3a034e15b4
Split DataChannel into two separate classes for RTP and SCTP.
...
Done in preparation for some threading changes that would be quite
messy if implemented with the class as-is.
This results in some code duplication, but is preferable to
one class having two completely different modes of operation.
RTP data channels are in the process of being removed anyway,
so the duplicated code won't last forever.
Bug: webrtc:9883
Change-Id: Idfd41a669b56a4bb4819572e4a264a4ffaaba9c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178940
Commit-Queue: Taylor <deadbeef@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#31691}
2020-07-10 00:03:21 +00:00
a0ff50c031
Reland "Improve outbound-rtp statistics for simulcast"
...
This reverts commit 9a925c9ce33a6ccdd11b545b11ba68e985c2a65d.
Reason for revert: The original CL is updated in PS #2 to
fix the googRtt issue which was that when the legacy sender
stats were put in "aggregated_senders" we forgot to update
rtt_ms the same way that we do it for "senders".
Original change's description:
> Revert "Improve outbound-rtp statistics for simulcast"
>
> This reverts commit da6cda839dac7d9d18eba8d365188fa94831e0b1.
>
> Reason for revert: Breaks googRtt in legacy getStats API
>
> Original change's description:
> > Improve outbound-rtp statistics for simulcast
> >
> > Bug: webrtc:9547
> > Change-Id: Iec4eb976aa11ee743805425bedb77dcea7c2c9be
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168120
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org >
> > Reviewed-by: Erik Språng <sprang@webrtc.org >
> > Reviewed-by: Henrik Boström <hbos@webrtc.org >
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org >
> > Commit-Queue: Eldar Rello <elrello@microsoft.com >
> > Cr-Commit-Position: refs/heads/master@{#31097}
>
> TBR=hbos@webrtc.org ,sprang@webrtc.org ,stefan@webrtc.org ,srte@webrtc.org ,hta@webrtc.org ,elrello@microsoft.com
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:9547
> Change-Id: I06673328c2a5293a7eef03b3aaf2ded9d13df1b3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174443
> Reviewed-by: Henrik Boström <hbos@webrtc.org >
> Commit-Queue: Henrik Boström <hbos@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#31165}
TBR=hbos@webrtc.org ,sprang@webrtc.org ,stefan@webrtc.org ,srte@webrtc.org ,hta@webrtc.org ,elrello@microsoft.com
# Not skipping CQ checks because this is a reland.
Bug: webrtc:9547
Change-Id: I723744c496c3c65f95ab6a8940862c8b9f544338
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174480
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#31169}
2020-05-05 20:22:19 +00:00
9a925c9ce3
Revert "Improve outbound-rtp statistics for simulcast"
...
This reverts commit da6cda839dac7d9d18eba8d365188fa94831e0b1.
Reason for revert: Breaks googRtt in legacy getStats API
Original change's description:
> Improve outbound-rtp statistics for simulcast
>
> Bug: webrtc:9547
> Change-Id: Iec4eb976aa11ee743805425bedb77dcea7c2c9be
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168120
> Reviewed-by: Sebastian Jansson <srte@webrtc.org >
> Reviewed-by: Erik Språng <sprang@webrtc.org >
> Reviewed-by: Henrik Boström <hbos@webrtc.org >
> Reviewed-by: Harald Alvestrand <hta@webrtc.org >
> Commit-Queue: Eldar Rello <elrello@microsoft.com >
> Cr-Commit-Position: refs/heads/master@{#31097}
TBR=hbos@webrtc.org ,sprang@webrtc.org ,stefan@webrtc.org ,srte@webrtc.org ,hta@webrtc.org ,elrello@microsoft.com
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9547
Change-Id: I06673328c2a5293a7eef03b3aaf2ded9d13df1b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174443
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#31165}
2020-05-05 13:38:51 +00:00
da6cda839d
Improve outbound-rtp statistics for simulcast
...
Bug: webrtc:9547
Change-Id: Iec4eb976aa11ee743805425bedb77dcea7c2c9be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168120
Reviewed-by: Sebastian Jansson <srte@webrtc.org >
Reviewed-by: Erik Språng <sprang@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Eldar Rello <elrello@microsoft.com >
Cr-Commit-Position: refs/heads/master@{#31097}
2020-04-17 11:28:00 +00:00
b33a0ca1ee
Remove deprecated ssl_identity methods
...
This is a followup to
https://webrtc-review.googlesource.com/c/src/+/170637
Bug: webrtc:11450
Change-Id: I69928ed7236c6a8a569c7dc0383f7debb4408179
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171224
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#31086}
2020-04-16 14:21:41 +00:00
ac0a4cbbd8
Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received"
...
This is a reland of fbde32e596f06893d6dda13eb7d29f4c251cf08b
The chromium problem should be fixed with
https://chromium-review.googlesource.com/c/chromium/src/+/1862437
Original change's description:
> Fix GetStats bytesSent/Received, wireup headerBytesSent/Received
>
> Changes the standard GetStats, legacy GetStats unchanged.
