Commit Graph

84 Commits

Author SHA1 Message Date
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
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
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
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
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
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
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
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
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
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
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
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
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
0a5ed896e2 Adds remote estimates to rtc event log.
Bug: webrtc:10742
Change-Id: I0db998a05492603fcdeedca780d9ee3d64aa00d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151651
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29233}
2019-09-19 09:22:37 +00:00
317a1f09ed Use std::make_unique instead of absl::make_unique.
WebRTC is now using C++14 so there is no need to use the Abseil version
of std::make_unique.

This CL has been created with the following steps:

git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt
git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt
git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt

diff --new-line-format="" --unchanged-line-format="" \
  /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \
  uniq > /tmp/only_make_unique.txt
diff --new-line-format="" --unchanged-line-format="" \
  /tmp/only_make_unique.txt /tmp/memory.txt | \
  xargs grep -l "absl/memory" > /tmp/add-memory.txt

git grep -l "\babsl::make_unique\b" | \
  xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g"

git checkout PRESUBMIT.py abseil-in-webrtc.md

cat /tmp/add-memory.txt | \
  xargs sed -i \
  's/#include "absl\/memory\/memory.h"/#include <memory>/g'
git cl format
# Manual fix order of the new inserted #include <memory>

cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \
  xargs sed -i '/#include "absl\/memory\/memory.h"/d'

git ls-files | grep BUILD.gn | \
  xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d'

python tools_webrtc/gn_check_autofix.py \
  -m tryserver.webrtc -b linux_rel

# Repead the gn_check_autofix step for other platforms

git ls-files | grep BUILD.gn | \
  xargs sed -i 's/absl\/memory:memory/absl\/memory/g'
git cl format

Bug: webrtc:10945
Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 15:47:29 +00:00
16cb1f61c0 Stop using rtc_event.h forward header
Bug: webrtc:10206
Change-Id: I16905ec745673178195d6715fda6175c31500163
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151601
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29149}
2019-09-11 08:20:29 +00:00
a4d873786f Format almost everything.
This CL was generated by running

git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \
grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \
grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \
grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \
grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \
| xargs clang-format -i ; git cl format

Most of these changes are clang-format grouping and reordering includes
differently.

Bug: webrtc:9340
Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28505}
2019-07-08 13:45:15 +00:00
166b45db26 Adds route changes in event logs.
Bug: webrtc:10614
Change-Id: Ifd859c977fc66cb606914ddb38a3fb3618e3ad90
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135952
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27924}
2019-05-13 10:41:40 +00:00
40889f35fc Removes TimeMicros interface from ThreadProcessingFakeClock.
Bug: webrtc:9883
Change-Id: Ib48872f81f734b09e3ffa4d9d26da79177b02303
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133341
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27668}
2019-04-17 15:37:48 +00:00
6a489f22c7 Fully qualify googletest symbols.
Semi-automatically created with:

git grep -l " testing::" | xargs sed -i "s/ testing::/ ::testing::/g"
git grep -l "(testing::" | xargs sed -i "s/(testing::/(::testing::/g"
git cl format

After this, two .cc files failed to compile and I have fixed them
manually.

Bug: webrtc:10523
Change-Id: I4741d3bcedc831b6c5fdc04485678617eb4ce031
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132018
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27526}
2019-04-09 17:18:20 +00:00
f1c9e21366 ClangTidy fixes for logging/
These are manual edits please verify there are no typos.
Feel free to auto-submit if there are no issues.

Bug: webrtc:10410
Change-Id: I2ea59dc66230182bee6ae7a0925aed0fe9ef823c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127643
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27133}
2019-03-14 14:51:00 +00:00
aec9794120 Fix DCHECK when encoding GenericPacket* events using the legacy RTC event log format.
Bug: webrtc:10319
Change-Id: I6764072281b8d66a8e4a5b2c0e547af3bfddda37
Reviewed-on: https://webrtc-review.googlesource.com/c/123121
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26693}
2019-02-14 18:11:19 +00:00
da825b1b12 Replace NOTREACHED with a break.
The old format is still used.

Bug: None
Change-Id: I2c8c9a8c7b09282ba8c375403cbee7fd7ab2d056
Reviewed-on: https://webrtc-review.googlesource.com/c/122960
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26679}
2019-02-14 09:59:49 +00:00
dfd5c4b15e Parse XR, FIR and PLI in rtc_event_log_parser.cc
Bug: webrtc:10312
Change-Id: I1b24e23f8002feef8a2ef928130ac6da19c3cd81
Reviewed-on: https://webrtc-review.googlesource.com/c/122580
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26670}
2019-02-13 19:11:42 +00:00
5e2aad1c95 Support GenericPacketReceived/Sent/AckReceived event logs.
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}
2019-02-13 17:06:22 +00:00
f6adac87b4 Add rtc event generic packet sent and received.
Bug: webrtc:9719
Change-Id: I2f692d9c1b33ac390975a9e695c7652cdc1b1e6e
Reviewed-on: https://webrtc-review.googlesource.com/c/121680
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26616}
2019-02-08 19:13:57 +00:00
c84f661b10 Stop using Googletest legacy APIs.
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}
2019-01-31 13:23:33 +00:00
733e087e63 Ignore duplicated incoming RTCP packets in RTC event log parser.
Bug: webrtc:8111
Change-Id: I1082ff66cac9c3744811713d686b3d7f85bd7584
Reviewed-on: https://webrtc-review.googlesource.com/c/120200
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26430}
2019-01-28 20:38:38 +00:00
739baf097b [clang-tidy] Apply performance-for-range-copy fixes.
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}
2019-01-28 09:53:50 +00:00
f380284035 (7) Rename files to snake_case: remove forwarding headers
Bug: webrtc:10159
Change-Id: I2ba899e0283b953538c7941c8790213e36d7c70e
Reviewed-on: https://webrtc-review.googlesource.com/c/118561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26417}
2019-01-26 00:33:46 +00:00
84ca69ad6e Add RTC event logging of LossNotification RTCP messages
Bug: webrtc:10226
Change-Id: Ib65970a8f13cd64529f3101993d40887168e313e
Reviewed-on: https://webrtc-review.googlesource.com/c/118933
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26391}
2019-01-24 13:33:57 +00:00
aec15aa810 (5) Rename files to snake_case: install forwarding headers
Mechanically generated with this command:

