Commit graph

341 commits

Author SHA1 Message Date
Björn Terelius
ab229b0706 Add documentation for RTC event log
Bug: webrtc:12841
Change-Id: I9312a4660b8fd039019795a0a90b2cda25dc773c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221045
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34210}
2021-06-03 09:03:18 +00:00
Björn Terelius
2fa4774067 Revert "Deprecate microsecond timestamps in RTC event log."
This reverts commit e6ee8fab7e.

Reason for revert: Breaks downstream test

Original change's description:
> Deprecate microsecond timestamps in RTC event log.
>
> (Microsecond timestamps are only used in the legacy wire-format,
> and the clocks only have microsecond resolution on some platforms.)
>
> Also convert structs on the parsing side to use a Timestamp instead
> of a uint64_t to represent the log time.
>
> Bug: webrtc:11933
> Change-Id: Ide5a0217d99f13f2e243115b163f13e0525648c7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219467
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34097}

TBR=terelius@webrtc.org,srte@webrtc.org,crodbro@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com

Change-Id: I478c9a4a1664b984891c4fcfc78f0ce9a51fe4c0
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11933
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219636
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34100}
2021-05-24 13:11:10 +00:00
Björn Terelius
e6ee8fab7e Deprecate microsecond timestamps in RTC event log.
(Microsecond timestamps are only used in the legacy wire-format,
and the clocks only have microsecond resolution on some platforms.)

Also convert structs on the parsing side to use a Timestamp instead
of a uint64_t to represent the log time.

Bug: webrtc:11933
Change-Id: Ide5a0217d99f13f2e243115b163f13e0525648c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219467
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34097}
2021-05-24 11:39:02 +00:00
Björn Terelius
61a287a3cb Add accessor for UTC start time in event log
Bug: webrtc:11933
Change-Id: Id9e63dc0487d5d07ac87e319695206ee4cd627e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219787
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34079}
2021-05-21 17:03:01 +00:00
Björn Terelius
0cff39137b Start with a BeginLog event in event log encoder unittest
Also rename encoding_ to encoding_type_

Bug: webrtc:11933
Change-Id: If4848199b96e9de612695dfe7ec52266ccd80bd3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219285
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34070}
2021-05-20 17:44:30 +00:00
Björn Terelius
a77e16ca2c Update BitBuffer methods to style guide
Specifically, use reference instead of pointer for out parameter
and place the out parameter last, for the following methods

ReadUInt8
ReadUInt16
ReadUInt32
ReadBits
PeekBits
ReadNonSymmetric
ReadSignedExponentialGolomb
ReadExponentialGolomb

Bug: webrtc:11933
Change-Id: I3f1efe3e29155985277b0cd18700ddea25fe7914
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218504
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34037}
2021-05-18 11:10:27 +00:00
Danil Chapovalov
6e661f47cf Change rtc event log packet messages implementation to save full rtp packet
Keeping just the header doesn't save memory because header is taken as slice
of the original packet (and thus keeps a reference to the buffer containing
full packet)
Keeping full packet is simpler and avoid extra unused buffer created during
RtpPacket default contruction

Bug: b/187593466
Change-Id: I78d7201d110092fc039203e1caa2fb9c3afbc079
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218161
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33974}
2021-05-11 08:06:54 +00:00
Björn Terelius
b37180fcf2 Remove use of istream in RTC event log parser.
Bug: webrtc:11933,webrtc:8982
Change-Id: I8008eb704549e690d7c778f743a5b9cd0c52892c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208941
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33603}
2021-03-31 13:21:58 +00:00
Bjorn Terelius
c4d3e34d36 Clean up temporary event log file after test.
Bug: webrtc:12084
Change-Id: If17140b6af8f88faf7808645ca8998a5540aad06
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212963
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33564}
2021-03-25 14:32:45 +00:00
Danil Chapovalov
e904161cec Replace RTC_DEPRECATED with ABSL_DEPRECATED
This remove webrtc-specific macro that has no reason to be webrtc specific
ABSL_DEPRECATED takes a message parameter encouraging to write text how class or function is deprecated.

