This change prevents decoding corruption by not allowing keyframes with a
newer frame id but an older timestamp to be decoded. This does not handle
reordering well.
Bug: none
Change-Id: I4a67ca84ee86a782da74a10530c531d893d3bd3c
Reviewed-on: https://webrtc-review.googlesource.com/c/107304
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25292}
This code is not used anymore, but it's not possible to land this CL
until issue webrtc:9665 is fixed.
Bug: webrtc:9642,webrtc:9665
Change-Id: Idb68e9bdf51b4239788cd6869dcb44dae87d7c56
Reviewed-on: https://webrtc-review.googlesource.com/c/95951
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25289}
Compute these states in jseptransportController and store them. Eventually they should be passed on to the peer connection observer and exposed in the blink layer.
Bug: webrtc:9308
Change-Id: Ifdec39c24a607fcb8211c4acf6b9704eaff371b1
Reviewed-on: https://webrtc-review.googlesource.com/c/103506
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25288}
This CL makes it possible to change transport routes while running
a scenario based test.
To make this possible in a consistent manner, the scenario test
framework is modified to only allow shared transport for all streams
between two CallClients. This is what typically is done in practice and
it is quite complex to even reason about the implications of using
mixed transports for a single call.
Bug: webrtc:9718
Change-Id: Ib836928feed98aa2bbbe0295e158157a6518348b
Reviewed-on: https://webrtc-review.googlesource.com/c/107200
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25287}
In this CL we change the signal that controls the updates of the ERLE estimator. Until now, the render signal was used which is not optimum for reverberant signals. In this CL, a reverberation has been added to the the render signal and this new signal has been used for controlling when to update the ERLE estimator.
Bug: webrtc:9873
Change-Id: I0ebea3fc208f97aa237af015ba543015d49ed978
Reviewed-on: https://webrtc-review.googlesource.com/c/105660
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25285}
These things are no longer brand new, so it makes even less sense
than it once did to warn users that they may change at any time.
Bug: none
Change-Id: I43a6915d9e00fbfef30fdb89869873b129297c8d
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/106980
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25283}
This is a reland of 8dc280d804bc49932f429955b27ff385054ddfa0
Original change's description:
> Make sure Chromium will pick the correct field_trial/metric impl.
>
> Chromium wants to pick its own field_trial and metrics implementation,
> and WebRTC has an escape hatch to let it remove the definition of these
> symbols. This obviously causes liker errors if Chromium will not pick
> the correct dependency (which cannot be forced by GN check since there
> is no header inclusion).
>
> Instead of hoping that the build target with the correct implementation
> will be added as a dependency somewhere in the build graph, this CL
> wants to explicitly add the dependency as close as possible to the
> headers, so if they are included and WebRTC is built as part of Chromium
> the correct implementation will be linked into the binary.
>
> Bug: webrtc:9631
> Change-Id: I2d1cfe541726341adfdf81e14e036464aa003d4d
> Reviewed-on: https://webrtc-review.googlesource.com/c/107040
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25269}
Bug: webrtc:9631
Change-Id: I9b6e6f3f91881e8e3fa8bc6e97f797e8b7e4e4ca
Reviewed-on: https://webrtc-review.googlesource.com/c/107163
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25281}
- Similar to network priority
- Still requires MediaConfig.enable_dscp = true (i.e. googDscp == true to peerconnection)
- Needs followups 1) Specify value in chrome renderer js idl 2) disable audio bwe when value differs from video 3)remove googDscp guard
Bug: webrtc:5008
Change-Id: Ibdcbb1183f0ca2ae85e3bced6d0aedbccae3ced4
Reviewed-on: https://webrtc-review.googlesource.com/c/93560
Commit-Queue: Tim Haloun <thaloun@chromium.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25280}
A blob is a string of binary information, whose length may not
necessarily be determined by looking into the string, so that
concatenating all blobs without explicitly including their lengths
as part of their encoding is not a viable option.
Bug: webrtc:8111
Change-Id: I89fdca660e89a6a71eff3ecb7b86416312b81f23
Reviewed-on: https://webrtc-review.googlesource.com/c/104201
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25278}
This is a reland of 5ccdc1331fcc3cd78eaa14408fe0c38d37a5a51d
Original change's description:
> Prefix flag macros with WEBRTC_.
>
> Macros defined in rtc_base/flags.h are intended to be used to define
> flags in WebRTC's binaries (e.g. tests).
>
> They are currently not prefixed and this could cause problems with
> downstream clients since these names are quite common.
>
> This CL adds the 'WEBRTC_' prefix to them.
