This class receives data about video frames from ReceiveStatisticsProxy,
calculates spatial and temporal quality metrics and outputs them to UMA
stats. It is all done in a separate class because it will be further
extended to calculate aggregated quality metrics in the future.
Bug: webrtc:9295
Change-Id: Ie36db83e10c0e8da0b9baa392651cb9a67a54a80
Reviewed-on: https://webrtc-review.googlesource.com/78220
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23387}
This CL replaces components in the congestion controller module
that are identical to equivalent components in the rtp and goog_cc
subfolder. Some redundant components are left as they were not
trivial to replace.
Bug: webrtc:8415
Change-Id: I86a1f164d7b100b8ec8ba7dbc1c9bda2128a4f37
Reviewed-on: https://webrtc-review.googlesource.com/78521
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23384}
After last update in a chromium repo ffmpeg support for MSVC was broken.
So for now we will freeze rolling of ffmpeg and continue it after
we'll restore of MSVC or we'll find a way around to keep MSVC support
in the WebRTC.
Change-Id: Ie7de7e6d367946f3ad77a81d6121dd704a56ca24
Bug: webrtc:9213
Reviewed-on: https://webrtc-review.googlesource.com/78402
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23379}
Code using the macro change to a plain declaration+init of a local
variable.
Also delete includes of <stdint.h> and <stddef.h> from basictypes.h.
Bug: webrtc:6853
Change-Id: I5ffceb449c1bf8f5badb595d5a343a47b0c6deae
Reviewed-on: https://webrtc-review.googlesource.com/78460
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23377}
Instead of making multiple calls to the std::stringstream << operator,
collect all the arguments and make a single printf-like variadic call
under the hood.
Besides reducing our reliance on iostreams, this makes each RTC_LOG_*
call site smaller; in aggregate, this reduces the size of
libjingle_peerconnection_so.so by 28-32 kB.
A quick benchmark indicates that this change makes log statements
a few percent slower.
Bug: webrtc:8982, webrtc:9185
Change-Id: I3137a4dd8ac510e8d910acccb0c97ce4fffb61c9
Reviewed-on: https://webrtc-review.googlesource.com/75440
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23375}
This reverts commit 3409cfa378e75c0c08d900e0848147929249a62b.
Reason for revert: Broke WebRtcBrowserTest.RunsAudioVideoWebRTCCallInTwoTabsH264 on Windows 7/10 bots
Original change's description:
> Start supporting H264 packetization mode 0.
>
> The work was already done to support it, but it wasn't being negotiated
> in SDP.
>
> This means we'll now see 8 H264 payload types instead of 4; one for each
> combination of BP/CBP profiles, packetization modes 0/1, and RTX/non-RTX.
> This could be problematic in the future, since we're starting to run
> out of dynamic payload types (using 25 of 32).
>
> Bug: chromium:600254
> Change-Id: Ief2340db77c796f12980445b547b87e939170fae
> Reviewed-on: https://webrtc-review.googlesource.com/77264
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23372}
TBR=deadbeef@webrtc.org,magjed@webrtc.org,sprang@webrtc.org
Change-Id: I2f2a2b4ca20ba883764cd5265911e1453d3df66e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:600254
Reviewed-on: https://webrtc-review.googlesource.com/78398
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23374}
The work was already done to support it, but it wasn't being negotiated
in SDP.
This means we'll now see 8 H264 payload types instead of 4; one for each
combination of BP/CBP profiles, packetization modes 0/1, and RTX/non-RTX.
This could be problematic in the future, since we're starting to run
out of dynamic payload types (using 25 of 32).
Bug: chromium:600254
Change-Id: Ief2340db77c796f12980445b547b87e939170fae
Reviewed-on: https://webrtc-review.googlesource.com/77264
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23372}
This avoid sending more padding than required for the current target
constraints.
Bug: webrt:8415
Change-Id: I3a668990f026414ab78f8406248cde18b81123cc
Reviewed-on: https://webrtc-review.googlesource.com/77763
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23364}
The BBR controller did not properly report updates to congestion
windows. This was due to a check to avoid the overhead of callbacks.
In the current design without callbacks in the controller, the check can
be removed. If helpful for performance, it should live outside of the
controller.
Bug: webrtc:8415
Change-Id: Idf6d6e76fe6d0450841e706019110307e559c11d
Reviewed-on: https://webrtc-review.googlesource.com/78181
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23363}
The network control tester did not handle congestion windows correctly.
Time passed when no packets were sent were not counted. This hindered
the buffer delays from decreasing in congested mode.
Bug: webrtc:8415
Change-Id: Id46116c6125eb5a50caa5766a3cc7291404ff920
Reviewed-on: https://webrtc-review.googlesource.com/77761
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23361}
This limits the SDP to 16KB, which sounds enough.
Bug: chromium:813328
Change-Id: I58c7b3e073108fd7b3495e8182b5c632e9619fe7
Reviewed-on: https://webrtc-review.googlesource.com/78280
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23360}
This prepares for making the BBR implementation more identical to the
implementation in Quic, this is to ensure that results are comparable.
Bug: webrtc:8415
Change-Id: I6b182246c988dd4a95681c063dcaa779088d0e99
Reviewed-on: https://webrtc-review.googlesource.com/76481
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23356}
This changeset adds some additional tests to validate that the SSLAdapter
behaves as intended when used with a Custom SSL Certificate Verifier. It
validates the following scenarios:
1. Handshake succeeds on TLS handshakes if certificate verifier returns true.
2. Handshake fails on TLS handshakes if certificate verifier returns false.
3. Handshake succeeds on DTLS handshakes if certificate verifier returns true.
4. Handshake fails on DTLS handshakes if certificate verifier returns false.
5. Handshake succeeds on TLS transfers if certificate verifier returns true.
6. Handshake succeeds on DTLS transfers if certificate verifier returns true.
Bug: webrtc:9258
Change-Id: I48b72c9762a7023ece12d882ac4a05d9881bf9e6
Reviewed-on: https://webrtc-review.googlesource.com/75720
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23355}
End to end test for media sent over a TCP TURN server with both clients in relay
This test validates that media can be sent between two clients who are set up
to relay information with the server configured to use TCP instead of UDP.
