Commit graph

152 commits

Author SHA1 Message Date
Jakob Ivarsson
e54914a79e Implement nack_count metric for inbound audio rtp streams.
Bug: webrtc:12925
Change-Id: I4542ca0f14a7dd7485ad5a2b6f2bd7051076f71f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224085
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34401}
2021-07-01 10:38:44 +00:00
Jakob Ivarsson
47a03e8743 Default enable sending transport sequence numbers on audio packets.
This enables send side bandwidth estimation for audio and removes field
trial "WebRTC-Audio-SendSideBwe" which this was controlled through.

Transport-cc extension still needs to be negotiated.

Bug: webrtc:12222
Change-Id: Ie2268fad13703eeb0f0d38fcf484baaa29715b7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194142
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32681}
2020-11-24 09:19:54 +00:00
Mirko Bonadei
8cc6695652 Reformat python files checked by pylint (part 1/2).
After recently changing .pylintrc (see [1]) we discovered that
the presubmit check always checks all the python files when just
one python file gets updated.

This CL moves all these files one step closer to what the linter
wants.

Autogenerated with:

# Added all the files under pylint control to ~/Desktop/to-reformat
cat ~/Desktop/to-reformat | xargs sed -i '1i\\'
git cl format --python --full

This is part 1 out of 2. The second part will fix function names and
will not be automated.

[1] - https://webrtc-review.googlesource.com/c/src/+/186664

No-Presubmit: True
Bug: webrtc:12114
Change-Id: Idfec4d759f209a2090440d0af2413a1ddc01b841
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190980
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32530}
2020-10-30 10:13:11 +00:00
Niels Möller
6b4d962947 Fix standard GetStats to not modify NetEq state.
Add a get_and_clear_legacy_stats flag to AudioReceiveStream::GetStats,
to distinguish calls from standard GetStats and legacy GetStats.

Add const method NetEq::CurrentNetworkStatistics to get current
values of stateless NetEq stats. Standard GetStats will then call this
method instead of NetEq::NetworkStatistics.

Bug: webrtc:11622
Change-Id: I3833a246a9e39b18c99657a738da22c6e2bd5f5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183600
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32092}
2020-09-14 09:51:21 +00:00
Artem Titov
1ff3c584cd Add TimeController to the CreatePeerConnectionE2EQualityTestFixture API
Add TimeController to the CreatePeerConnectionE2EQualityTestFixture
method as a first step to make PC level framework compatible with
TimeController abstraction.

Bug: webrtc:11743
Change-Id: I69305abc880059bf9fe1d4f2e3b7c10cf35417db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178485
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31607}
2020-07-01 15:18:34 +00:00
Mirko Bonadei
08ce986fda Switch to absl single target when building with Chromium.
The //third_party/abseil-cpp:absl target is currently a group that
depends on all the targets needed by WebRTC in Chromium.

It will be switched to a component starting from
https://chromium-review.googlesource.com/c/chromium/src/+/2174434.

Bug: chromium:1046390
Change-Id: I70d450fdbfa895084b481c9884b6361d2fb9580d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176901
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31498}
2020-06-11 11:53:48 +00:00
Artem Titov
506d4eb7e4 Add missing headers to fix chromium roll
Bug: None
Change-Id: If28819bbeebe739f07fcd8d6ea8ab841efc20f75
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176562
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31453}
2020-06-05 17:49:04 +00:00
Patrik Höglund
a7a0173713 Remove the histogram flag and all Chart JSON code.
Since the flag is now on by default, we can remove it (after all
callers stop passing it).

We can also remove all Chart JSON code from WebRTC since it is
no longer used.

Requires one recipe CL and one downstream CL to land first.

Bug: chromium:1029452
Change-Id: Ic1d62e8ab9dfcd255cd2bf51d153db80d59c564b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171878
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30927}
2020-03-28 13:44:43 +00:00
Patrik Höglund
a298fd54c1 Don't double import protobuf code.
The proto code is copied into the out dir, so always use that since
it is what isolate is using. Previously we pointed straight at the
checkout code.

I think copying python into the out dir is probably the right way
to do things, so we should go that way in the future.

Bug: chromium:1029452
Change-Id: I701cc84a674021d2f78c73db8808f55cd6ae5174
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171877
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30923}
2020-03-27 19:22:37 +00:00
Patrik Höglund
a782299e91 Add PYTHONPATH entry for proto internal.
Turns out my new protobuf internal entries are copied to the out dir,
so we need a new entry for that.

Tbr: mbonadei@webrtc.org
Bug: chromium:1029452
Change-Id: I5bcae3a7ff1163e051382ae741646f206ccc7324
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171869
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30918}
2020-03-27 09:53:01 +00:00
Patrik Höglund
e4ae1ced9d Reland: Add in missing protobuf code.
Turns out the import of histogram_pb2 failed not on the stub itself
(which I thought for a long, long time), but because of the protobuf
support code it includes in turn. This is a drawback of catching
the ImportError in histogram_proto.py.

