Googletest recently started replacing the term Test Case by Test Suite.
From now on, the preferred API is TestSuite*; the older TestCase* API
will be slowly deprecated.
This CL moves WebRTC to the new set of APIs.
More info in [1].
This CL has been generated with this script:
declare -A items
items[TYPED_TEST_CASE]=TYPED_TEST_SUITE
items[TYPED_TEST_CASE_P]=TYPED_TEST_SUITE_P
items[REGISTER_TYPED_TEST_CASE_P]=REGISTER_TYPED_TEST_SUITE_P
items[INSTANTIATE_TYPED_TEST_CASE_P]=INSTANTIATE_TYPED_TEST_SUITE_P
items[INSTANTIATE_TEST_CASE_P]=INSTANTIATE_TEST_SUITE_P
for i in "${!items[@]}"
do
git ls-files | xargs sed -i "s/\b$i\b/${items[$i]}/g"
done
git cl format
[1] - https://github.com/google/googletest/blob/master/googletest/docs/primer.md#beware-of-the-nomenclature
Bug: None
Change-Id: I5ae191e3046caf347aeee01554d5743548ab0e3f
Reviewed-on: https://webrtc-review.googlesource.com/c/118701
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26494}
This CL applies clang-tidy's performance-for-range-copy [1] on the
WebRTC codebase.
All changes in this CL are automatically generated by both clang-tidy
and 'git cl format'.
[1] - https://clang.llvm.org/extra/clang-tidy/checks/performance-for-range-copy.html
Bug: webrtc:10215
Change-Id: I7c83290b8866d76129bbec4e24e6701f5014102e
Reviewed-on: https://webrtc-review.googlesource.com/c/120043
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26420}
This is a follow-up to
https://webrtc-review.googlesource.com/c/src/+/106280.
This time the whole code base is covered.
Some files may have not been fixed though, whenever the IWYU tool
was breaking the build.
Bug: webrtc:8311
Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef
Reviewed-on: https://webrtc-review.googlesource.com/c/111965
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25830}
RSID is only useful if we store the RSID header extension.
Since we don't do that at the moment, there is no need to
store RSID in the stream configs.
Bug: webrtc:8111
Change-Id: I978f335d05984346f225c4781a8bfaa228f3f4c8
Reviewed-on: https://webrtc-review.googlesource.com/c/111759
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25763}
When delta encoding, use ToUnsigned() and ToSigned() on
timestamp_ms, since it's a signed type. This is only relevant
for delta-encoding/decoding.
Bug: webrtc:8111
Change-Id: I1fabfcb2be64793c281f5bc0d38a2f8035dd0d18
Reviewed-on: https://webrtc-review.googlesource.com/c/110504
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25602}
Since the number of channels is always greater than 0, smaller
deltas can be accomplished by encoding a sequence of (1, 2, 1)
as if the sequence were (0, 1, 0). This way, wrap around to the
first value is a delta of 1, rahter than a delta of 3.
For simplicity's sake, though at the cost of consistency, we still
encode the base event's number of channels unshifted. We do so
because there are no bits to be gained by doing it otherwise, and
the value there is more likely to be manually inspected, than are
the deltas, so a simpler scheme has merit.
Bug: webrtc:8111
Change-Id: I2d4def67da85c42802fe13cd0494fdd9f2b38f7a
Reviewed-on: https://webrtc-review.googlesource.com/c/110242
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25601}
The new event log format makes use of delta encoding to compress
parts of the log.
Bug: webrtc:8111
Change-Id: I7bec839555323a7537dcec831d4ac1d5eb109932
Reviewed-on: https://webrtc-review.googlesource.com/c/109161
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25584}
Before this CL, when we encoded a sequence with a non-existent
base, we pretended that the delta was 0, and the first delta was
based on that. However, in a sequence where the deltas are small,
but where the first element is big, that would produce
unnecessarily wide deltas. Therefore, we change the behavior in
cases where the base is non-existent, to encode the first existent
value (if any) as a varint; the delta width may then be smaller.
This CL include two piggy-backed changes:
1. Varint encoding/decoding moved to its own file (and an
additional flavor added).
2. The unit tests for delta encoding are further parameterized
with a random seed.
Bug: webrtc:8111
Change-Id: I76fff577c86d019c8334bf74b76bd35db06ff68d
Reviewed-on: https://webrtc-review.googlesource.com/c/107860
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25395}
This is a reland of ece3c228a2cbd1c1b05eee3a7f55dbb6f020acbc
Original change's description:
> Encode RTC event logs in new format.
