mollyim/webrtc
1
0
Fork 0
mirror of https://github.com/mollyim/webrtc.git synced 2025-05-16 07:10:38 +01:00
Code Issues Projects Releases Packages Wiki Activity Actions
33393 commits 2 branches 91 tags 415 MiB
Commit graph

51 commits

Author SHA1 Message Date
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
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
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
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
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
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
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
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
Sebastian Jansson
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
Mirko Bonadei
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
Jonas Olsson
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
Sebastian Jansson
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
Bjorn Terelius
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
Piotr (Peter) Slatala
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
Elad Alon
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
Steve Anton
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
Bjorn Terelius
ba661fe11a Remove support for having multiple SSRCs in an RtcEventVideoSendStreamConfig. 
This has been deprecated for a long time. Simulcast streams are now logged as
one RtcEventVideoSendStreamConfig per SSRC instead of one RtcEventVideoSendStreamConfig
containing a group of SSRCs

Bug: webrtc:8111
Change-Id: I4da62a4b2151a841413cde222a5154638dbb2e47
Reviewed-on: https://webrtc-review.googlesource.com/c/113811
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25957}
 2018-12-10 19:59:12 +00:00
Bjorn Terelius
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
Zach Stein
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
Zach Stein
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
Zach Stein
92c4289ada Reland "Add transaction id to CandidatePairEvents." 
This is a reland of c308bdfa45

Original change's description:
> Add transaction id to CandidatePairEvents.
>
> The transaction id is a randomly generated number used to link stun
> requests and responses (https://tools.ietf.org/html/rfc5389#section-6).
> Logging this will help us debug ICE network issues.
>
> Bug: webrtc:9972
> Change-Id: I93167cb119aad99156e8727b6e4eeeff5198f924
> Reviewed-on: https://webrtc-review.googlesource.com/c/109720
> Commit-Queue: Zach Stein <zstein@webrtc.org>
> Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> Reviewed-by: Elad Alon <eladalon@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25848}

TBR=terelius@webrtc.org,qingsi@webrtc.org

Bug: webrtc:9972
Change-Id: I32b55befddfcb8dc98babd0b64e756eaeb9fab09
Reviewed-on: https://webrtc-review.googlesource.com/c/112661
Reviewed-by: Zach Stein <zstein@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25862}
 2018-11-30 21:10:51 +00:00
Erik Språng
286ee0123e Revert "Add transaction id to CandidatePairEvents." 
This reverts commit c308bdfa45.

Reason for revert: The msan bot has been consistently failing since this commit. See eg https://ci.chromium.org/p/webrtc/builders/luci.webrtc.ci/Linux%20MSan/16989

Original change's description:
> Add transaction id to CandidatePairEvents.
> 
> The transaction id is a randomly generated number used to link stun
> requests and responses (https://tools.ietf.org/html/rfc5389#section-6).
> Logging this will help us debug ICE network issues.
> 
> Bug: webrtc:9972
> Change-Id: I93167cb119aad99156e8727b6e4eeeff5198f924
> Reviewed-on: https://webrtc-review.googlesource.com/c/109720
> Commit-Queue: Zach Stein <zstein@webrtc.org>
> Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> Reviewed-by: Elad Alon <eladalon@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25848}

TBR=eladalon@webrtc.org,terelius@webrtc.org,zstein@webrtc.org,qingsi@webrtc.org,jeroendb@webrtc.org

Change-Id: Ib3b0a845f2300f4fcba2061650e17522735f08b3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9972
Reviewed-on: https://webrtc-review.googlesource.com/c/112581
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25852}
 2018-11-30 09:28:25 +00:00
Zach Stein
c308bdfa45 Add transaction id to CandidatePairEvents. 
The transaction id is a randomly generated number used to link stun
requests and responses (https://tools.ietf.org/html/rfc5389#section-6).
Logging this will help us debug ICE network issues.

Bug: webrtc:9972
Change-Id: I93167cb119aad99156e8727b6e4eeeff5198f924
Reviewed-on: https://webrtc-review.googlesource.com/c/109720
Commit-Queue: Zach Stein <zstein@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25848}
 2018-11-29 22:52:35 +00:00
Zach Stein
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
Yves Gerey
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
Bjorn Terelius
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
Bjorn Terelius
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
Elad Alon
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
Elad Alon
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
Elad Alon
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
Bjorn Terelius
d932fba3bc Track padding and header size in log event. 
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}
 2018-10-17 15:52:17 +00:00
Bjorn Terelius
a240daaae9 Change verification of stream configs in RTC event log unittest. 
We're no longer verifying CSRCs or configurations for remb, rtcp mode
and codec since we're planning to drop those fields from the log in an upcoming CL.

Bug: webrtc:8111
Change-Id: I38a7d87b21f8e6d8a791d8e27a0f54c293f3d340
Reviewed-on: https://webrtc-review.googlesource.com/c/106380
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25229}
 2018-10-17 11:46:11 +00:00
Bjorn Terelius
23e48fb5fd Move expectations from eventlog unittests to helper functions. 
Bug: webrtc:8111
Change-Id: I47fd6c1651f2630ebaf2752b471a36b1d4f98769
Reviewed-on: https://webrtc-review.googlesource.com/c/105482
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25147}
 2018-10-12 15:12:08 +00:00
Bjorn Terelius
55d1af1475 Remove support for microsecond resolution in RtcEventLogs. 
Microsecond resolution is system dependent anyway, so it wasn't reliable.
This CL verifies millisecond timestamps instead of microsecond in tests.

