which have shown that it is not easily possible to restrict
the pool size to 1 and combine this with max-bundle
BUG=webrtc:12383,chromium:1328218
Change-Id: I3a7ae4a263238c1b5faa079c3cbdaf84d1b40cbc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279141
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#38396}
Trigger probes next process intervals if the loss based current state is either increasing or decreasing. 0/ first probe at the loss based estimate. 1/ if increasing: allow further probing. 2/ if decreasing: not allow further probing.
Bug: webrtc:12707
Change-Id: I4e99edcbe4e2c315e8498ffb7fb2e589cdb4e666
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279041
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38395}
Libdrm is an essential library and should be available everywhere where needed. It also looks it's a dependency for Chromium already.
Bug: webrtc:13429
Change-Id: Id81497b4f29bbd80f7d94f57333aa533288c3538
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279023
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#38392}
This CL increases the test coverage for webrtc::SimualtedNetwork, adds
some more comments to the class and the interface it implements and
simplify the logic around capacity and delay management in the
simulated network.
More CLs will follow to continue the refactoring but this is the
ground work to make this more modular in the future.
Bug: webrtc:14525, b/243202138
Change-Id: Ib0408cf6e2c1cdceb71f8bec3202d2960c5b4d3c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278042
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38388}
Also call out the places where it happens explicitly - these are places
that need to be redesigned.
Bug: chromium:1177125
Change-Id: I3237d028dbb22380e8fbf7cedb03e965d1fcf2aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279022
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38384}
- Loss based bwe has 3 states: increasing (increasing when loss limited), decreasing (decreasing when loss limited), or delay based bwe (the same as delay based estimate).
- When bandwidth is loss limited and decreasing, and probe result is available, GetLossBasedResult = min(estimate, probe result).
- When bandwidth is loss limited and increasing, and the estimate is bounded by acked bitrate * a factor.
- When bandwidth is loss limited and probe result is available, use probe bitrate as the current estimate, and reset probe bitrate.
Bug: webrtc:12707
Change-Id: I53cb82aa16397941c0cfaf1035116f775bdce72b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277400
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38382}
Recent WebRTC stats spec changes have added restrictions on what stats
are available to JavaScript. This is done to reduce that fingerprinting
surface of WebRTC getStats. For example, stats exposing hardware
capabilities have requirements that must be met by the browser. See [1]
for more details.
This CL adds the types and the enumerations. Stats with these
restrictions should not be added until Chromium has implemented
filtering based on the stat type.
[1] https://w3c.github.io/webrtc-stats/#limiting-exposure-of-hardware-capabilities
Bug: webrtc:14546
Change-Id: I6dae5d4921c7a2bc828a4fc8f7d68e0c59f3be82
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279043
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38381}
The parameterized tests run in two versions.
1. Default with separate network and worker thread
2. The network thread is used as a worker thread. All packet sending and receiving is done on that thread.
Bug: webrtc:14502
Change-Id: Iba09295f6c1d7030d726f387f9cab8c2bf7c03d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278980
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38379}
This CL:
- makes it easier to understand the (nontrivial) metric interpretation
- corrects the computation of BufferDelay to use 0 for absent delay
- deletes metric MaxSkewShiftCount, unused since https://webrtc-review.googlesource.com/c/src/+/119701
- updates the unit test to directly test metric reporting
Corresponding update to histograms.xml:
https://crrev.com/c/3944909
Previous revert:
https://webrtc-review.googlesource.com/c/src/+/279040
This CL is identical to the original, except:
- the test is updated to spam fewer EXPECT_EQ failures on failure (EXPECT_EQs moved out of inner loop)
- the test not resets metrics (metrics::Reset()) at the beginning, like other histogram tests
Bug: webrtc:8671, chromium:1349051
Change-Id: Ie802e1f9d03a22ff7018f522a63b19e0b6eec2e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279046
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38376}
This is a speculative fix for the DCHECK at the top of
ScreenCapturerX11::CaptureScreen(). Whenever |selected_monitor_rect_|
changes, |queue_| should be reset, so that new frames are allocated
with the correct size. This CL adds a reset to UpdateMonitors() which
modifies |selected_monitor_rect_| and is called whenever an X11
configuration-change event is received (for example, when a monitor is
resized).
