This change removes the old `ContributingSources` class. It has been replaced by the new `SourceTracker`.
Bug: webrtc:10793
Change-Id: Ibd481cf6584837c46b229b9fc2a071362f07d361
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147878
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28756}
optional<int> initial_bitrate_interval_ms: time interval since start of call
where fast adapt down is allowed.
optional<double> initial_bitrate_factor: try fast adapt down if bw estimate is
below initial bitrate * factor.
Bug: none
Change-Id: I63e1fdaac6556d8e9a961a42e11c925f9ecb9771
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147725
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28753}
This reverts commit 7f1c58938db72b1508e383d94a0e59dd70ff306e.
Reason for revert: this has been temporarily postponed.
Original change's description:
> Adding new top-level directory crypto/
>
> Adding the crypto root directory to WebRTC. The goal with this change is to
> centralize the management of crypto code into a single location.
>
> Currently we have cryptography code scattered across pc/ and rtc_base/
> which makes it difficult audit and maintain.
>
> By having a crypto/ directory we gain:
> 1. A clear first point of contact for auditing the cryptography in WebRTC.
> 2. Fine grain ownership to cryptography maintainers, we can include BoringSSL
> maintainers in this directory.
> 3. It improves maintanability of crypto code as we have improved modularization.
> It will not be deeply nested in all different parts of WebRTC.
> 4. Improved testability. We can cleanly build crypto libraries which plug into
> pc/ which we can more easily mock.
> 5. Enforce stricter rules. For example we may want to enforce ZeroOnFreeBuffer
> for all sensitive material. This is easier to enforce in a single directory.
>
> Bug: webrtc:9600
> Change-Id: I8e76332c7dcdac0a45a470ba2e930196e1ccf395
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125142
> Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27028}
TBR=steveanton@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,benwright@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9600
Change-Id: I3c99e733d53d76071179f0ff9ffdec965d20829d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147871
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28750}
We want to evaluate more data in order to make better choices in the
bitrate allocators.
In order to freely update the parameter list without
breaking the API many times for projects customizing them, we'll use a
struct instead.
Bug: webrtc:10126
Change-Id: I443f86781c5134950294cdd1e3197a47447cf973
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141418
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28748}
Old way to produce this histogram was based on RtcpStatisticsCallback
reporting sent RTCP messages, with some additional processing by the
ReportBlockStats class. After this cl, to grand average fraction loss
is computed by StreamStatistician, queried by VideoReceiveStream when
the stream is closed down, and passed to ReceiveStatisticsProxy which
produces histograms.
This is a preparation for deleting the RtcpStatisticsCallback from
ReceiveStatistics.
Bug: webrtc:10679
Change-Id: Ie37062c1ae590fd92d3bd0f94c510e135ab93e8d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147722
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28747}
This reverts commit 67008dfb366237469fe088a61b62c0cad852c024.
Reason for revert: Tests in the Chromium repo have been changed to accomodate this CL: https://chromium-review.googlesource.com/c/chromium/src/+/1728565
Original change's description:
> Revert "Replace the implementation of `GetContributingSources()` on the audio side."
>
> This reverts commit 8fa7151e4bbad40fec1f964fe0c003b8787bb78a.
>
> Reason for revert: Speculative revert to fix roll of webrtc into chrome. Right now tests related to RTCRtpReceiver failing and looks like it is main candidate, who can affect that behavior.
>
> Original change's description:
> > Replace the implementation of `GetContributingSources()` on the audio side.
> >
> > This change replaces the `ContributingSources`-implementation of `GetContributingSources()` and `GetSynchronizationSources()` on the audio side with the spec-compliant `SourceTracker`-implementation.
> >
> > The most noticeable impact is that the per-frame dictionaries are now updated when frames are delivered to the RTCRtpReceiver's MediaStreamTrack rather than when RTP packets are received on the network.
> >
> > This change is almost identical to the previous video side change at: https://webrtc-review.googlesource.com/c/src/+/143177
> >
> > Bug: webrtc:10545
> > Change-Id: Ife7f08ee8ca1346099b7466837a3756947085fc5
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144422
> > Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> > Commit-Queue: Chen Xing <chxg@google.com>
> > Cr-Commit-Position: refs/heads/master@{#28459}
>
> TBR=ossu@webrtc.org,chxg@google.com
>
> Change-Id: I5c631d4dcfb39601055ffce9b104f45eea871fd3
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10545
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144562
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28478}
TBR=ossu@webrtc.org,titovartem@webrtc.org,chxg@google.com
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10545
Change-Id: I609cca4f0ca4e1d31a156ba9eb44407518409f57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147865
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28746}
This reverts commit add7ef974ee2642a3b55a36ec80be50a615bc60a.
Reason for revert: Cause regression in pc_full_stack_tests.cc
Original change's description:
> Sanitize the codec list before sending it to the media engine
>
> The SDP can assign the same codec to two different payload types
> which gets represented as two separate codecs in the SDP structure.
> The media engine assumes that the client does not pass down
> duplicate codecs. This change adds logic to BaseChannel to filter
> out codecs of the same name with different payload types, picking
> the one which is listed first in the m= line.
