See also b/175210069 for more context.
Bug: None
Change-Id: I06e9848028c0f11362db373af54b42cbc67aee77
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198780
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Henrik Grunell <henrikg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32874}
By enabling the field trial, REMB caps the output target bitrate, but
does not change any internal BWE state variables.
Bug: webrtc:12306
Change-Id: I43e9ac1d1b7dff292d7aa5800c01d874bc91aaff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197809
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32867}
removes the unused
x-google-port
x-google-max-message-size
which were probably sctp-related and variants of max-message-size
and sctp-port parameters. Similarly for the
protocol
streams
parameters which seem to have been fmtp parameters.
BUG=chromium:943975
Change-Id: I21b543f717d6e12fd737f91c7e159362488cc2e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198122
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32866}
While RTC_EXPORT is aware of component builds (selecting "default"
visibility only when WebRTC is built as a shared library),
RTC_OBJC_EXPORT (which predates RTC_EXPORT) was always marking symbols
as "default" visible.
This CL fixes the problem but on the other hand it will require
standalone builds of the WebRTC.framework to set the GN argument
`rtc_enable_symbol_export` to true.
No-Presubmit: True
Bug: chromium:1159620
Change-Id: I4a16f9bd3c1564140a5a30f03b3e77caed1df591
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198082
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Nico Weber <thakis@chromium.org>
Cr-Commit-Position: refs/heads/master@{#32856}
The calculation of the necessary number of shifts is not correct, leading to an overflow.
Bug: chromium:1158070
Change-Id: I6545e9da46debf33ce169c33d762915fe755d606
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197981
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32855}
This CL makes it possible to use a low-latency mode on Android O and later. This should help to reduce the audio latency. The feature is disabled by default and needs to be enabled when creating the audio device module.
Bug: webrtc:12284
Change-Id: Idf41146aa0bc1206e9a2e28e4101d85c3e4eaefc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196741
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32854}
for style consistency. This check is already done outside the method.
BUG=None
Change-Id: Ie1366fa57417258a301b02503ad76f304f4279a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198040
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#32853}
In order to land http://crrev.com/c/2593640 and make Chromium
backwards compatible with [1], a new empty header needs to be created.
[1] - https://webrtc-review.googlesource.com/c/src/+/196903
Bug: webrtc:9987
Change-Id: I48c4c47b56f9334ce329826c584c817919d59dad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197941
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32849}
Tests appear to not depend on any CallStats behaviour, and the usage
is not compatible with the threading requirements of the new
internal::CallStats class.
Bug: webrtc:11581
Change-Id: I8802a46842930eb58bd7609bcd8384ae97e3cf59
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197814
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32847}
VideoReceiveStream2 unnecessarily posts a decode complete call to
its own queue while already being executed on it. A popular use
case in downstream project has a large amount of decoders
in use so the cost of this is multiplied by the number of active
decoders.
Fix this by removing the unnecessary task post. To allow for this,
this change also changes the frame buffer to call out to it's
handler without any locks held.
Bug: webrtc:12297
Change-Id: Ib2e26445458228a44c53dad9d585d4025f2f2945
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197420
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32845}
Lowering a bit since it is currently failing after one of CLs from https://aomedia.googlesource.com/aom.git/+log/87c414ed32..43927e4611
The error is "error: Expected: (video_stat.min_ssim) > (quality_thresholds->min_min_ssim), actual: 0.919629 vs 0.92"
Bug: None
Change-Id: I35e1e989961c6794a7f5f2015f5a8a786f1e25f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197808
Reviewed-by: Jerome Jiang <jianj@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32844}
and reorganise the parsing
Bug: None
Change-Id: I21f08297429a0cc0265da00daa681d934fc43d66
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196643
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32843}
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}
This change moves several functions used to create test windows from
WindowCaptureUtilsTest into a reusable file test_window.cc. This is so
we can reuse this code in the window capturer tests in this CL:
196624: Finish implementing WGC Window Capturer and add unit tests. |
https://webrtc-review.googlesource.com/c/src/+/196624
Bug: webrtc:9273
Change-Id: I62dc53e6c3475e49ca4ae4cf6d1ef02fc8781b89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196623
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Austin Orion <auorion@microsoft.com>
Cr-Commit-Position: refs/heads/master@{#32838}
This test that a new allocation is reported if the input resolution
changes.
Bug: webrtc:12000
Change-Id: Iaf8be1af62bbc8a2ca19b58f0587ceacfcfa5991
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197807
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32837}
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 removes two owners who don't actively work on WebRTC and adds
mbonadei@.
Bug: None
Change-Id: I771e3ce2f97e20d043e074428829d05b39154025
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196650
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32833}
Add a timestamp for last data sent in Connection.
Move calling of rtc::TimeMillis() to Connection and remove it from RateTracker::AddSamples.
This timestamp will be used to further improve fail over logic.
BUG=None
Change-Id: I4cbc7693a0e081277590b9cb13264dc2a998202e
No-Try: True
Change-Id: I4cbc7693a0e081277590b9cb13264dc2a998202e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197421
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32831}
Adding the account that performs the auto-update of the WebRTC version
string.
No-Presubmit: True
No-Try: True
Bug: webrtc:12159
Change-Id: Ie24d0de30a08bb5e21955a90059af982d019110c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197803
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32830}
This is needed in order to prepare Chromium to depend on the new
target to land (https://webrtc-review.googlesource.com/c/src/+/196903).
No-Try: True
Bug: webrtc:9987
Change-Id: Ic7337f1c57c135d541eb424fb06dae32acacaea8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197801
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32829}
This CL adds a string to the resulting WebRTC library (trying to make
sure the version string will be there no matter how WebRTC is packaged).
This CL should be followed by some process to regularly and
automatically update the version string.
No-Try: True
No-Presubmit: True
Bug: webrtc:12159
Change-Id: I9143aeae2cd54d0d4048c138772888100d7873cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191223
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32825}