Bug: webrtc:12484
Change-Id: I89f1398f91dacadc37f7db469dcd985e3724e444
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208282
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33314}
2021-02-22 12:53:23 +00:00
Artem Titov
d15a575ec3 Use SequenceChecker from public API
Bug: webrtc:12419
Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 15:04:55 +00:00
Björn Terelius
4ef5638871 Parse and plot RTCP BYE in RTC event log.
Bug: webrtc:12432
Change-Id: I9a98876044e0e75ee4f3ef975ae75237606d108d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204380
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33161}
2021-02-04 11:28:46 +00:00
Mirko Bonadei
14cad9fa35 Fix clang-tidy: performance-inefficient-vector-operation.
Bug: None
Change-Id: Ieb3b49436c075047e1d9e0293dd94f754c652b01
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205520
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33150}
2021-02-03 15:18:51 +00:00
Andrey Logvin
e7c79fd3d6 Remove from chromium build targets that are not compatible with it.
We need to be able build chromium with rtc_include_tests = true. It
reveals a lot of targets that are not compatible with chromium but
aren't marked so.

`rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated.

Bug: webrtc:12404
Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 13:46:19 +00:00
Danil Chapovalov
4f281f142a Cleanup FakeRtcEventLog from thread awareness
To avoid it relying on AsyncInvoker.

Bug: webrtc:12339
Change-Id: I086305a74cc05fc8ed88a651e71a8f707c2c1d5c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202252
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33044}
2021-01-20 14:06:47 +00:00
Mirko Bonadei
e5f4c6b8d2 Reland "Refactor rtc_base build targets."
This is a reland of 69241a93fb

Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which
affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5.
The original CL didn't attach the definition of the macro
NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have
to be related to //rtc_base anymore but to //rtc_base:threading).

Original change's description:
> Refactor rtc_base build targets.
>
> The "//rtc_base:rtc_base" build target has historically been one of the
> biggest targets in the WebRTC build. Big targets are the main source of
> circular dependencies and non-API types leakage.
>
> This CL is a step forward into splitting "//rtc_base:rtc_base" into
> smaller targets (as originally started in 2018).
>
> The only non-automated changes are (like re-wiring the build system):
> * The creation of //rtc_base/async_resolver.{h,cc} which allows to
>   break a circular dependency (is has been extracted from
>   //rtc_base/net_helpers.{h,cc}).
> * The creation of //rtc_base/internal/default_socket_server.{h,cc} to
>   break another circular dependency.
>
> Bug: webrtc:9987
> Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32941}

Bug: webrtc:9987
Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33001}
2021-01-15 17:00:05 +00:00
Niels Möller
d76dcbd963 Simplify FakeRtcEventLog, delete rtc::Bind usage
Bug: webrtc:11339
Change-Id: I2a250934daf0a9114ef8c03464034b1efd8c4c35
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201722
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32990}
2021-01-15 08:48:47 +00:00
Mirko Bonadei
7acc2d9fe3 Revert "Refactor rtc_base build targets."
This reverts commit 69241a93fb.

Reason for revert: Breaks WebRTC roll into Chromium.

Original change's description:
> Refactor rtc_base build targets.
>
> The "//rtc_base:rtc_base" build target has historically been one of the
> biggest targets in the WebRTC build. Big targets are the main source of
> circular dependencies and non-API types leakage.
>
> This CL is a step forward into splitting "//rtc_base:rtc_base" into
> smaller targets (as originally started in 2018).
>
> The only non-automated changes are (like re-wiring the build system):
> * The creation of //rtc_base/async_resolver.{h,cc} which allows to
>   break a circular dependency (is has been extracted from
>   //rtc_base/net_helpers.{h,cc}).
> * The creation of //rtc_base/internal/default_socket_server.{h,cc} to
>   break another circular dependency.
>
> Bug: webrtc:9987
> Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32941}

TBR=mbonadei@webrtc.org,hta@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

No-Try: True
Bug: webrtc:9987
Change-Id: I1e36ad64cc60092f38d6886153a94f1a58339256
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201840
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32986}
2021-01-14 21:27:38 +00:00
Björn Terelius
6f597bd2ab Move logged types for RTC event log into event headers.
Bug: webrtc:11933
Change-Id: Idf5c85a3b33147b20e8646903de7e704b0cef18c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201203
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32951}
2021-01-12 16:45:40 +00:00
Mirko Bonadei
69241a93fb Refactor rtc_base build targets.
The "//rtc_base:rtc_base" build target has historically been one of the
biggest targets in the WebRTC build. Big targets are the main source of
circular dependencies and non-API types leakage.

This CL is a step forward into splitting "//rtc_base:rtc_base" into
smaller targets (as originally started in 2018).

The only non-automated changes are (like re-wiring the build system):
* The creation of //rtc_base/async_resolver.{h,cc} which allows to
  break a circular dependency (is has been extracted from
  //rtc_base/net_helpers.{h,cc}).
* The creation of //rtc_base/internal/default_socket_server.{h,cc} to
  break another circular dependency.