>
> Generated with:
>
> for x in DECLARE DEFINE; do
> for y in bool int float string FLAG; do
> git grep -l "\b$x\_$y\b" | \
> xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
> done
> done
> git cl format
>
> Bug: webrtc:9884
> Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
> Reviewed-on: https://webrtc-review.googlesource.com/c/106682
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25270}
TBR=kwiberg@webrtc.org
Bug: webrtc:9884
Change-Id: I5ba5368a231a334d135ed5e6fd7a279629ced8a3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/107161
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25277}
This reverts commit 5ccdc1331fcc3cd78eaa14408fe0c38d37a5a51d.
Reason for revert: Breaks downstream project.
Original change's description:
> Prefix flag macros with WEBRTC_.
>
> Macros defined in rtc_base/flags.h are intended to be used to define
> flags in WebRTC's binaries (e.g. tests).
>
> They are currently not prefixed and this could cause problems with
> downstream clients since these names are quite common.
>
> This CL adds the 'WEBRTC_' prefix to them.
>
> Generated with:
>
> for x in DECLARE DEFINE; do
> for y in bool int float string FLAG; do
> git grep -l "\b$x\_$y\b" | \
> xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
> done
> done
> git cl format
>
> Bug: webrtc:9884
> Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
> Reviewed-on: https://webrtc-review.googlesource.com/c/106682
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25270}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
Change-Id: Ia79cd6066ecfd1511c34f1b30fd423e560ed6854
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9884
Reviewed-on: https://webrtc-review.googlesource.com/c/107160
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25276}
This reverts commit 8dc280d804bc49932f429955b27ff385054ddfa0.
Reason for revert: Breaks NDK compile on Chrome bots: https://logs.chromium.org/logs/chromium/buildbucket/cr-buildbucket.appspot.com/8932231862774278176/+/steps/compile__with_patch_/0/stdout
Original change's description:
> Make sure Chromium will pick the correct field_trial/metric impl.
>
> Chromium wants to pick its own field_trial and metrics implementation,
> and WebRTC has an escape hatch to let it remove the definition of these
> symbols. This obviously causes liker errors if Chromium will not pick
> the correct dependency (which cannot be forced by GN check since there
> is no header inclusion).
>
> Instead of hoping that the build target with the correct implementation
> will be added as a dependency somewhere in the build graph, this CL
> wants to explicitly add the dependency as close as possible to the
> headers, so if they are included and WebRTC is built as part of Chromium
> the correct implementation will be linked into the binary.
>
> Bug: webrtc:9631
> Change-Id: I2d1cfe541726341adfdf81e14e036464aa003d4d
> Reviewed-on: https://webrtc-review.googlesource.com/c/107040
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25269}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
Change-Id: I4c04b60d198aa1b89278083d518bfa93c15d09bb
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9631
Reviewed-on: https://webrtc-review.googlesource.com/c/107140
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Niklas Enbom <niklas.enbom@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25275}
We want sanitizer bots to show failure only for sanitizer defects.
To do so, this CL forces exit code to 0 unconditionally.
Sanitized binaries will turn it to 66 if there is any defect with diagnostic.
Bug: webrtc:9849
Change-Id: I46b683dcae12b76f1be177603af59e3f34bff3a9
Reviewed-on: https://webrtc-review.googlesource.com/c/107060
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25273}
This should prevent us from posting and deadlocking if EglRenderer
thread crashes.
Bug: b/117400268
Change-Id: I978738249917cb5194917b0b2b12f67bb2a8642e
Reviewed-on: https://webrtc-review.googlesource.com/c/107043
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25271}
Macros defined in rtc_base/flags.h are intended to be used to define
flags in WebRTC's binaries (e.g. tests).
They are currently not prefixed and this could cause problems with
downstream clients since these names are quite common.
This CL adds the 'WEBRTC_' prefix to them.
Generated with:
for x in DECLARE DEFINE; do
for y in bool int float string FLAG; do
git grep -l "\b$x\_$y\b" | \
xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
done
done
git cl format
Bug: webrtc:9884
Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
Reviewed-on: https://webrtc-review.googlesource.com/c/106682
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25270}
Chromium wants to pick its own field_trial and metrics implementation,
and WebRTC has an escape hatch to let it remove the definition of these
symbols. This obviously causes liker errors if Chromium will not pick
the correct dependency (which cannot be forced by GN check since there
is no header inclusion).
Instead of hoping that the build target with the correct implementation
will be added as a dependency somewhere in the build graph, this CL
wants to explicitly add the dependency as close as possible to the
headers, so if they are included and WebRTC is built as part of Chromium
the correct implementation will be linked into the binary.
Bug: webrtc:9631
Change-Id: I2d1cfe541726341adfdf81e14e036464aa003d4d
Reviewed-on: https://webrtc-review.googlesource.com/c/107040
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25269}
This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
to mark WebRTC symbols as visible from a shared library, this doesn't
mean these symbols are part of the public API (please continue to refer
to [1] for info about what is considered public WebRTC API).
[1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md
Bug: webrtc:9419
Change-Id: I452117a8385bb08f86c4863bb1079d3774a16a0d
Reviewed-on: https://webrtc-review.googlesource.com/c/107042
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25268}
These two files were using absl::make_unique without #including the
header that declares it.
