Id: I8bc067ade90fba0be66a9be57d3429ba54ba1ae0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215241
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Austin Orion <auorion@microsoft.com>
Cr-Commit-Position: refs/heads/master@{#33809}
Fix simulcast svc controller to reuse dropped frame configuration,
same as full svc and k-svc controllers do.
This fuzzer reminded the issue was still there.
This is a reland of https://webrtc-review.googlesource.com/c/src/+/212281
Bug: webrtc:11999
Change-Id: Id3b2cd6c7e0923adfffb4e04c35ed2d6faca6704
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215921
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33802}
Logic for throttling how often REMB messages are sent is added to ReceiveSideCongestionController as well as a new method SetMaxDesiredReceiveBitrate. These are based on the logic in PacketRouter. The logic for throttling REMB and setting the max REMB will be removed from PacketRouter in a follow up cl.
The purpose is to eventually decouple PacketRouter from sending RTCP messages when RtcpTransceiver is used.
Bug: webrtc:12693
Change-Id: I9fb5cbcd14bb17d977e76d329a906fc0a9abc276
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215685
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33801}
The change introduces support for detachable PlatformThreads, for which
the Stop() call doesn't wait until the thread has finished executing.
The change also introduces rtc::ThreadAttributes that carries priority
and detachability thread attributes. It additionally refactors all
known use to use the new semantics.
Bug: b:181572711, webrtc:12659
Change-Id: Id96e87c2a0dafabc8047767d241fd5da4505d14c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214704
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33796}
- Received FEC packets are purged from the queue if:
1. All media packets protected by the FEC packet are received.
2. All media packets protected by the FEC packet are recovered.
3. Newer FEC packet(s) with sequence number '0x3fff' larger than an old FEC packet is received.
- When FEC packets get separated from their protected media packets by more than 48, none of the first conditions ever delete that FEC packet, no matter how old/ irrelevant it gets.
- Under specific circumstances, the new FEC packet (condition 3) is not received before the media sequence number space wraps around, and incorrectly activates the old FEC packet, resulting in FEC decode for the wrong packet.
- This change purges such old FEC packets in time before the media sequence numbers wrap around.
Bug: webrtc:12656
Change-Id: I6ddf5382638c8c7e9a65724b2544dfbbc4803342
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215100
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33795}
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}
This CL adds a new property to the DesktopFrame interface to indicate
that the capturer supports cursor capture and the frame may contain
an image of the cursor (if the cursor was over the window or screen
being captured). This allows the DesktopAndCursorComposer to avoid
compositing another image of the cursor on the frame.
This is preferred because natively capturing the cursor will likely
be more efficient, and for WGC the API to disable cursor capture
is only availabe on later versions of Win10, reducing the number
of users that could use it.
Bug: webrtc:12654
Change-Id: I992804ff2a65eb423fb8ecc66e066408dc05e849
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215341
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Austin Orion <auorion@microsoft.com>
Cr-Commit-Position: refs/heads/master@{#33780}
This changes modifies EnumerateCapturableWindows to accept an optional
parameter consisting of extended window styles that will prevent windows
with the specified styles from being returned. This allows us to filter
out windows with the WS_EX_TOOLWINDOW style for the WgcCapturerWin,
which does not support capture of such windows.
Bug: webrtc:12679
Change-Id: Id9ac28afd331ba20fcb7f9e7be54ea5eee2e022e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215161
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Austin Orion <auorion@microsoft.com>
Cr-Commit-Position: refs/heads/master@{#33779}
Biggest change is a new helper class used to read data from the
bitstream and then pass the result to a function if reading was
successful. There's also helper to do if/else flow based on the read
values. This avoids a bunch of temporaries and in my view makes the
code esaier to read.