>
> Bug: chromium:987598
> Change-Id: I6fa813db1769e572ff7c3f322dc9b1de39817ea2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147602
> Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28726}
TBR=steveanton@webrtc.org,amithi@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:987598
Change-Id: I4ffbfcd90c81c6c6c8ee8f872f7e217d8291c857
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147864
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28744}
1. Prevents deadlocks from AsyncInvoker destructor
2. Makes future state() calls are guaranteed to return the new state after
SetState() completes.
I am not sure if it is allowed to call FireOnChanged from non-signaling
threads so I will leave the post for now.
Bug: webrtc:10813
Change-Id: I5712a45f71431765898037867382397d537570a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147727
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28741}
This CL adds thread annotations and ensure that neither data races
nor deadlocks occur.
It prevents weird results and helps detecting other concurrency issues.
As a bonus, some dead code has been removed.
Bug: webrtc:10834
Change-Id: Ibd140db9e4dbf81b212044647e2d85bd18ef8d78
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147278
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#28737}
Some of the constants and structure definitions used are only available with
specific and recent versions of the windows SDK. This change allows this
to build with a toolchain targeting WINVER 0x0601 (Windows 7)
Bug: None
Change-Id: I3339f7c44c375fb7d583b78aa137f748c9776a07
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147440
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Paul Roberts <pacaro@google.com>
Cr-Commit-Position: refs/heads/master@{#28730}
The SDP can assign the same codec to two different payload types
which gets represented as two separate codecs in the SDP structure.
The media engine assumes that the client does not pass down
duplicate codecs. This change adds logic to BaseChannel to filter
out codecs of the same name with different payload types, picking
the one which is listed first in the m= line.
Bug: chromium:987598
Change-Id: I6fa813db1769e572ff7c3f322dc9b1de39817ea2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147602
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28726}
This simplifies creations of frame generator capturers in a reusable
way. It's modelled on the scenario VideoSendStreamConfig,
Bug: webrtc:10839
Change-Id: Ibe0709cd94521f78c6267eece533b048607d0994
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147272
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28722}
The ID of stats was based on the datachannel's "id"
attribute, but that could change - it was -1 before ID
allocation, and a number afterwards.
This CL changes the stats ID to depend on a monotonically
increasing counter for allocated datachannels.
Bug: webrtc:10842
Change-Id: I3e0c5dc07df8a7a502396de06bbedc9f676994a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147642
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28720}
This is a race that can happen if a nack arrives before media is
disabled, but the packet is not processed until after the disabling
is complete.
Bug: webrtc:10633, b/138636698
Change-Id: Ic90462b815163ab58c324e5cdb95c8d199c0b772
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147277
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28718}
Use the newly added total_decode_time_ms to get an accurate value
for the average decode time. The sparsely sampled decode_ms is
sensitive to the sampling instance.
Bug: chromium:980853
Change-Id: I9b63c8d1053fa95f74918807b83d1edb5cd726fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147268
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28716}
It's planned to be deprecated so it should not be required.
Bug: webrtc:9883
Change-Id: I7daa922786d3cbf6bca38e205f4f57773f3f8448
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147275
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28715}
This field trial was read in RTPSender, and the altered packet size
passed along to the pacer. Now, the pacer packet queue looks directly
at the packet instance, so it needs to be aware of the experiment flag
in order to make the right decision.
Bug: webrtc:10633, b/138582168
Change-Id: If1148f39c463e11ad49a659913465f131cf9b526
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147270
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28714}
This change includes windows owned by the primary captured window in the
captured frames if these conditions are met:
1) The owned window (e.g. dialog) overlaps the primary window (in whole
or part)
2) The primary window is otherwise eligible for the crop-from-screen
path (CroppingWindowCapturer is being used, and other conditions in
ShouldUseScreenCapturer are met)
In practice, this means that dialog windows / message boxes are captured
in many cases where they aren't today. This seems beneficial to some
scenarios (e.g. demonstrating / recording how to do something, or
requesting help with something, that involves dialogs).
This is a logical revert of a change for https://crbug.com/webrtc/8062 .
There's some commentary in the newer bug that attempts to make a case
for revisiting that change. (In summary: cases where a dialog would be
substantialy clipped / partial seem relatively uncommon and have
workarounds. Clipping may already occur for menus & tooltips. Clipping
seems less surprising than complete absence.)
Changing the GA_ROOT flag back to GA_ROOTOWNER is sufficient to restore
the older behavior. The removal of the EnumChildWindows call is just a
minor optimization (it was unnecessary/superfluous, since every child
window would match the GA_ROOT check; dialogs are owned root windows,
not child windows).
Removing condition (2) above (capturing dialogs & other related
overlapping windows when not using the crop-from-screen path) is tracked
by https://crbug.com/980864 .
Bug: webrtc:10767
Change-Id: If7b418365685a7b96dc93901ef9367844f9ee99e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147421
Commit-Queue: Jamie Walch <jamiewalch@chromium.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#28711}
We've observed a crash on Windows when the strings are empty, skipping the conversion seems reasonable in that case.
Bug: None
Change-Id: I3acf3060a88741fb750d7a0cc02e9422713c59cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147380
Commit-Queue: Noah Richards <noahric@chromium.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28709}