Bug: webrtc:8111
Change-Id: I14aab9a807f747a88b2b84f51becf54f4097931e
Reviewed-on: https://webrtc-review.googlesource.com/c/105561
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25138}
 2018-10-12 09:19:21 +00:00
Danil Chapovalov
f7fcaf0885 Use zero octets for rtp packet padding 
RFC3550 Section 4 mention
"Octets designated as padding have the value zero."

Bug: None
Change-Id: Ife4c6226143c79ad7d152bc6099ba1d81f5492dd
Reviewed-on: https://webrtc-review.googlesource.com/c/103983
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25109}
 2018-10-11 10:22:36 +00:00
Johannes Kron
07ba2b9445 Parse two-byte header extensions. 
Bug: webrtc:7990
Change-Id: I967d2065b85d6a2ca938ac0e83035cb92b45a907
Reviewed-on: https://webrtc-review.googlesource.com/98160
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24881}
 2018-09-28 08:32:17 +00:00
Minyue Li
1a80018a3c Avoid wrong parsing of padding length and its use in NetEq simulation. 
Bug: b/113648474, webrtc:9730
Change-Id: Ieff7ab8697f5c8742548897a9b452a20b0bd2e7c
Reviewed-on: https://webrtc-review.googlesource.com/98461
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24703}
 2018-09-12 11:23:03 +00:00
Karl Wiberg
918f50c5d1 Use absl::make_unique and absl::WrapUnique directly 
Instead of going through our wrappers in ptr_util.h.

This CL was generated by the following script:

  git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
  git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
  git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
  git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
  git cl format

Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.

Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
 2018-07-05 10:59:49 +00:00
Bjorn Terelius
546bdeda77 Add return after NOT_REACHED() in eventlog unittest. 
Bug: webrtc:9457
Change-Id: If4728d05d832f72871c25ddce93a72be5089be40
Reviewed-on: https://webrtc-review.googlesource.com/86122
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23763}
 2018-06-28 09:48:30 +00:00
Bjorn Terelius
63c82a78d3 Style fixes in event log unittest. 
Bug: webrtc:8111
Change-Id: Ib3774429109524ff35b9413d672374d6c99a8614
Reviewed-on: https://webrtc-review.googlesource.com/85373
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23740}
 2018-06-26 12:27:55 +00:00
Bjorn Terelius
f4db542a9b Rewrite the RtcEventLog unit test. 
Bug: webrtc:8111
Change-Id: I36780940b54bf500244c6755113153c84c997942
Reviewed-on: https://webrtc-review.googlesource.com/80660
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23736}
 2018-06-26 10:23:29 +00:00
Bjorn Terelius
25f0c206df Log probing bitrate and cluster id as int32 in event log. 
Bug: webrtc:8111
Change-Id: I0eca0b443f27ece6d2473c5287faa84978eee0dd
Reviewed-on: https://webrtc-review.googlesource.com/73800
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23122}
 2018-05-04 14:40:44 +00:00
Bjorn Terelius
c4ca1d3f37 Reland "Create new API for RtcEventLogParser." 
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}
 2018-04-27 14:46:51 +00:00
Björn Terelius
ff61273c01 Revert "Create new API for RtcEventLogParser." 
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}
 2018-04-25 14:23:14 +00:00
Bjorn Terelius
9e336ec0b8 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}
 2018-04-25 09:37:03 +00:00
Elad Alon
15608c6a34 Prevent RtcEventLogTestHelper from crashing 
Prevent RtcEventLogTestHelper from crashing; use ASSERT instead (soft-failure in the gtest suite).

Bug: webrtc:8111
Change-Id: Ibaabdaa64d0778af83e9a49485fb880793091d24
Reviewed-on: https://webrtc-review.googlesource.com/8620
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20254}
 2017-10-11 18:12:57 +00:00
Elad Alon
1d87b0e40f Create RtcEventLogEncoderLegacy 
We're moving to an RtcEventLog interface that accepts std::unique_ptr<EventLog> and stores the event for encoding when encoding becomes necessary, rather than before. This will be useful while we maintain the legacy (current) encoding alongside the new encoding on which we're working.

This CL introduces RtcEventLogEncoderLegacy, which takes provides the encoding currently done by RtcEventLogImpl. After this, we can modify RtcEventLogImpl to use a dynamically chosen encoding, allowing us to easily choose between the current encoding and the new one on which we're working.

BUG=webrtc:8111
TBR=stefan@webrtc.org

Change-Id: I3dde7e222a40a117549a094a59b04219467f490a
Reviewed-on: https://webrtc-review.googlesource.com/1364
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20116}
 2017-10-03 13:51:59 +00:00
Bjorn Terelius
440216fcf3 Split LogRtpHeader and LogRtcpPacket into separate versions for incoming and outgoing packets. 
Change LogIncomingRtcpPacket and LogOutgoingRtcpPacket to take ArrayView<uint8_t>.
Split LogSessionAndReadBack into three functions and create class to share state between them.
Split VerifyRtpEvent into one incoming and one outgoing version.

Originally uploaded as https://codereview.webrtc.org/2997973002/

Bug: webrtc:8111
Change-Id: I22bdc35163bef60bc8293679226b19e41e8f49b3
Reviewed-on: https://webrtc-review.googlesource.com/5020
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20063}
 2017-10-02 08:44:20 +00:00
Mirko Bonadei
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/ 
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
 2017-09-15 05:02:56 +00:00