For example, this block:
uint32_t bit;
RETURN_FALSE_IF_ERROR(br->ReadBits(&bit, 1));
if (bit) {
RETURN_FALSE_IF_ERROR(br->ConsumeBits(7));
}
...is now written as:
RETURN_IF_FALSE(
br->IfNextBoolean([br] { return br->ConsumeBits(7); }));
In addition, we parse and put a few extra things in FrameInfo:
show_existing_frame, is_keyframe, and base_qp.
Bug: webrtc:12354
Change-Id: Ia0b707b223a1afe0a4521ce2b995437d41243c06
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215239
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33776}
A couple crashes have been reported in Chromium due to us dereferencing
|result.frame| which can be a nullptr.
This bug tracks the addition of new test cases which will help us
avoid issues like this in the future:
https://bugs.chromium.org/p/webrtc/issues/detail?id=12682
Bug: chromium:1199257
Change-Id: I720dd6ceb38938dc392f0924acf2cac287bfcffc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215340
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Austin Orion <auorion@microsoft.com>
Cr-Commit-Position: refs/heads/master@{#33746}
Like aom and openh264, VP9 can be disabled with the gn argument.
Bug: None
Change-Id: I7d67e3946afae0bb4cac8a7e591445604dda9ce1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215260
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33737}
- Bug fix: the desired initial gain quickly dropped to 0 dB hence
starting a call with a too low level
- New tuning to make AGC2 more robust to VAD mistakes
- Smarter max gain increase speed: to deal with an increased threshold
of adjacent speech frames, the gain applier temporarily allows a
faster gain increase to deal with a longer time spent waiting for
enough speech frames in a row to be observed
- Saturation protector isolated from `AdaptiveModeLevelEstimator` to
simplify the unit tests for the latter (non bit-exact change)
- AGC2 adaptive digital config: unnecessary params deprecated
- Code readability improvements
- Data dumps clean-up and better naming
Bug: webrtc:7494
Change-Id: I4e36059bdf2566cc2a7e1a7e95b7430ba9ae9844
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215140
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33736}
In RtpVideoStreamReceiver2 it can be protected by the `worker_task_checker_` instead.
Bug: webrtc:12579
Change-Id: I4f7d64f16172139eddc7a3e07d1dbbf338beaf2e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215224
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33734}
The current noise level estimator has a bug due to which the estimated
level decays to the lower bound in a few seconds when speech is observed.
Instead of fixing the current implementation, which is based on a
stationarity classifier, an alternative, lightweight, noise floor
estimator has been added and tuned for AGC2.
Tested on several AEC dumps including HW mute, music and fast talking.
Bug: webrtc:7494
Change-Id: Iae4cff9fc955a716878f830957e893cd5bc59446
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214133
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33733}
This reverts commit a743303211b89bbcf4cea438ee797bbbc7b59e80.
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}
Many things are omitted in this doc and it can definitely be improved,
but I hope it captures the most important parts.
Bug: webrtc:12568
Change-Id: I13097d633ca19cecc9dd43bdb777b0ca48f151dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215142
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33724}
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}
Erle Uncertainty changes the residual echo computation during saturated
echo. However, the case of saturated echo is already handled by the
residual echo estimator causing the ErleUncertainty to be a no-op.
The change has been tested for bit-exactness.
Bug: webrtc:8671
Change-Id: I779ba67f99f29d4475a0465d05da03d42d50e075
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215072
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33719}
As part of adding the new WgcCapturerWin implementation of the
DesktopCapturer interface, we should ensure that we can measure the
health and success of this new code. In order to quantify that, I've
added telemetry to measure the usage of each capturer implementation,
the time taken to capture a frame, and any errors that are encountered
in the new implementation.
I've also set the capturer id property of frames so that we can measure
error rates and performance of each implementation in Chromium as well.
