Also add explicit includes of rtc_base/string_utils.h in files depending on it.
Bug: webrtc:6424
Change-Id: Id6b53937ab2d185d092a5d8863018fd5f1a88e27
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135744
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27903}
This CL makes it possible to configure the priority of audio streams in
bitrate allocations using field trials.
It also adds the option to forcibly ignore any injected audio allocation
strategy, so that experimentation with allocation won't be blocked on
the work to remove the strategy injection.
Bug: webrtc:10603
Change-Id: Ic36ceee6c15eb0fad275866f77e2a121066e516c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135467
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27881}
optional<int> min_frames: The minimum number frames to observe to make a
scaling decision.
Default: kMinFramesNeededToScale in quality_scaler.cc
optional<double> initial_scale_factor: The sample period scale factor.
Default: kSamplePeriodScaleFactor in quality_scaler.cc
optional<double> scale_factor: Option to use a reduced sampling interval when
last check did not result in an adaptation (if
unset the initial_scale_factor is used).
Bug: none
Change-Id: I3bb955d1f8d7d7d49bc118361614b5aa59605231
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135125
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27860}
(Re-land reverted cr).
Use (in order from lowest to highest precedence):
-- fixed 32000bps
-- fixed target bitrate from codec
-- explicit values from the rtp encoding parameters
-- Final precedence is given to field trial values from
WebRTC-Audio-Allocation
Bug: webrtc:10487
Change-Id: I573e996fa1f243e673785cdbe687e029fd5cbf4a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133483
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Daniel Lee <dklee@google.com>
Cr-Commit-Position: refs/heads/master@{#27847}
Compiling without BoringSSL fails since g_use_time_callback_for_testing
is defined inside a OPENSSL_IS_BORINGSSL block.
Bug: webrtc:10160
Change-Id: I25c27fa8ed128a50aa855db2012026c97954b91b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134226
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27779}
Adding additional usage bits to the UsagePattern to:
- Track whether a mDNS candidate was collected
- Track whether a mDNS candidate was received from the remote peer
- Track whether a private IP address was received from the remote peer
The definition of a private IP address is extended to include 100.64/10 addresses.
Bug: None
Change-Id: I77182685120413d5c13c5f67e480d33fdcaefc6a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134000
Commit-Queue: Jeroen de Borst <jeroendb@webrtc.org>
Reviewed-by: Justin Uberti <juberti@google.com>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27747}
Use (in order from lowest to highest precedence):
-- fixed 32000bps
-- fixed target bitrate from codec
-- explicit values from the rtp encoding parameters
-- Final precedence is given to field trial values from
WebRTC-Audio-Allocation
Bug: webrtc:10487
Change-Id: I7e289f209a927785572058b6fbfdf60fa14edf05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126229
Reviewed-by: Minyue Li <minyue@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Daniel Lee <dklee@google.com>
Cr-Commit-Position: refs/heads/master@{#27667}
This is used to avoid thread processing in simulated time
controller. This saves up to 30% execution time in debug builds.
Bug: webrtc:10365
Change-Id: Ie83dfb2468d371e4687d28c776acf7e23eb411d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133173
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27666}
This CL adds a field trial that enables the EncoderBitrateAdjuster to
allow higher target bitrate if we are not network constrained. We still
don't allow the bitrate to go higher than the average target media rate
though.
Bug: webrtc:10155
Change-Id: Id5995070aa0cbe84b9305a422279141b38664bb1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132717
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27627}
Instead use WebRtcKeyValueConfig and FieldTrialBasedConfig.
The purpose is to allow a user of GoogCC to use different settings on different instances.
BUG=webrtc:10335
Change-Id: I2f837688c9fdd341eecb44484cc784b1c80da1a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132791
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27617}
This CL adds an experiment where aggressiveness of the rate controller
is tuned based on if the application is network constrained or not.
Bug: webrtc:10155
Change-Id: I6c8cd116f57321c5b36cf5a69840913936091aaa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132786
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27615}
We no longer have a need for a HKDF implementation in WebRTC. To keep
code quality high it makes sense to delete this dead code path.