tools_webrtc/do-renames.sh install all-renames.txt && git cl format

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: Ic8e99f71f2da62e5c99863c6d24a8cfe311466cd
Reviewed-on: https://webrtc-review.googlesource.com/c/115682
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26227}
2019-01-11 17:13:36 +00:00
10542f21c8 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
Mechanically generated by running this command:

tools_webrtc/do-renames.sh update all-renames.txt && git cl format

Then manually updating:

tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 17:11:39 +00:00
1c05765831 (3) Rename files to snake_case: move the files
Mechanically generated with this command:

tools_webrtc/do-rename.sh move all-renames.txt

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I8b05b6eab9b9d18b29c2199bbea239e9add1e690
Reviewed-on: https://webrtc-review.googlesource.com/c/115481
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26225}
2019-01-11 17:05:20 +00:00
b290a6d767 Renames RtcEventLogParseNew to RtcEventLogParser
Bug: webrtc:10170
Change-Id: I9232c276229a64fa4d8321b6c996387fe130f68b
Reviewed-on: https://webrtc-review.googlesource.com/c/116064
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26128}
2019-01-03 19:39:04 +00:00
ebb677773a Allow access to specific RTCP types in new format event log parser.
Bug: webrtc:8111
Change-Id: Ia727c8fd9ddaa9f0c17fea5af8f544ad01e66367
Reviewed-on: https://webrtc-review.googlesource.com/c/112382
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25886}
2018-12-04 10:49:21 +00:00
b3033c448f Add transaction id to candidate pair event log parser and encoder.
Covered by these tests:
RandomSeeds/RtcEventLogEncoderTest.RtcEventIceCandidatePair/*
RtcEventLogTest/RtcEventLogSession.*

Bug: webrtc:9972
Change-Id: I05473176357804e7ad0dedb51a659ab9481a4e4a
Reviewed-on: https://webrtc-review.googlesource.com/c/110103
Commit-Queue: Zach Stein <zstein@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25867}
2018-12-01 01:15:29 +00:00
846dfdfd01 Log DTLS writable changes to RtcEventLog
Bug: webrtc:9972
Change-Id: I8d29dcbc200bd0aad7f4c2600459c4d344c61bbb
Reviewed-on: https://webrtc-review.googlesource.com/c/111453
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Zach Stein <zstein@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25866}
2018-12-01 01:00:28 +00:00
c64078fdc0 Log DTLS state changes to RtcEventLog
Bug: webrtc:9972
Change-Id: Icb2cdda87c1c42607ed06653e053699532f25e84
Reviewed-on: https://webrtc-review.googlesource.com/c/111023
Commit-Queue: Zach Stein <zstein@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25832}
2018-11-28 19:49:29 +00:00
3e70781361 [Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
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}
2018-11-28 18:25:07 +00:00
2c977b4cc5 Remove RSID from stream configs in new event log format.
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}
2018-11-22 17:54:06 +00:00
e78b465d45 Add version and UTC time fields to RTC event log.
Bug: webrtc:8111
Change-Id: I843e1f8e50e4d4faf74819e0104aeb4fb869b834
Reviewed-on: https://webrtc-review.googlesource.com/c/110721
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25658}
2018-11-15 14:32:43 +00:00
c42d62495c Event log - Use ToUnsigned() and ToSigned() on timestamp_ms
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}
2018-11-12 14:27:38 +00:00
19084f89d2 Event logs - encode N channels as N-1
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}
2018-11-12 14:05:11 +00:00
f4a3f9cc25 Add RtcEvent::timestamp_ms()
Bug: webrtc:8111
Change-Id: I0ec7eda2b2afcd945625fb9f5d592e73a97992e3
Reviewed-on: https://webrtc-review.googlesource.com/c/109861
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25597}
2018-11-12 13:18:07 +00:00
0b1b5c1b2a Hide RtcEvent members behind accessors
Bug: webrtc:8111
Change-Id: I3d350a6e159330aed7362162006860ac86ed7c32
Reviewed-on: https://webrtc-review.googlesource.com/c/109881
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25590}
2018-11-10 23:34:07 +00:00
eb809f30d1 Event logs - separate audio_level and voice_activity
Bug: webrtc:8111
Change-Id: I44d81c5b4f5b854e8accd84521fbbd7b50228903
Reviewed-on: https://webrtc-review.googlesource.com/c/109571
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25589}
2018-11-10 01:41:28 +00:00
56a4b32398 Rename fields in rtc_event_log2.proto
1. s/deltas_ms/ms-deltas
2. s/deltas_bps/bps_deltas
3. s/raw_packet_deltas/raw_packet_blobs

Bug: webrtc:8111
Change-Id: Ib2f7457275e0b930a6aa73d628a707676c74a2a4
Reviewed-on: https://webrtc-review.googlesource.com/c/109142
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25587}
2018-11-09 22:29:43 +00:00
d95b0a2fbd Use delta-encoding in new WebRTC event logs
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}
2018-11-09 16:39:16 +00:00
ff43541927 Delta compression efficiency improvement for non-existent base
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}
2018-10-26 18:17:40 +00:00