The test may spawn threads that reference the global clock, so the
code that resets the fake clock, needs to run when those threads
have been stopped.
Bug: webrtc:13947
Change-Id: I1f7dc921b7a8ff8e7e89e03e4397e635b67d77be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258361
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36489}
This is a reland of 43fb16921b
Original change's description:
> Refactor AnalyzerConfig to use Timestamps instead of microseconds.
>
> Add optional offset-to-UTC parameter to output. This allows aligning
> the x-axis in the generated charts to other UTC-based logs.
>
> Bug: b/215140373
> Change-Id: I65bcd295718acbb8c94e363907c1abc458067bfd
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250203
> Reviewed-by: Kristoffer Erlandsson <kerl@google.com>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35992}
Bug: b/215140373
Change-Id: Id2b88cc4b8078a97275d49a617581cbbd02d2c6f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252380
Reviewed-by: Kristoffer Erlandsson <kerl@google.com>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36066}
Add optional offset-to-UTC parameter to output. This allows aligning
the x-axis in the generated charts to other UTC-based logs.
Bug: b/215140373
Change-Id: I65bcd295718acbb8c94e363907c1abc458067bfd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250203
Reviewed-by: Kristoffer Erlandsson <kerl@google.com>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35992}
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}
First CL to try to understand the extent of the cleanup needed in
order to remove -Wno-shadow and follow Chromium on enabling this
diagnostic.
Bug: webrtc:13219
Change-Id: Ie699762da50fe3dbc08b1fd92220962d4b7da86b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233641
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35134}
This is a reland of e6ee8fab7e
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}
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}
(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}
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}
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}
- 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}
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}
This change does not include receive_timestamps for ACKs, because there is 1 problem.
That problem will be resolved in a separate change.
I am getting receive_timestamp errors that have to do with delta compression with optional fields.
Two failure modes that I noticed:
1) the base event does not have the timestamp: it crashes with length validation
# Check failed: base <= MaxUnsignedValueOfBitWidth(params_.value_width_bits()) (1820716 vs. 131071)
2) all events are null, it crashes with assert that X events were expected, but no events were deserialized.
Bug: webrtc:9719
Change-Id: I5d1bbb95dfd15ca7321667aad5e4d89c085e9c06
Reviewed-on: https://webrtc-review.googlesource.com/c/122360
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26668}
Hard-coding default values forces IDs over 14 to be used even
when we offer less than 15 different extensions.
Note that the code relies on MergeRtpHdrExts for making sure
that extension IDs are kept consistent and non-colliding between
different streams (audio/video).
Bug: webrtc:10288
Change-Id: I3e59f7ddc8ca43cea91084a6b7f36df70fb6be4a
Reviewed-on: https://webrtc-review.googlesource.com/c/121646
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26622}
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 was previously only done for the legacy wire format.
Bug: webrtc:8111
Change-Id: I82767c8eafb35a50967c6ff3d0131c3981957c1f
Reviewed-on: https://webrtc-review.googlesource.com/c/112590
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25909}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script passing top level directories except rtc_base and api
find $@ -type f \( -name \*.h -o -name \*.cc -o -name \*.mm \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I9465c172e65ba6e6ed4e4fdc35b0b265038d6f71
Reviewed-on: https://webrtc-review.googlesource.com/84584
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23697}
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
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 9e336ec0b8.
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}