>
> This CL adds the encoder and wires it up to the event log.
> Parser and unit tests are uploaded in a separate CL.
>
> Bug: webrtc:8111
> Change-Id: I6470003e55c2c4006cd8349a2c4bdc3f9491d869
> Reviewed-on: https://webrtc-review.googlesource.com/c/106708
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25333}
Bug: webrtc:8111
Change-Id: I22eeca36d6b1f7cfa1ac65347571ebe33cecc1fc
Reviewed-on: https://webrtc-review.googlesource.com/c/108082
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25382}
Optional fields are those which only occur sometimes. For example,
the sequence number field in an RTP packet always occurs, but
fields in optional RTP extensions only occur sometimes.
Bug: webrtc:8111
Change-Id: Iff2c35b73530c0a1db68e547b4caf34434aa4ace
Reviewed-on: https://webrtc-review.googlesource.com/c/103362
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25336}
This CL adds the encoder and wires it up to the event log.
Parser and unit tests are uploaded in a separate CL.
Bug: webrtc:8111
Change-Id: I6470003e55c2c4006cd8349a2c4bdc3f9491d869
Reviewed-on: https://webrtc-review.googlesource.com/c/106708
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25333}
Signed deltas can yield a more efficient encoding when the encoded
sequence sometimes moves backwards.
Bug: webrtc:8111
Change-Id: Ib1a50192851214ccc3f2bd7eaf88f4be97e4beb0
Reviewed-on: https://webrtc-review.googlesource.com/c/100423
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25324}
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}
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}
Padding size and header size are not part of the header, but we still
want to log them. Add the values as separate fields to the log events.
Bug: webrtc:8111
Change-Id: I8dfa2ccafe679f96b8911b538a8512b0170bc642
Reviewed-on: https://webrtc-review.googlesource.com/c/106321
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25237}
The Copy() function previously did not copy the logging timestamp.
To be able to use Copy() in this test, we add private copy
constructors for RtcEvents which the Copy() can use to copy
everything including the timestamp.
Also adds missing test for RtcEventAlrState,
RtcEventIceCandidatePairConfig and RtcEventIceCandidatePair.
Bug: webrtc:8111
Change-Id: I3901231735baa4e671173c921eada0a4be6de7c9
Reviewed-on: https://webrtc-review.googlesource.com/86042
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23774}
Disjoint subsets of the enum values are used for Ice candidate config
events and Ice candidate check events. This CL breaks out the config
part to a separate enum and by extension changes the icelogger interface
for config events.
Bug: webrtc:9336, webrtc:8111
Change-Id: I405b5c3981905c3c504b45afdddb3649469ed141
Reviewed-on: https://webrtc-review.googlesource.com/79943
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23464}
The new API stores events gathered by event type. For example, it is
possible to ask for a list of all incoming RTCP messages or all audio
playout events.
The new API is experimental and may change over next few weeks. Once
it has stabilized and all unit tests and existing tools have been
ported to the new API, the old one will be removed.
This CL also updates the event_log_visualizer tool to use the new
parser API. This is not a funcional change except for:
- Incoming and outgoing audio level are now drawn in two separate plots.
- Incoming and outgoing timstamps are now drawn in two separate plots.
- RTCP count is no longer split into Video and Audio. It also counts
all RTCP packets rather than only specific message types.
- Slight timing difference in sendside BWE simulation due to only
iterating over transport feedbacks and not over all RTCP packets.
This timing changes are not visible in the plots.
Media type for RTCP messages might not be identified correctly by
rtc_event_log2text anymore. On the other hand, assigning a specific
media type to an RTCP packet was a bit hacky to begin with.
Bug: webrtc:8111
Change-Id: Ib244338c86a2c1a010c668a7aba440482023b512
Reviewed-on: https://webrtc-review.googlesource.com/73140
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23056}
This reverts commit 9e336ec0b8a77c3461d13677cff3563c11c88daa.
Reason for revert: Code can accidentally include the deprecated parser but link with the new one, or vice versa. Reverting to fix naming.
Original change's description:
> Create new API for RtcEventLogParser.
>
> The new API stores events gathered by event type. For example, it is
> possible to ask fo a list of all incoming RTCP messages or all audio
> playout events.
>
> The new API is experimental and may change over next few weeks. Once
> it has stabilized and all unit tests and existing tools have been
> ported to the new API, the old one will be removed.