Bug: webrtc:9987
Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32941}
2021-01-11 18:32:30 +00:00
Harald Alvestrand
cffaf0aea4 Inclusive language: Remove a couple of occurences of "whitelist"
No-Try: True
Bug: webrtc:11680
Change-Id: I50e2d313be962551a8a1f530f430fbd551a8d3e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/200701
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32933}
2021-01-11 07:53:03 +00:00
Björn Terelius
b26335a116 Add static constexpr type to RTC event log events.
This allows (among other things) type-checked down-casts (similar to dynamic_cast) This will be used in a follow-up CL.

This CL also moves some one-liner functions from the .cc file to the .h file.

Bug: webrtc:11933
Change-Id: Ic89de8fa6c445ecbe108f2fbf68b44b655f819f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/199970
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32905}
2021-01-05 13:48:41 +00:00
Niels Möller
08d2c2bf46 Delete unneeded dependencies on the Module abstraction
Bug: webrtc:7219
Change-Id: I1bcbab7e30f9964798a093e888b07d758cf226e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198124
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32865}
2020-12-21 09:09:57 +00:00
Niels Möller
0d863f72a8 Cleanup of bwe_defines.h
Delete unused macros BWE_MIN and BWE_MAX.

Move enum RateControlState: Make it a private enum class in
AimdRateControl, the only user.

Change users of the header file that only need BandwidthUsage, to
instead include api/network_state_predictor.h, the file defining this
class. As a result, fewer dependencies on
modules/remote_bitrate_estimator.

Bug: None
Change-Id: I4450c79dd58e1875d64dd74d1ae2cb7d911a14b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195222
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32704}
2020-11-26 12:26:02 +00:00
Mirko Bonadei
20e4c80fbe Reland "Introduce RTC_NO_UNIQUE_ADDRESS."
This is a reland of f5e261aaf6

This CL disables RTC_NO_UNIQUE_ADDRESS on MSan builds since
there have been some issues.

Original change's description:
> Introduce RTC_NO_UNIQUE_ADDRESS.
>
> This macro introduces the possibility to suggest the compiler that a
> data member doesn't need an address different from other non static
> data members.
>
> The usage of a macro is to maintain portability since at the moment
> the attribute [[no_unique_address]] is only supported by clang
> with at least -std=c++11 but it should be supported by all the
> compilers starting from C++20.
>
> Bug: webrtc:11495
> Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32246}

Bug: webrtc:11495, webrtc:12218
Change-Id: I4e6c7cc37d3daffad2407c9a2acfa897fa5b426a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189968
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32668}
2020-11-23 11:29:36 +00:00
Björn Terelius
55b3ccd021 Fix incorrect ToUnsigned in RTC event log.
Bug: None
Change-Id: I9038ac69c253975a4fc5e074aa13b2573efab9ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181462
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32537}
2020-11-02 16:50:36 +00:00
Bjorn Terelius
945b7d8e31 Add test for logging of large compound RTCP packets.
Bug: chromium:1134107
Change-Id: Ic6ce50d33700c05733747584ce45480660cf64c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188583
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32445}
2020-10-19 21:52:38 +00:00
Mirko Bonadei
0abd518abd Revert "Introduce RTC_NO_UNIQUE_ADDRESS."
This reverts commit f5e261aaf6.

Reason for revert: Breaks downstream projects.

Original change's description:
> Introduce RTC_NO_UNIQUE_ADDRESS.
>
> This macro introduces the possibility to suggest the compiler that a
> data member doesn't need an address different from other non static
> data members.
>
> The usage of a macro is to maintain portability since at the moment
> the attribute [[no_unique_address]] is only supported by clang
> with at least -std=c++11 but it should be supported by all the
> compilers starting from C++20.
>
> Bug: webrtc:11495
> Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32246}

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:11495
Change-Id: Ice318d1b11ca3dff09c190187a0b0a32ca945fe3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186944
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32335}
2020-10-07 07:37:01 +00:00
Bjorn Terelius
9f0c89bd56 Allow RTCP packets longer than 1500 bytes in RTC event log.
Bug: chromium:1134107
Change-Id: I05da32c57537c3c2fddae96918ff4e4685d62043
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186720
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32315}
2020-10-05 15:26:54 +00:00
Mirko Bonadei
f5e261aaf6 Introduce RTC_NO_UNIQUE_ADDRESS.
This macro introduces the possibility to suggest the compiler that a
data member doesn't need an address different from other non static
data members.