Bug: webrtc:7668
Change-Id: I3efd04048589c144494f90f2cdf3df5f9f80300e
Reviewed-on: https://webrtc-review.googlesource.com/76507
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23354}
The Quic BBR implementation uses packet sequence numbers to keep track
of the time slots used for calculation of send receive rates. To avoid
protocol dependence the port were initially written to use send times
instead.
As there are issues with running BBR in WebRTC, it makes sense to
use an identical implementation as in Quic to ensure that there
aren't implementation issues causing bad behavior. This requires
providing sequence numbers.
This prepares for making the BBR implementation more identical to the
implementation in Quic, this is to ensure that results are comparable.
Bug: webrtc:8415
Change-Id: I2cd96bc6ffb88042bb2b91421bfe6cbf7c1ff8ac
Reviewed-on: https://webrtc-review.googlesource.com/76583
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23353}
Deletes the ALIGNP and RTC_ALIGNED_P macros from basictypes.h.
ALIGNP was used by MemoryStream, supposedly to make it more efficient.
If it really provided an efficiency improvement is unclear, and in any
case, MemoryStream is used for tests only, and doesn't need high
performance.
Bug: webrtc:6853
Change-Id: If835e881e3857dcc22c7a544491b92829a81d1b3
Reviewed-on: https://webrtc-review.googlesource.com/78021
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23350}
Unfortunately, pthread_condattr_setclock is lacking in the
versions of MacOS and iOS we support, and we have to stay
with gettimeofday on those platforms.
Bug: webrtc:9269
Change-Id: I8554e56496cc7b6948cb9b8a4c0bcf886c3544be
Reviewed-on: https://webrtc-review.googlesource.com/77122
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23349}
Reland of CL https://webrtc-review.googlesource.com/c/src/+/77421
Copied description:
--
Add presubmit check for changes in 3pp
Presubmit check will test will new changes be overriden by autoroll
or not. In more details presubmit will check:
1. Each dependency in third_party have to be specified in one of:
a. THIRD_PARTY_CHROMIUM_DEPS.json
b. THIRD_PARTY_WEBRTC_DEPS.json
2. Each dependency not specified in both files from #1
3. Changes won't be overriden by chromium third_party deps autoroll:
a. Changes were made in WebRTC owned dependency
b. Changes were addition of new Chromium owned dependency
--
Also if commit message contains tag NO_AUTOIMPORT_DEPS_CHECK equal
to True, than changes in chromium specific deps will be permitted.
It is required for autoroller to be able to commit its changes and
not to fail on presubmit check.
Bug: webrtc:8366
Change-Id: I545a4778445855cf3db7cf257ca0cb63753aac06
Reviewed-on: https://webrtc-review.googlesource.com/78042
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23348}
This CL implements a fix behind a field trial for a NetEq issue. NetEq restarts audio too quickly after a buffer underrun, which can quickly lead to another underrun in some circumstances. The fix changes NetEq's behavior to wait with restarting playback until sufficient audio is buffered.
Bug: webrtc:9289
Change-Id: I5968c9478ce8d84caf77f00b8d0a39156b47fc8d
Reviewed-on: https://webrtc-review.googlesource.com/77423
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23347}
These files are required when implementing tests based on the test fixture,
and should be exposed as part of the test api.
This CL also removes a usage of stringstream and fixes some chromium-style
lint issues.
Bug: webrtc:8982, webrtc:163
Change-Id: I132aea0da79a79587887f21897236fc9802b7574
Reviewed-on: https://webrtc-review.googlesource.com/74586
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23346}
NetEq tapers down the audio produced through loss concealment when the
expansion has been going on for some time. When the audio packets starts
coming in again, there is a ramp-up that happens. This ramp-up could
before this change extend over more than one 10 ms block, which made
keeping track of the scaling factor necessary. With this change, we make
this ramp-up quicker in the rare cases when it lasted more than 10 ms,
so that it always ramps up to 100% within one block. This way, we can
remove the mute_factor_array.
This change breaks bit-exactness, but careful listening could not reveal
an audible difference.
This change is a part of a larger refactoring of NetEq's PLC code.
Bug: webrtc:9180
Change-Id: I4c513ce3ed8d66f9beec2abfb1f0c7ffaac7a21e
Reviewed-on: https://webrtc-review.googlesource.com/77180
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23342}
Force auto roll of chromium third_party into WebRTC to fail if any one
of required chromium-specific dependencies is missing in chromium
third_party repo of after checking it into WebRTC repo.
Also try to fix some flakes in autoroller.
Bug: webrtc:8366
Change-Id: I781cd4d4a4a230fb126cc32d8147310f70ab8b91
Reviewed-on: https://webrtc-review.googlesource.com/77722
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23340}
This CL fixes the rounding of the estimated average call skew. Before it
was rounded down (toward INT_MIN). Now it is rounded to the nearest integer.
This avoids unnecessary fluctuations of the estimated call skew (and
unnecessary resets).
Bug: webrtc:9283,chromium:888042
Change-Id: Id5b3c593f812f5f9fd3dcdafb7e388a6ef1ac153
Reviewed-on: https://webrtc-review.googlesource.com/77684
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23338}