>
> This CL also updates the event_log_visualizer tool to use the new
> parser API. This is not a funcional change except for:
> - Incoming and outgoing audio level are now drawn in two separate plots.
> - Incoming and outgoing timstamps are now drawn in two separate plots.
> - RTCP count is no longer split into Video and Audio. It also counts
> all RTCP packets rather than only specific message types.
> - Slight timing difference in sendside BWE simulation due to only
> iterating over transport feedbacks and not over all RTCP packets.
> This timing changes are not visible in the plots.
>
>
> Media type for RTCP messages might not be identified correctly by
> rtc_event_log2text anymore. On the other hand, assigning a specific
> media type to an RTCP packet was a bit hacky to begin with.
>
> Bug: webrtc:8111
> Change-Id: I8e7168302beb69b2e163a097a2a142b86dd4a26b
> Reviewed-on: https://webrtc-review.googlesource.com/60865
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23015}
TBR=terelius@webrtc.org,srte@webrtc.org,minyue@webrtc.org
Change-Id: Ib4bbcf0563423675a3cc1dce59ebf665e0c5dae9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8111
Reviewed-on: https://webrtc-review.googlesource.com/72500
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23026}
The new API stores events gathered by event type. For example, it is
possible to ask fo a list of all incoming RTCP messages or all audio
playout events.
The new API is experimental and may change over next few weeks. Once
it has stabilized and all unit tests and existing tools have been
ported to the new API, the old one will be removed.
This CL also updates the event_log_visualizer tool to use the new
parser API. This is not a funcional change except for:
- Incoming and outgoing audio level are now drawn in two separate plots.
- Incoming and outgoing timstamps are now drawn in two separate plots.
- RTCP count is no longer split into Video and Audio. It also counts
all RTCP packets rather than only specific message types.
- Slight timing difference in sendside BWE simulation due to only
iterating over transport feedbacks and not over all RTCP packets.
This timing changes are not visible in the plots.
Media type for RTCP messages might not be identified correctly by
rtc_event_log2text anymore. On the other hand, assigning a specific
media type to an RTCP packet was a bit hacky to begin with.
Bug: webrtc:8111
Change-Id: I8e7168302beb69b2e163a097a2a142b86dd4a26b
Reviewed-on: https://webrtc-review.googlesource.com/60865
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23015}
more standard optional<T> inlines compares instead of converting second argument to T.
that leads to warnings about comparing unsigned to signed integers.
Bug: webrtc:9078
Change-Id: I43cc729d3b85d789b0c394064dc7e11dc27a37aa
Reviewed-on: https://webrtc-review.googlesource.com/66782
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22779}
This is a reland of eed5aa8904d09179971d3f4e7e10c109d7c62bfc
Original change's description:
> Structured ICE logging via RtcEventLog.
>
> This change list contains the structured logging module for ICE using
> the RtcEventLog infrastructure, and also extension to the log parser
> and analyzer.
>
> Bug: None
> Change-Id: I6539cf282155c2cde4d3161c53500c0746671a02
> Reviewed-on: https://webrtc-review.googlesource.com/34622
> Commit-Queue: Qingsi Wang <qingsi@google.com>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21816}
TBR=pthatcher@webrtc.org,terelius@webrtc.org,deadbeef@webrtc.org
Bug: None
Change-Id: I3df585bf636315ceb0273967146111346a83be86
Reviewed-on: https://webrtc-review.googlesource.com/47545
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21881}
This change list contains the structured logging module for ICE using
the RtcEventLog infrastructure, and also extension to the log parser and
analyzer.
Bug: None
Change-Id: I6539cf282155c2cde4d3161c53500c0746671a02
Reviewed-on: https://webrtc-review.googlesource.com/34622
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21816}
Use EncodeBatch method in unittest. (Same as in production code.)
Bug: webrtc:8111
Change-Id: Ia194f5138f244da7f348821277f6c712a3ffab0d
Reviewed-on: https://webrtc-review.googlesource.com/34560
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21696}
This prevents the programmer from accidentally adding LOG_START and LOG_END events to the log without actually starting the log. This also makes it easier to ensure that the LOG_START event always ends up first and the LOG_END event always ends up last in the log file.
Bug: webrtc:8111
Change-Id: I4e6c9306f8559ff184b5185f8728409f8dcebfa0
Reviewed-on: https://webrtc-review.googlesource.com/34400
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21486}