This CL must be completed after this Chromium CL lands:
2806094: Add histograms to record new WebRTC DesktopCapturer telemetry | https://chromium-review.googlesource.com/c/chromium/src/+/2806094
Bug: webrtc:9273
Change-Id: I33b0a008568a4df4f95e705271badc3313872f17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214060
Commit-Queue: Austin Orion <auorion@microsoft.com>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#33716}
This prepares for ability to defer sequence number assignment to after
the pacing stage - a scenario where the RtpRtcp module rather than than
RTPSender class has responsibility for sequence numbering.
Bug: webrtc:11340
Change-Id: Ife88f60258b9b7cfd9dbd3326f02ac34da8f7603
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214967
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33702}
Following up on https://webrtc-review.googlesource.com/c/src/+/213000
This CL prevents scheduling work before TaskQueuePacedSender::EnsureStarted(),
making it necessary to function.
Bug: chromium:1152887
Change-Id: I848c9e6d6057a404626ad693b1f4dc7fba797a9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214320
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Etienne Pierre-Doray <etiennep@chromium.org>
Cr-Commit-Position: refs/heads/master@{#33695}
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}
VideoStreamEncoderTest: Remove unneeded set_timestamp_rtp in CreateFrame methods (the timestamp is set based on ntp_time_ms in VideoStreamEncoder::OnFrame).
Bug: none
Change-Id: I6b5531a9ac21cde5dac54df6de9b9d43261e90c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214488
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33683}
To maintain interoperability between different capturer implementations
this change updates WgcScreenSourceEnumerator to return a list of
device indices instead of a list of HMONITORs, and WgcScreenSource to
accept a device index as the input SourceId. WGC still requires an
HMONITOR to create the capture item, so this change also adds a utility
function GetHmonitorFromDeviceIndex to convert them, as well as new
tests to cover these changes.
Bug: webrtc:12663
Change-Id: Ic29faa0f023ebc26b4276cf29ef3d15d976e8615
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214600
Commit-Queue: Austin Orion <auorion@microsoft.com>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#33673}
This CL fixes a buffer copying issue introduced in this CL:
https://webrtc-review.googlesource.com/c/src/+/196485
In the BasicDesktopFrame::CopyOf function, the src and dst params
were swapped. For me this manifested as a missing cursor when using
Chrome Remote Desktop. I don't know of any other bugs this caused
but I have to assume it affects all callers of the function given
that the copy will never occur.
Bug: chromium:1197210
Change-Id: I076bffbad1d658b1c6f4b0dffea17d339c867bef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214840
Commit-Queue: Joe Downing <joedow@google.com>
Commit-Queue: Jamie Walch <jamiewalch@chromium.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#33672}
There's been reports of dropped frames that are not counted and
correctly reported by getStats().
If a HW decoder is used and the system is provoked by stressing
the system, I've been able to reproduce this problem. It turns out
that we've missed frames that are dropped because there is no
callback to the Decoded() function.
This CL restructures the code so that dropped frames are counted
even in cases where there's no corresponding callback for some frames.
Bug: webrtc:11229
Change-Id: I0216edba3733399c188649908d459ee86a9093d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214783
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33671}
It was recommened to me to move this define to the top level BUILD.gn
file to avoid potential issues with the define not being available
where we need it.
Bug: webrtc:9273
Change-Id: Id0e939a51d1e381f684a3ae970569a255f52a5bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214101
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Austin Orion <auorion@microsoft.com>
Cr-Commit-Position: refs/heads/master@{#33661}
There is only one NTP clock now.
Bug: webrtc:11327
Change-Id: I8c2808cf665f92bd251d68e32062beeffabb0f43
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214132
Commit-Queue: Paul Hallak <phallak@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33657}
Note: google3 needs to set this clock before we can start using it.
Bug: webrtc:11327
Change-Id: I0436c6633976afe208f28601fdfd50e0f6f54d6e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214480
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Paul Hallak <phallak@google.com>
Cr-Commit-Position: refs/heads/master@{#33653}
This helper method does not belong to the Clock class. Also, it's simple enough that it's not needed.
