129 Commits

Author SHA1 Message Date
Dor Hen
a154b73097 Comment unused variables in implemented functions 11\n
Bug: webrtc:370878648
Change-Id: Ic31d7744cc8516e4c014bc044fbe2dba9e4d835b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366525
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Dor Hen <dorhen@meta.com>
Cr-Commit-Position: refs/heads/main@{#43328}
2024-10-29 17:25:36 +00:00
Dor Hen
bec7015797 Comment unused variables in implemented functions 9\n
Bug: webrtc:370878648
Change-Id: I2cdc8456c9fe1131fa09f02cdb4ba4ab13beccc2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366740
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Dor Hen <dorhen@meta.com>
Cr-Commit-Position: refs/heads/main@{#43326}
2024-10-29 17:23:30 +00:00
Dor Hen
fe6ed1364b Remove unused parameter from FixedLengthEncodingParameters::ValidParameters
Bug: webrtc:370878648
Change-Id: I0031426ebc7ea9b95d7d322a6637c57cb6344ae9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364506
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43161}
2024-10-03 10:40:56 +00:00
Florent Castelli
8037fc6ffa Migrate absl::optional to std::optional
Bug: webrtc:342905193
No-Try: True
Change-Id: Icc968be43b8830038ea9a1f5f604307220457807
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361021
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42911}
2024-09-02 12:16:47 +00:00
Björn Terelius
ac4e0b6f46 Include-what-you-use logging/rtc_event_log/
Forwarding headers like rtc_event_log2_proto_include.h and test/gtest.h
were omitted.

Presubmit gn checks for existing (implicit) dependencies were disabled.

Bug: webrtc:42226242
Change-Id: Id08ae1b244db1a6f65069775f47deec05191ff89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350923
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42393}
2024-05-28 14:33:25 +00:00
Björn Terelius
0596503938 Split BandwidthUsage from network_state_predictor.h
Bug: None
Change-Id: Ie1d0c1fd4b32fc5f9e4252bfe0fd2ca1412bd594
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351002
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42376}
2024-05-24 13:44:37 +00:00
Joachim Reiersen
5075cb4a60 Expose AudioLevel as an absl::optional struct in api/rtp_headers.h
Start migrating away from `hasAudioLevel`, `voiceActivity`, `audioLevel` fields in RTPHeaderExtension and switch usages to a more modern absl::optional<AudioLevel> accessor instead.

The old fields are preserved for compatibility with downstream projects, but will be removed in the future.

Bug: webrtc:15788
Change-Id: I76599124fd68dd4d449f850df3b9814d6a002f5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336303
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41947}
2024-03-22 10:07:47 +00:00
Markus Handell
97df932ecc Remove multiplex codec.
The feature isn't in use by Google and has proven to contain security
issues. It's time to remove it.

Bug: b/324864439
Change-Id: I80344eb2f2060469d2d69a54dc4519fdd02ab4ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340324
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41808}
2024-02-26 11:26:04 +00:00
Joachim Reiersen
4a97488714 Rename AudioLevel to AudioLevelExtension in rtp_header_extensions.h
To prepare for a new AudioLevel struct to be added to the public WebRTC API, rename the internal RTP extension reader/writer class to AudioLevelExtension. A temporary alias is provided to avoid breaking downstream projects.

Bug: webrtc:15788
Change-Id: Ie231668f25932fd9b539229114128b1d0b949a6e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339887
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41787}
2024-02-22 23:12:52 +00:00
Tommi
2c169aef97 Rename kLocal to kHost and kStun to kSrflx
Bug: none
Change-Id: I92845014024e5780365057e81b613b0882724d1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336740
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41652}
2024-02-01 12:31:08 +00:00
Danil Chapovalov
d071dc1725 Bypass global field trial string in RtcEventLog unittests
Pass explicit field trials for object under test,
Bypass field trial altogether for setting expectations

Bug: webrtc:10335
Change-Id: Id17d70aa2c650bd9a00f4bca0035f37b3b415b76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/337183
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41651}
2024-02-01 11:16:15 +00:00
Danil Chapovalov
14b016fbf9 In RtcEventLogEncoderNewFormat use propagated instead of global field trials
Bug: webrtc:10335
Change-Id: Id407a7bc25375cadccba4cf4ae0c771f22a19a0e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333581
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41644}
2024-01-31 15:23:24 +00:00
Tommi
7cb4ce0079 Remove IceCandidateType::kNumValues
Bug: none
Change-Id: I108a006d6ff00f436c87dc9ee5b7e3c27b7b6c83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336242
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41642}
2024-01-31 13:39:30 +00:00
Tommi
c4dd03dfcb Remove kUnknown as a possible value for IceCandidateType.
Subsequently also tighten IceCandidateType error checking.
The Candidate type in `cricket` should be using something similar
(currently using a string for the type), so I'm making sure that
types that we have already, align with where we'd like to be overall.
Possibly we can move IceCandidateType to where Candidate is defined.