Bug: webrtc:9600
Change-Id: Ibe6ee9150acd9dbf59452372242d857c5ffa65c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132802
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27604}
Make the warning timeout for Event::Wait configurable, and let
NullSocketServer::Wait pass kForever to completely eliminate the
warning.
3000 ms is a good default warning timeout for Event::Wait, but in some
cases---such as when a message queue is waiting for a message to
arrive---we don't want the warning, since a long wait isn't a reliable
indicator that the system is deadlocked. It might just be that no one
is posting messages.
Bug: webrtc:10531
Change-Id: Ic5969b8bfedb96376bd6d6a72ba6a4591750a920
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132017
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27574}
List elements are separated by a |. If the key is given without a : we
treat that as a empty list.
We also support parsing multiple lists as a list-of-structs, see the
unit test for usage examples.
Bug: webrtc:9346
Change-Id: I32d3ce612fef476b1c481c00a893d7fa2f339e92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130464
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27560}
This CL implements Welford's algorithm for a
numerically stable computation of the variance.
This implementation is plugged in SamplesStatsCounter class (adapter pattern).
A 'NumericalStability' unit test has been added,
whose previous implementation of SamplesStatsCounter failed to pass.
Follow-up CLs will factorize more occurences of duplicated and misbehaved
computations.
Bug: webrtc:10412
Change-Id: Id807c3d34e9c780fb1cbd769d30b655c575c88ac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131394
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27547}
this allows merging two stats counter objects, will be used in a future
CL to merge statistics for multiple video layers.
Bug: webrtc:10365
Change-Id: Iee9c48b68dfd7ba29537c14fc5f4a7c1c333d145
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131942
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@google.com>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27534}
Semi-automatically created with:
git grep -l " testing::" | xargs sed -i "s/ testing::/ ::testing::/g"
git grep -l "(testing::" | xargs sed -i "s/(testing::/(::testing::/g"
git cl format
After this, two .cc files failed to compile and I have fixed them
manually.
Bug: webrtc:10523
Change-Id: I4741d3bcedc831b6c5fdc04485678617eb4ce031
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132018
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27526}
When ALR was made default-on we removed the ability to use field trials
to configure alternative ALR detector values. This CL just restores
the ability to force them, defaults are unaffected.
Bug: webrtc:10509
Change-Id: Ibc09e27f1f7b72513de1482d280683802e962497
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131145
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27442}
Wine implements ::QueueUserAPC incorrectly and returns
ERROR_ACCESS_DENIED when the thread is terminating instead of
ERROR_GEN_FAILURE. This is (hopefully) safe to do, assuming
the correct Windows implementation would never use ERROR_ACCESS_DENIED
in an actual failure case. I can't find documentation that says one
way or the other.
Bug: None
Change-Id: If74a40bb7e1cd49cc2266c31684bd88f1c667422
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131042
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27432}
This reverts commit 7276b974b78ea4f409d8738b1b6f1515f7a8968e.
Reason for revert: Changing to a later Chrome release.
Original change's description:
> Disable DTLS 1.0, TLS 1.0 and TLS 1.1 downgrade in WebRTC.
>
> This change disables DTLS 1.0, TLS 1.0 and TLS 1.1 in WebRTC by default. This
> is part of a larger effort at Google to remove old TLS protocols:
> https://security.googleblog.com/2018/10/modernizing-transport-security.html
>
> For the M74 timeline I have added a disabled by default field trial
> WebRTC-LegacyTlsProtocols which can be enabled to support these cipher suites
> as consumers move away from these legacy cipher protocols but it will be off
> in Chrome.
>
> This is compliant with the webrtc-security-arch specification which states:
>
> All Implementations MUST implement DTLS 1.2 with the
> TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256 cipher suite and the P-256
> curve [FIPS186]. Earlier drafts of this specification required DTLS
> 1.0 with the cipher suite TLS_ECDHE_ECDSA_WITH_AES_128_CBC_SHA, and
> at the time of this writing some implementations do not support DTLS
> 1.2; endpoints which support only DTLS 1.2 might encounter
> interoperability issues. The DTLS-SRTP protection profile
> SRTP_AES128_CM_HMAC_SHA1_80 MUST be supported for SRTP.