The usage of a macro is to maintain portability since at the moment
the attribute [[no_unique_address]] is only supported by clang
with at least -std=c++11 but it should be supported by all the
compilers starting from C++20.

Bug: webrtc:11495
Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32246}
2020-09-30 09:52:49 +00:00
Björn Terelius
03fd7930c6 Allow more than 2 encoders in RtcEventLogEncoderTest
Bug: webrtc:11933
Change-Id: Iabec44eecbd41b0834a1a7105d344ea52fa1aeae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184513
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32139}
2020-09-18 13:59:01 +00:00
Ying Wang
bd8409b70f Minor fixes to avoid crash on some traces that have unexpected data.
Bug: webrtc:0
Change-Id: I6950004be2c725c1d13889f37e4a6208ca41f47e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178909
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32138}
2020-09-18 11:54:24 +00:00
Per Kjellander
db724a1a23 Ensure RtcEventLogEncoderNewFormat::EncodeRemoteEstimate handles infite
numbers

Bug: webrtc:11878
Change-Id: I3c2a2ef6b8cba0ddb2bf00d84c279d89cbe64478
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182500
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31988}
2020-08-25 09:22:49 +00:00
Björn Terelius
bcdfc8975e Group decoded frame events by SSRC when compressing RTC event log.
Correspondingly, change the parser so that it provides the frames
grouped by SSRC.

Also fix a small bug that made the audio playout test terminate
too early before verifying correct logging of all events.

Bug: webrtc:8802
Change-Id: I363ef120cf88fe99290998cbc14ab5dbf32e9607
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181066
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31962}
2020-08-19 09:47:20 +00:00
Björn Terelius
e61f38cf43 Add missing tests for DTLS state logging in RTC event log.
Bug: None
Change-Id: I43842d330b9575825445053a0142988af86f432f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181065
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31881}
2020-08-07 11:49:43 +00:00
Björn Terelius
00c12f6779 Add logging of decoded video frames.
This CL adds the possibility to log metainformation about
decoded frames in RTC event log, including encoding parsing
and tests. It will be wired up in a followup CL.


Bug: webrtc:8802
Change-Id: Ied598b266513d0f63fce0484d741af1782607e74
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181061
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31873}
2020-08-06 17:33:24 +00:00
Magnus Flodman
55afe3885b Search and replace gendered terms according to style guide:
https://chromium.googlesource.com/chromium/src/+/master/styleguide/inclusive_code.md#tools

Not changin the transcipt in
resources/audio_processing/conversational_speech/README.md

BUG=webrtc:11680

Change-Id: I36af34e4a4e0ec6161093c0045b7bbe1dbe4eb45
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177016
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31514}
2020-06-12 14:12:54 +00:00
Mirko Bonadei
2dcf348011 Use absl_deps in order to preapre to the Abseil component build release.
Bug: webrtc:1046390
Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-08 12:59:40 +00:00
Danil Chapovalov
ed5d594730 Replace mock macros with unified MOCK_METHOD macro
Bug: webrtc:11564
Change-Id: I6398b052ec85d2f739755723629bc5da98fb30e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176180
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31376}
2020-05-28 14:11:53 +00:00
Björn Terelius
0d1b28cf09 Replace inconsistent log_segments() function in RTC event log parser
Bug: webrtc:11566
Change-Id: I739bbc29ae5423f3fedcc08e991e27fa0af840c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176081
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31371}
2020-05-27 20:05:13 +00:00
Bjorn Terelius
b914819944 Reduce alert spam in rtc_event_log_visualizer
Bug: webrtc:11564
Change-Id: I4fdd6284b35cedded4d8b623dc0b7f8e1534c495
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175649
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31324}
2020-05-19 13:22:20 +00:00
Bjorn Terelius
48b8279813 Refactor/reimplement RTC event log triage alerts.
- Moves AnalyzerConfig and helper functions IsAudioSsrc, IsVideoSsrc, IsRtxSsrc, GetStreamNam and GetLayerName to analyzer_common.h
- Moves log_segments() code to rtc_event_log_parser.h
- Moves TriageAlert/Notification code to a new file with a couple of minor fixes to make it less spammy.