Bug: webrtc:11327
Change-Id: I95a33f08fd568b293b591171ecaf5e7aef8d413c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214123
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Paul Hallak <phallak@google.com>
Cr-Commit-Position: refs/heads/master@{#33652}
Done in preparation for the child CL which adds an alternative
implementation.
Bug: webrtc:7494
Change-Id: I4963376afc917eae434a0d0ccee18f21880eefe0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214125
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33646}
...though the big issue was probably that pending frames weren't being
culled properly in the case of frame dropping.
Bug: webrtc:12596
Change-Id: I9a03282b2a99087aa7c5650e57ce30fe0f0d3036
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214127
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33638}
Code style improvements done in preparation for a bug fix (TODO added)
which requires changes in the unit tests.
Note that one expected value in the unit tests has been adjusted since
the white noise generator is now instanced in each separate test and
therefore, even if the seed remained the same, the generated sequences
differ.
Bug: webrtc:7494
Change-Id: I497513b84f50b5c66cf6241a09946ce853eb1cd2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214122
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33636}
While debugging https://crbug.com/1195144 I found it useful to clarify
this log statement.
The log would say "When scaling [kNative], the image was unexpectedly
converted to [kI420]..." but not saying what it was trying to convert
it to. This CL adds: "... instead of [kNV12]."
Bug: chromium:1195144
Change-Id: I13e0040edf5d7d98d80ce674812f67dfb73be36e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214040
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33634}
This is a reland of 89cb65ed663a9000b9f7c90a78039bd85731e9ae
... and f28aade91dcc2cb8f590dc1379ac7ab5c1981909
... and 2072b87261a6505a88561bdeab3e7405d7038eaa
Reason for revert: Failing DuoGroupsMediaQualityTest due to missing
TaskQueuePacedSender::EnsureStarted() in google3.
Fix: This CL adds the logic behind TaskQueuePacedSender::EnsureStarted,
but initializes with |is_started| = true. Once the caller in google3 is
updated, |is_started| can be switched to false by default.
> Original change's description:
> Reason for revert: crashes due to uninitialized pacing_bitrate_
> crbug.com/1190547
> Apparently pacer() is sometimes being used before EnsureStarted()
> Fix: Instead of delaying first call to SetPacingRates(),
> this CL no-ops MaybeProcessPackets() until EnsureStarted()
> is called for the first time.
> Original change's description:
> > [Battery]: Delay start of TaskQueuePacedSender.
> >
> > To avoid unnecessary repeating tasks, TaskQueuePacedSender is started
> > only upon RtpTransportControllerSend::EnsureStarted().
> >
> > More specifically, the repeating task happens in
> > TaskQueuePacedSender::MaybeProcessPackets() every 500ms, using a self
> > task_queue_.PostDelayedTask().
> >
> > Bug: chromium:1152887
> > Change-Id: I72c96d2c4b491d5edb45a30b210b3797165cbf48
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208560
> > Commit-Queue: Etienne Pierre-Doray <etiennep@chromium.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33421}
>
> Bug: chromium:1152887
> Change-Id: I9aba4882a64bbee7d97ace9059dea8a24c144f93
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212880
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Etienne Pierre-Doray <etiennep@chromium.org>
> Cr-Commit-Position: refs/heads/master@{#33554}
Bug: chromium:1152887
Change-Id: Ie365562bd83aefdb2757a65e20a4cf3eece678b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213000
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Etienne Pierre-Doray <etiennep@chromium.org>
Cr-Commit-Position: refs/heads/master@{#33629}
libaom is compiled with REALTIME_ONLY option. Soon it will be impossible
to create encoder or request default config with usage other than
AOM_USAGE_REALTIME. Fixing the wrapper to use proper usage parameter
Bug: None
Change-Id: I862741a724e4a8524f22ae79700b3da6517dbfb2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214100
Commit-Queue: Fyodor Kyslov <kyslov@google.com>
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33624}