Bug: none
Change-Id: Iffeba7268f2a393e18a5f33249efae46e6e08252
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335980
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41640}
2024-01-31 11:26:53 +00:00
Björn Terelius
efeeba0864 Try removing RTC_PUSH_IGNORING_WUNDEF() around proto includes
Bug: webrtc:15623
Change-Id: Ia184993769f74d51e68a5a536d5fdde26890bcfd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325481
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41058}
2023-11-01 08:21:05 +00:00
qwu16
ae82df718c Add codec name H265 to support H265 in WebRTC
Bug: webrtc:13485
Change-Id: I352b15a65867f0d56fc8e9a9e03081bd3258108e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316283
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40773}
2023-09-20 09:25:32 +00:00
philipel
d1e5dedffe Use DD encoder/decoder in RTC event log encoder/parser.
Bug: webrtc:14801
Change-Id: I7013c42765e81d147bf8284f8c29666e67fdb91f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296765
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39929}
2023-04-24 10:35:22 +00:00
philipel
f3e90cd03a OptionalBlobEncoder encode empty string on no input.
Bug: webrtc:14801
Change-Id: I934849322294ae10f0d7cd405e73bc48892543c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296661
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39502}
2023-03-08 13:11:41 +00:00
philipel
579a7b498c Add OptionalBlobEncoder for RTC event logs.
Bug: webrtc:14801
Change-Id: I7c14597e39b312c26573f034dca444cc1d90e332
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295480
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39449}
2023-03-02 12:02:07 +00:00
Lionel Koenig
612872b29d Add RtcEvent to store when MinimumSetDelay is set on NetEq
To be able to simulate offline some scenario in which the javascript
layer set the minimum base buffer size of neteq, it is required to
record those API calls. This change introduces this.

Bug: webrtc:14763
Change-Id: Ic817913eda60978d6fca3f8e12229aeec505ca25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287122
Auto-Submit: Lionel Koenig <lionelk@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Lionel Koenig <lionelk@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39104}
2023-01-13 17:15:48 +00:00
Markus Handell
f015a12802 RtcEventLogImpl: Add mocked time test.
This change adds mocked time unit tests to RtcEventLogImpl. In
order to simplify test implementation, the Impl ctor was changed
to accept an already created event log encoder. The previous
factory was made public in the Impl interface and relevant
code sites were updated.

Bug: chromium:1288710
Change-Id: Ifbfd899c5a06a3350c7e5fbc3bb7280f67124f2b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290382
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38987}
2023-01-03 14:08:16 +00:00
Björn Terelius
12053ec64a Encode remote link capacity estimates in legacy RTC event log format
Bug: None
Change-Id: I36037d0c654e773d5e1c6e9821031eafea54fe0d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271162
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37749}
2022-08-11 12:57:02 +00:00
Danil Chapovalov
300a230f16 Delete inter arrival jitter rtcp packet as unused
WebRTC doesn't produces such packet and ignores it when receive.

Bug: None
Change-Id: I4af8cb3308cb2422808bdfc420a85fa175085bfb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269181
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37627}
2022-07-27 14:53:05 +00:00
Ali Tofigh
277766f55e adopt absl::string_view in logging/
Bug: webrtc:13579
Change-Id: Ibc5fa7842d52321d61cc4cdd4770635af737ddff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267170
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37533}
2022-07-15 16:08:39 +00:00
Björn Terelius
9a6097046e Add noop stubs for encoding/parsing all RTC event log events.
The actual event definitions will be added in upcoming CLs.

Bug: webrtc:11933
Change-Id: Ie10b08a71aeb12118612b7717a08b6acbc699c4d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249361
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35978}
2022-02-11 15:00:14 +00:00
Björn Terelius
d12a14e21f Add new RTC event log encoding for AudioPlayout and DelayBasedBwe events.
Bug: webrtc:11933
Change-Id: Ia54d973099916c8dba9fedf362f25e46fe5cc541
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246204
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35959}
2022-02-08 23:09:24 +00:00
Björn Terelius
01ef37c08e Reland "Wire up proto-free event log format in encoder and parser."
This is a reland of 46333dbf6211ea197965c30fdbecbeb62bc81e5b

Original change's description:
> Wire up proto-free event log format in encoder and parser.
>
> Encode ALR state events as an example. The ALR state unit tests pass with the new format, but the tests are not enabled in this CL since the other event types aren't encoded yet.
>
> Bug: webrtc:11933
> Change-Id: I3ba22778b55f24e2e2bd7d95bb9b17de29ef899f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234520
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35752}

Bug: webrtc:11933
Change-Id: Ia8b23cfb134b61c9ef02aa21189ecbd239783c40
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/248141
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35762}
2022-01-21 13:14:27 +00:00
Björn Terelius
6053903ab9 Revert "Wire up proto-free event log format in encoder and parser."
This reverts commit 46333dbf6211ea197965c30fdbecbeb62bc81e5b.