Bug: webrtc:11566
Change-Id: Ib33941d8185f7382fc72ed65768e46015e0320de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174824
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31318}
2020-05-19 09:45:16 +00:00
Danil Chapovalov
cad3e0e2fa Replace DataSize and DataRate factories with newer versions
This is search and replace change:
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataSize::Bytes<\(.*\)>()/DataSize::Bytes(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataSize::bytes/DataSize::Bytes/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::BitsPerSec<\(.*\)>()/DataRate::BitsPerSec(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::BytesPerSec<\(.*\)>()/DataRate::BytesPerSec(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::KilobitsPerSec<\(.*\)>()/DataRate::KilobitsPerSec(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::bps/DataRate::BitsPerSec/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::kbps/DataRate::KilobitsPerSec/g"
git cl format

Bug: webrtc:9709
Change-Id: I65aaca69474ba038c1fe2dd8dc30d3f8e7b94c29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168647
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30545}
2020-02-18 16:09:50 +00:00
Danil Chapovalov
0c626afcf3 Use newer version of TimeDelta and TimeStamp factories in webrtc
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g"
git cl format

Bug: None
Change-Id: I87469d2e4a38369654da839ab7c838215a7911e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168402
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30491}
2020-02-10 12:21:17 +00:00
Mirko Bonadei
ccbe95fd8a Reformat GN files.
`gn format` recently [1] changed its formatting behavior
for deps, source, and a few other elements when they
are assigned (with =) single-element lists to be consistent
with the formatting of updates (with +=) with single-element.

Now that we've rolled in a GN binary with the change,
reformat all files so that people don't get presubmit
warnings due to this.

CL generated with:
$ git ls-files | grep BUILD.gn | xargs gn format
$ gn format build_overrides/build.gni
$ gn format build_overrides/gtest.gni
$ gn format modules/audio_coding/audio_coding.gni
$ gn format webrtc.gni
$ gn format .gn

Plus a few manual changes to add exceptions for
"public_deps" (after changing these lines the presubmit
started to complain).

[1] - https://gn-review.googlesource.com/c/gn/+/6860

Bug: webrtc:11302
Change-Id: Iac29d23c1618ebef925c972e2891cd9f4e8cd613
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166882
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30334}
2020-01-21 12:13:11 +00:00
Jonas Olsson
b2b2031457 Concatenate string literals at compile time.
This CL was generated by running:
git ls-files | grep ".cc" | xargs perl -i -ne 'BEGIN {undef $/}; s/("[\s\n]*<<[\s\n]*")/" "/g; print;'; git cl format

After that I manually edited modules/audio_processing/gain_controller2.cc to preserve its original
formatting.

This primary benefit of this change is a small reduction in binary size.

Bug: None
Change-Id: I689fa7ba9c717c314bb167e5d592c3c4e0871e29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165961
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30251}
2020-01-14 14:47:48 +00:00
Mirko Bonadei
c7a3b08f07 Prefix ENABLE_RTC_EVENT_LOG with WEBRTC_.
Since this macro can be considered public, it makes sense to prefix it
with WEBRTC_ (also to avoid potential conflicts with client code).

This CL also removes some definitions of this macro in order to define
it only where it is strictly needed (it is only used in a .cc file).

Bug: webrtc:11142
Change-Id: Idce7389301e71d8434e238b3cf4ceaa9cf97cd87
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161008
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29957}
2019-11-29 09:45:50 +00:00
Danil Chapovalov
912b3b83b3 Make rtc::Thread a TaskQueue
in support of converging on single way to run asynchronous tasks in webrtc

Bug: b/144982320
Change-Id: I200ad298136d11764a3f5c0547ebcba51aceafa0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158782
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29896}
2019-11-25 11:38:27 +00:00
Sebastian Jansson
60bd1aea3d Detach thread checker in RtcEventLogImpl destructor.
Otherwise we require that the destructor must run on the same thread as
previous calls. This is not necessary since we can assume there's no
other references to the instance when we enter the destructor.

Bug: webrtc:9883
Change-Id: Ia254bce9265979da0e25ba33598edd8f807d7e00
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159704
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29793}
2019-11-13 15:42:19 +00:00
Björn Terelius
a06048a41e Return status instead of CHECKing in event log parser.
This CL adds ParseStatus/ParseStatusOr classes and returns those instead
of CHECKing that the log is well formed. Some refactoring was required.

We also add a allow_incomplete_logs parameter to the parser which by
default is false. Setting it to true will make the parser log a warning
but return success for errors that typically indicate that the log has
been truncated. "Deeper" errors indicating log corruption still return
an error.

Bug: webrtc:11064
Change-Id: Id5bd6e321de07e250662ae3aaa5ef15f48db6d55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158746
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29679}
2019-11-04 12:42:57 +00:00