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}
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}
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 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 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}
After some recent change current thread while creating the receive stream is
used as a task queue for stats calculation.
Currently, video_replay tool doesn't create streams inside a task queue, so
it ends up posting tasks to a "dead" task queue, which doesn't run message
processing loop at all.
Bug: webrtc:12204
Change-Id: Ieb97a10f44a11e92e2ac08df5b39b7cd695c852e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196860
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philipp Hancke <philipp.hancke@googlemail.com>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32824}
webrtc::test::GetPerfResults() relies on a singleton and this makes
some tests be order dependent (running in a different order makes them
fail).
A good fix is to remove the singleton but this CL at least makes the
fragile test set up the environment correctly.
No-Try: True
Bug: None
Change-Id: I7ad25f685f0bc5d246beeadedfa9f5a39f3547e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197425
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32823}