Reason for revert: Downstream test broken by changed error message.

Original change's description:
> Wire up proto-free event log format in encoder and parser.
>
> Encode ALR state events as an example. The ALR state unit tests pass with the new format, but the tests are not enabled in this CL since the other event types aren't encoded yet.
>
> Bug: webrtc:11933
> Change-Id: I3ba22778b55f24e2e2bd7d95bb9b17de29ef899f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234520
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35752}

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

Change-Id: I9a21ec3c4f876102da146898b840c740f575e03c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11933
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247901
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35753}
2022-01-20 16:15:28 +00:00
Björn Terelius
46333dbf62 Wire up proto-free event log format in encoder and parser.
Encode ALR state events as an example. The ALR state unit tests pass with the new format, but the tests are not enabled in this CL since the other event types aren't encoded yet.

Bug: webrtc:11933
Change-Id: I3ba22778b55f24e2e2bd7d95bb9b17de29ef899f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234520
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35752}
2022-01-20 15:22:28 +00:00
Byoungchan Lee
c065e739e2 Remove RTC_DISALLOW_COPY_AND_ASSIGN more.
Bug: webrtc:13555, webrtc:13082
Change-Id: I9c07708108da0a26f5e228384fd56cef4d1540b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247300
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: (Daniel.L) Byoungchan Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#35749}
2022-01-20 11:00:18 +00:00
Björn Terelius
c15bced118 Prepare for new event log parser.
Minor clean up of BUILD file.
Add explicit events for begin and end of log.
Add a helper function to populate timestamps.
Add a GroupKey method that will be used for grouping events by for example SSRC in additon to event type.

Bug: webrtc:11933
Change-Id: Ie3c5f5a5582c89805a0273f4b27978f47ed0fb4f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234260
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35725}
2022-01-18 16:13:13 +00:00
Artem Titov
d3251968d1 Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.

Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
2021-11-15 21:44:59 +00:00
Harald Alvestrand
97597c0f51 Remove usage of INFO alias for LS_INFO in log messages
Bug: webrtc:13362
Change-Id: Ifda893861a036a85c045cd366f9eab33c62ebde0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237221
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35310}
2021-11-04 13:46:17 +00:00
Björn Terelius
98952d7f0f Prepare for new encoding of RTC event log numeric fields.
Bug: webrtc:11933
Change-Id: I32e59059ea6166b2fc089d9d19d3ab3829c2190e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228942
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35071}
2021-09-22 21:00:24 +00:00
Danil Chapovalov
707e3d187c Migrate rtc event log from rtc::BitBuffer to BitstreamReader
BitstreamReader allows to write easier to reader parser

Bug: None
Change-Id: I9da88c86ee04be4c0b06e181e409a915ba1a5123
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231232
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34939}
2021-09-07 14:03:27 +00:00
Artem Titov
cfea2182f8 Use backticks not vertical bars to denote variables in comments
Bug: webrtc:12338
Change-Id: I89c8b3a328d04203177522cbdfd9e606fd4bce4c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228246
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34696}
2021-08-10 10:40:03 +00:00
Artem Titov
af9a3c6676 Use backticks not vertical bars to denote variables in comments for /logging
Bug: webrtc:12338
Change-Id: I87d33f201d4acfb26ca1d2da8a52cc188ff2c791
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226948
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34609}
2021-07-30 19:19:34 +00:00
Björn Terelius
ada810aab2 Reland "Deprecate microsecond timestamps in RTC event log."
This is a reland of e6ee8fab7eac915b2b6abc9b71b6d33ad086f3d1

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}

Bug: webrtc:11933
Change-Id: I295be966ee96b50719ceb4690dad7e7ce958dbac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221361
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34321}
2021-06-17 12:08:54 +00:00
Björn Terelius
2fa4774067 Revert "Deprecate microsecond timestamps in RTC event log."
This reverts commit e6ee8fab7eac915b2b6abc9b71b6d33ad086f3d1.

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
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
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
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
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
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
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
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