>
> Bug: webrtc:10525
> Change-Id: Ie10fe8079f1d8b4cc6bbe513f6a2fc91477b5441
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156084
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
> Reviewed-by: Henrik Boström <hbos@webrtc.org >
> Reviewed-by: Harald Alvestrand <hta@webrtc.org >
> Commit-Queue: Niels Moller <nisse@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#29462}
Tbr: kwiberg@webrtc.org
Bug: webrtc:10525
Change-Id: I3b61f9535aa3f1fca2ed84f068233803d4ec9fe2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157045
Reviewed-by: Niels Moller <nisse@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29485}
2019-10-15 10:43:59 +00:00
ef0627fb50
Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received"
...
This reverts commit fbde32e596f06893d6dda13eb7d29f4c251cf08b.
Reason for revert: It seems to break WebRTC FYI tests in Chromium.
https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Linux%20Tester/4763
Original change's description:
> Fix GetStats bytesSent/Received, wireup headerBytesSent/Received
>
> Changes the standard GetStats, legacy GetStats unchanged.
>
> Bug: webrtc:10525
> Change-Id: Ie10fe8079f1d8b4cc6bbe513f6a2fc91477b5441
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156084
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
> Reviewed-by: Henrik Boström <hbos@webrtc.org >
> Reviewed-by: Harald Alvestrand <hta@webrtc.org >
> Commit-Queue: Niels Moller <nisse@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#29462}
TBR=kwiberg@webrtc.org ,hbos@webrtc.org ,nisse@webrtc.org ,hta@webrtc.org
Change-Id: I6a983ea4d5ff38e49f096a8ff5cd9b426768f955
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10525
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157043
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29478}
2019-10-15 08:55:06 +00:00
fbde32e596
Fix GetStats bytesSent/Received, wireup headerBytesSent/Received
...
Changes the standard GetStats, legacy GetStats unchanged.
Bug: webrtc:10525
Change-Id: Ie10fe8079f1d8b4cc6bbe513f6a2fc91477b5441
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156084
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29462}
2019-10-14 13:07:13 +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
149dc72dfa
Add support for RTCTransportStats.selectedCandidatePairChanges
...
This patch adds accounting and reporting needed for
newly added RTCTransportStats.selectedCandidatePairChanges,
https://w3c.github.io/webrtc-stats/#dom-rtctransportstats-selectedcandidatepairchanges
a) P2PTransportChannel counts everytime selected_connection_
is modified and reports this counter in the GetStats()-call.
b) RTCStatsCollector puts the counter into the standardized
stats object.
Bug: webrtc:10900
Change-Id: Ibaeca18706b8edcbcb44b0c6f2754854bcb545ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149830
Reviewed-by: Qingsi Wang <qingsi@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Jonas Oreland <jonaso@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#28987}
2019-08-28 13:22:08 +00:00
224c69d527
Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo
...
It's propagated from ReceiveStatistics up to VoiceReceiverInfo,
and then not used. It's not part of the standard stats.
Bug: None
Change-Id: I90ce6a72e3ca846adbbba5d3023fef18a2169018
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149164
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#28933}
2019-08-22 07:23:04 +00:00
5b5d97c938
Reland of "Reporting of decoding_codec_plc events""
...
This is a reland of 0a88ea050cda58de81d624cf2764d46929447ed5.
The new stat will not be reported unless it is GT 0.
Reporting of decoding_codec_plc events
Bug: webrtc:10838
Change-Id: Ic8585b4eeae9a2643374f15bc2578d1141e59683
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148448
Reviewed-by: Magnus Flodman <mflodman@webrtc.org >
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org >
Commit-Queue: Alex Narest <alexnarest@google.com >
Cr-Commit-Position: refs/heads/master@{#28797}
2019-08-07 18:41:46 +00:00
bedb7a8aea
Revert "Reporting of decoding_codec_plc events"
...
This reverts commit 0a88ea050cda58de81d624cf2764d46929447ed5.
Reason for revert: This CL breaks Chromium's FYI bots (example: https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Linux%20Tester/4033 ).
Original change's description:
> Reporting of decoding_codec_plc events
>
> Change-Id: Id71b37244bc98bffaf25131a519127b3d2b86a8f
>
> Bug: webrtc:10838
> Change-Id: Id71b37244bc98bffaf25131a519127b3d2b86a8f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147263
> Commit-Queue: Alex Narest <alexnarest@google.com >
> Reviewed-by: Magnus Flodman <mflodman@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#28700}
TBR=mflodman@webrtc.org ,alexnarest@google.com
Change-Id: I5e5dd29ee375ba422f79932d4b8c3fd028a53db4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10838
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147269
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#28707}
2019-07-30 14:39:09 +00:00
0a88ea050c
Reporting of decoding_codec_plc events
...