Bug: None
Change-Id: I03019c9a7e06370631680b474d04dd33716b0fe3
Reviewed-on: https://webrtc-review.googlesource.com/c/107041
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25266}
In the past, it would incorrectly set up a state for 'use_media_transport' (i.e. it could say "use_media_transport" is true, but jseptransportcontroller wouldn't know about that).
Also, removes unnecessary field (unused).
Bug: webrtc:9719
Change-Id: I7e5c0ce81b3b70f63c49d661d95b95b5bcbb0c68
Reviewed-on: https://webrtc-review.googlesource.com/c/106960
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25263}
This CL moves the implementations of the fake media engine from
fakemediaengine.h to fakemediaengine.cc.
Bug: webrtc:9883
Change-Id: I0f91ef63a366abe9638fc885bc14aba7dd5436aa
Reviewed-on: https://webrtc-review.googlesource.com/c/106923
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25260}
This change integrates the FrameDecryptorInterface and the FrameEncryptorInterface into
the video send and receive path. If a FrameEncryptorInterface is set on an outgoing video RTPSender
then each outgoing video frame will first pass through the provided FrameEncryptor which
will have a chance to modify the payload contents for the purposes of encryption. In addition to
this the new GenericFrameDescriptor will be added as additional data.
If a FrameDecryptorInterface is set on an incoming video RtpReceiver then each incoming
video payload will first pass through the provided FrameDecryptor which have a chance to
modify the payload contents for the purpose of decryption.
Bug: webrtc:9795
Change-Id: I9f743ce0cb63df0cf070f6144be7ada078b4e5d2
Reviewed-on: https://webrtc-review.googlesource.com/c/103920
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25258}
There is no need to redefine SSL_CTX. base.h/ossl_typ.h defines it
already. Additionally, switch the base.h includes to the
OpenSSL-compatible ossl_typ.h spelling. That just got landed in
https://webrtc-review.googlesource.com/c/104120, so I'm guessing
OpenSSL consumers just didn't notice yet.
While getting the current BoringSSL name mangling scheme working with
WebRTC is a ways off, one of the requirements will almost certainly be
that WebRTC never forward-declare any BoringSSL types itself, instead
leaving it to openssl/base.h (or openssl/ossl_typ.h, the
OpenSSL-compatible alias). This is because we'd need to rename the
struct names themselves where they participate in C++ name mangling.
E.g. std::pair<RSA*, int> would mangle as rsa_st.
Bug: webrtc:5664
Change-Id: Ib9695d4ae4bc07d2bc54c9fdfb8600f44b5ec7bb
Reviewed-on: https://webrtc-review.googlesource.com/c/106675
Commit-Queue: David Benjamin <davidben@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25257}
Add code for delta-encoding and decoding, to be used when producing
WebRTC event logs of the new format.
This CL supports fixed-size encoding only. Also, no support for
signed deltas or optional values yet. These will be added in
subsequent CLs.
Bug: webrtc:8111
Change-Id: I531abd99fd924f4c9e692abe565bc6f66c875ad5
Reviewed-on: https://webrtc-review.googlesource.com/c/100304
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25256}
Now WebRTC.Video.MediaBitrateReceived.S0 UMA metric will be counted more
correctly. Before, only keyframes were counted there. Now except some
occasional reorderings near content_type switch, all frames should be
counted correctly.
Note,
WebRTC.Video.MediaBitrateReceived will still be larger than sum of sliced
variants because it includes header overhead while sliced metrics do not.
Bug: none
Change-Id: Ia25d6e3efb572f3fe2e9651996b2243716698140
Reviewed-on: https://webrtc-review.googlesource.com/c/106702
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25253}
With this CL, the main and shadow filters are no longer fully reset to
0 as the delay changes. This allows for more robust echo removal for
some scenarios.
Bug: webrtc:9879,chromium:895838
Change-Id: I859aa3df3ae41648bc8efde01ec2e2a5cb392279
Reviewed-on: https://webrtc-review.googlesource.com/c/106345
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25251}
This is needed for absl::make_unique. absl/memory/memory.h is included
through absl/types/optional.h on C++14 mode, but is not on C++17 mode.
Bug: chromium:752720
Change-Id: I28c0dfc9c37910bcb8f0c0bbe40cdd47f2105e50
Reviewed-on: https://webrtc-review.googlesource.com/c/106760
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25247}
This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
to mark WebRTC symbols as visible from a shared library, this doesn't
mean these symbols are part of the public API (please continue to refer
to [1] for info about what is considered public WebRTC API).
[1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md
Bug: webrtc:9419
Change-Id: I12ef6f85ccef7dae3afe9ecff99725af13d551e2
Reviewed-on: https://webrtc-review.googlesource.com/c/106684
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25246}