> Implementations MUST favor cipher suites which support (Perfect
> Forward Secrecy) PFS over non-PFS cipher suites and SHOULD favor AEAD
> over non-AEAD cipher suites.
>
> Bug: webrtc:10261
> Change-Id: I847c567592911cc437f095376ad67585b4355fc0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125141
> Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> Reviewed-by: David Benjamin <davidben@webrtc.org>
> Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27006}
TBR=steveanton@webrtc.org,davidben@webrtc.org,qingsi@webrtc.org,benwright@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10261
Change-Id: I34727e65c069e1fb2ad71838828ad0a22b5fe811
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130367
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27403}
This is useful in tests as it allows overriding the default after
construction. It's not intended for use in production (as it can
be confusing to readers).
Bug: webrtc:10365
Change-Id: I8ac2541f2626e7fddbb61bdae72e9571ce9d7b97
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130468
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27389}
And delete corresponding dependencies on :webrtc_common. After this
change, common_types.h is included directly only from code in the
following directories:
api/
api/video/
api/video_codecs/
common_video/libyuv/include/
media/base/
modules/remote_bitrate_estimator/
modules/rtp_rtcp/source/
modules/video_coding/codecs/vp9/
There remains plenty of indirect dependencies on the types declared in
common_types.h, but the fewer direct dependencies should make it
easier to find the proper place for each type.
Bug: webrtc:5876
Change-Id: I93e8f214025ecb613c19fdec2015bd3f96c59aae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27376}
Some NaCl system headers live in a special directory and the
toolchain doesn't propagate the -I compiler flag [2].
A common workaround in Chromium is to use 'public_deps' in order
to propagate //native_client_sdk/src/libraries/nacl_io:nacl_io_include_dirs
one step further in the build graph.
[1] - https://cs.chromium.org/chromium/src/native_client_sdk/src/libraries/nacl_io/
[2] - -Inative_client_sdk/src/libraries/third_party/newlib-extras
Bug: chromium:925028
Change-Id: I5145b80c2ae6969f79fcbfcf93a6b05c8a122746
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129701
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Nico Weber <thakis@chromium.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27373}
Functions htonll and ntohll are only available when
NTDDI_VERSION>=NTDDI_WIN8 or INCL_EXTRA_HTON_FUNCTIONS is defined,
instead of assuming this to be true, this CL replaces them with
_byteswap_uint64 [1].
On top of that, the following functions were assuming host to be
little endian on Windows and NaCl:
- htobe16(v)
- htobe32(v)
- be16toh(v)
- be32toh(v)
- htobe64(v)
- be64toh(v)
But it is the application's responsibility to check the host
endianness before calling ntohs, ntohl (and probably also htons and
htonl). See [2], especially: "The ntohs function returns the value
in host byte order. If the netshort parameter is already in host byte
order, then this function will reverse it. It is up to the application
to determine if the byte order must be reversed.".
After this CL, WebRTC should do the right thing based on the value
of WEBRTC_ARCH_{BIG,LITTLE}_ENDIAN.
[1] - https://docs.microsoft.com/en-us/cpp/c-runtime-library/reference/byteswap-uint64-byteswap-ulong-byteswap-ushort
[2] - https://docs.microsoft.com/en-us/windows/desktop/api/winsock/nf-winsock-ntohs
Bug: None
Change-Id: I61ca882ad81dd090fd164b0fdfeec64cd088a71d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129901
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Noah Richards <noahric@chromium.org>
Reviewed-by: Yves Gerey <yvesg@google.com>
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27371}
Introduces SequenceChecker, merging the functionality of ThreadChecker
and SequencedTaskChecker. Also making the two latter use the former as
the underlying implementation for backwards compatibility.
This allows code that uses thread checker to accept running on a thread
pool backed task queue.
Bug: webrtc:10365
Change-Id: Ifefc4925694f263088a8a095fdf98a2407c62081
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129721
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27365}