Change-Id: Id71b37244bc98bffaf25131a519127b3d2b86a8f
Bug: webrtc:10838
Change-Id: Id71b37244bc98bffaf25131a519127b3d2b86a8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147263
Commit-Queue: Alex Narest <alexnarest@google.com >
Reviewed-by: Magnus Flodman <mflodman@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#28700}
2019-07-29 16:40:23 +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
3472b9ae22
Delete RTCInboundRTPStreamStats::fraction_lost
...
And delete corresponding plumbing via the internal stats attribute
MediaReceiverInfo::fraction_lost. The latter attribute is not deleted
yet, since downstream projects have to be updated first.
Bug: webrtc:10744
Change-Id: Id5401aeee7e5637a406ddf2fa33fbfe336abec9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143178
Commit-Queue: Niels Moller <nisse@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Oskar Sundbom <ossu@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#28385}
2019-06-26 11:43:23 +00:00
fc02a793c2
Revert "Piping audio interruption metrics to API layer"
...
This reverts commit 299c4e68461f1c4428b2a919913b161115025dff.
Reason for revert: https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Win10%20Tester/2753
../../chrome/browser/media/webrtc/webrtc_browsertest_base.cc(539): error: Expected equality of these values:
"ok-got-stats"
ExecuteJavascript("verifyLegacyStatsGenerated()", tab)
Which is: "Test failed: Error: \"googInterruptionCount\" is not a whitelisted stat. Exposing new metrics in the legacy getStats() API is not allowed. Please follow the standardization process: https://docs.google.com/document/d/1q1CJVUqJ6YW9NNRc0tENkLNny8AHrKZfqjy3SL89zjc/edit?usp=sharing\n at failTest (http://127.0.0.1:50650/webrtc/test_functions.js:46:15 )\n at http://127.0.0.1:50650/webrtc/peerconnection.js:481:19 "
With diff:
@@ -1,1 +1,3 @@
-ok-got-stats
+Test failed: Error: \"googInterruptionCount\" is not a whitelisted stat. Exposing new metrics in the legacy getStats() API is not allowed. Please follow the standardization process: https://docs.google.com/document/d/1q1CJVUqJ6YW9NNRc0tENkLNny8AHrKZfqjy3SL89zjc/edit?usp=sharing
+ at failTest (http://127.0.0.1:50650/webrtc/test_functions.js:46:15 )
+ at http://127.0.0.1:50650/webrtc/peerconnection.js:481:19
Original change's description:
> Piping audio interruption metrics to API layer
>
> Bug: webrtc:10549
> Change-Id: Ie6abe5819c5b73dc5f5f89bdc375bad77f44ce97
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134303
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#27788}
TBR=henrik.lundin@webrtc.org ,kwiberg@webrtc.org ,ivoc@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10549
Change-Id: I345306ba9758c0a3b1597724fa860d3e3fdb8995
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134464
Commit-Queue: Henrik Andreassson <henrika@webrtc.org >
Reviewed-by: Henrik Andreassson <henrika@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#27802}
2019-04-29 11:23:16 +00:00
299c4e6846
Piping audio interruption metrics to API layer
...
Bug: webrtc:10549
Change-Id: Ie6abe5819c5b73dc5f5f89bdc375bad77f44ce97
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134303
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#27788}
2019-04-26 13:32:34 +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
efe4c92d54
Use RtpSender/RtpReceiver track ID for legacy GetStats
...
Previously, legacy GetStats would look up the track ID by querying the
local/remote SDP by SSRC. This doesn't work with Unified Plan since the
RtpSender/RtpReceiver track IDs may not correspond to the track ID
stored in the SDP.
This CL changes legacy GetStats to pull the track ID directly from the
RtpSenders and RtpReceivers as it generates the stats. This has a few
additional benefits:
1) Unsignaled receive SSRC stats should now get correctly matched to
the unsigneled RtpReceiver track ID for both Plan B and Unified
Plan.
2) Removes a couple methods on PeerConnection that were only used by
the legacy StatsCollector.
3) Keeps the SSRC -> track ID mapping more localized which should make
the code easier to understand.
Bug: chromium:943493
Change-Id: I43ecde8c3a3d1c5f9c749ba6c8dfb11e8c4950fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129782
Commit-Queue: Steve Anton <steveanton@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Amit Hilbuch <amithi@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#27324}
2019-03-27 18:14:00 +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
64b626b03f
Use Abseil container algorithms in pc/
...
Bug: None
Change-Id: If784461b54d95bdc6f8a7d4e5d1bbfa52d1a390e
Reviewed-on: https://webrtc-review.googlesource.com/c/119862
Commit-Queue: Steve Anton <steveanton@webrtc.org >
Reviewed-by: Amit Hilbuch <amithi@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26433}
2019-01-29 02:33:50 +00:00
d970807e0c
Remove rtc_base/scoped_ref_ptr.h.
...
The type rtc::scoped_refptr<T> is now part of api/. Please include it from
api/scoped_refptr.h.
More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o .
Bug: webrtc:9887, webrtc:8205
No-Try: True
Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf
Reviewed-on: https://webrtc-review.googlesource.com/c/119041
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:29:58 +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