Bug: chromium:1372579
Change-Id: I9cc84a8b6990802f9d7dde05966ee17a80ddd48e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279065
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Lambros Lambrou <lambroslambrou@chromium.org>
Auto-Submit: Lambros Lambrou <lambroslambrou@chromium.org>
Cr-Commit-Position: refs/heads/main@{#38374}
- Set the initial input volume to that forced by startup min volume
since the latter is removed in a follow-up CL
- Remove unwanted expectations
Bug: webrtc:7494
Change-Id: I2df28f5bfaf4e592dfeae5e03b157268473cc822
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278784
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38370}
This is a reland of commit c4b0bde7f2daabc4e0667fb73d096d1cf0819226
which changes the name of the new method and adds a deprecated
backward compatible variant with the old name.
Original change's description:
> ice server parsing: return RTCError with more details
>
> surfacing those errors to the API (without specific details) instead of just the logging.
>
> BUG=webrtc:14539
>
> Change-Id: Id907ebeb08b96b0e4225a016a37a12d58889091b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278340
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Cr-Commit-Position: refs/heads/main@{#38356}
Bug: webrtc:14539
Change-Id: I0a5482e123f25867582d62101b31ed207b95ea1a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278962
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38364}
If the network thread and worker thread is the same, this log will spam.
Bug: webrtc:14502
Change-Id: Icb283f38fe6fbbca06ce911b9c0793148d459eef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278790
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38363}
This CL:
- makes it easier to understand the (nontrivial) metric interpretation
- corrects the computation of BufferDelay to use 0 for absent delay
- deletes metric MaxSkewShiftCount, unused since https://webrtc-review.googlesource.com/c/src/+/119701
- updates the unit test to directly test metric reporting
Corresponding update to histograms.xml:
https://crrev.com/c/3944909
Bug: webrtc:8671, chromium:1349051
Change-Id: If73b6fca4de7343bff2c53f72cedda458d36c599
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278782
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38362}
This reverts commit c4b0bde7f2daabc4e0667fb73d096d1cf0819226.
Reason for revert: Breaks downstream tests.
Basically, ParseIceServers() and other functions have changed
the return type, and this breaks tests at compile time.
Is it possible to reland with backwards compatible versions that return
the previous type? Then they can be removed afterwards.
Original change's description:
> ice server parsing: return RTCError with more details
>
> surfacing those errors to the API (without specific details) instead of just the logging.
>
> BUG=webrtc:14539
>
> Change-Id: Id907ebeb08b96b0e4225a016a37a12d58889091b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278340
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Cr-Commit-Position: refs/heads/main@{#38356}
Bug: webrtc:14539
Change-Id: I4df936ff865f87759936c5d0425478fe51051dc8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278960
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#38359}
surfacing those errors to the API (without specific details) instead of just the logging.
BUG=webrtc:14539
Change-Id: Id907ebeb08b96b0e4225a016a37a12d58889091b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278340
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#38356}
Only use the network thread for sending and receiving packets.
The one and only network thread is used as a worker thread in all
PeerConnections. Pacing when sending packets is done on the worker thread.
Bug: webrtc:14502
Change-Id: Ib373315688ae4d810ae1e4421101a859fca93b31
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278621
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38354}
RTPHeader is not exported, so the TransformableIncomingAudioFrame can't be mocked in chrome tests, using a getter instead.
Bug: chromium:1247260
Change-Id: I2af4e6a88b3f4772b3bb50ee0ae9d5c80fed3ae4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278785
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38352}
This is needed in order to use jint and make the header self contained.
Bug: b/251890128
Change-Id: Ie6c323113370a1d49f68c783137292e1c0be07d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278780
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38351}