This has a decent chance of fixing the problem.

Tbr: mbonadei@webrtc.org
Bug: chromium:1029452
Change-Id: If7ae2439b01ad1b3129d8cc8b158385101082e6f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171867
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30917}
2020-03-27 08:47:44 +00:00
Patrik Höglund
749dff12b5 Revert "Add in missing protobuf code."
This reverts commit 0a663bf9f1.

Reason for revert: Breaks isolate tests

Original change's description:
> Add in missing protobuf code.
> 
> Turns out the import of histogram_pb2 failed not on the stub itself
> (which I thought for a long, long time), but because of the protobuf
> support code it includes in turn. This is a drawback of catching
> the ImportError in histogram_proto.py.
> 
> This has a decent chance of fixing the problem.
> 
> Tbr: mbonadei@webrtc.org
> Bug: chromium:1029452
> Change-Id: I3c07a362dcfd174a388b3cc34449c08951cea626
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171860
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30912}

TBR=phoglund@webrtc.org,mbonadei@webrtc.org

Change-Id: I05bf2f65905afcb7dfdc1e3fca7c01b4af377410
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1029452
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171866
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30915}
2020-03-27 08:04:42 +00:00
Patrik Höglund
0a663bf9f1 Add in missing protobuf code.
Turns out the import of histogram_pb2 failed not on the stub itself
(which I thought for a long, long time), but because of the protobuf
support code it includes in turn. This is a drawback of catching
the ImportError in histogram_proto.py.

This has a decent chance of fixing the problem.

Tbr: mbonadei@webrtc.org
Bug: chromium:1029452
Change-Id: I3c07a362dcfd174a388b3cc34449c08951cea626
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171860
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30912}
2020-03-27 07:47:09 +00:00
Patrik Höglund
28b8a0b2bc Partial revert of flag simplification.
Unfortunately it turns out the Android test runner requires
the isolated script flag to be in its current form, or it
doesn't work. This means we have to keep translating the
flag name.

We can get rid of the isolated_script_test_output flag
at least.

Tbr: mbonadei@webrtc.org
Bug: chromium:1051927
Change-Id: I4fdbff980e65332b757b1c95aa6587328411c0ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171809
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30907}
2020-03-26 20:09:33 +00:00
Patrik Höglund
7ac010cea4 Fix low bw test.
The problem turned out to be that it passes . as the path, and that
does not work in the PYTHONPATH.

Also remove debug logging.

Bug: chromium:1029452
Change-Id: Ied5211f6c039b41da9d77638801e67b7ea8f192f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171806
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30903}
2020-03-26 13:34:11 +00:00
Patrik Höglund
324646626f Add debug info to low bw test.
Bug: chromium:1029452
Tbr: mbonadei@webrtc.org
Change-Id: I7685dd9a2bb5ad1333836603ee8f8f7b10553179
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171804
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30900}
2020-03-26 11:07:16 +00:00
Patrik Höglund
3b4bbf5bcc Reland "Switch low bw audio test to histograms."
The reland makes sure the relevant code gets pulled by the isolate.

Also requires a recipe change so the results processor switches to
histogram mode when this CL is landed (see Chromium change 2119683).

Bug: chromium:1029452
Change-Id: I18bc9de72c8d21cb2942ca9af774d3227a8bf874
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171693
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30894}
2020-03-26 07:54:47 +00:00
Patrik Höglund
ad502fa727 Revert "Switch low bw audio test to histograms."
This reverts commit 71c9a18f25.

Reason for revert: Relevant python code isn't pulled when the test runs on swarming.

Original change's description:
> Switch low bw audio test to histograms.
> 
> Also requires a recipe change so the results processor switches to
> histogram mode when this CL is landed.
> 
> Bug: chromium:1029452
> Change-Id: Ic09deefc3f4f9d7a82ffeafeb5209fcfc361aece
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171683
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30884}

TBR=phoglund@webrtc.org,titovartem@webrtc.org

Change-Id: I6b3645ff939943a21185a1a1c8c5a0877e29db8c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1029452
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171692
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30893}
2020-03-26 06:46:33 +00:00
Patrik Höglund
71c9a18f25 Switch low bw audio test to histograms.
Also requires a recipe change so the results processor switches to
histogram mode when this CL is landed.

Bug: chromium:1029452
Change-Id: Ic09deefc3f4f9d7a82ffeafeb5209fcfc361aece
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171683
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30884}
2020-03-25 14:52:10 +00:00
Patrik Höglund
1b20c41dcb Greatly simplify flags for test binaries.
Since we're now calling the shots of what flags get passed in the
recipes, we can just pass the right ones right away and remove all
the flag renaming.

--isolated-script-test-output is no longer passed, so we can just
remove it. The recipe is currently passing
--isolated-script-perf-test-output but I will start passing the
underscore version shortly.

Bug: chromium:1051927
Change-Id: I571090e62f79ea17c793295df7f5abb21f45d207
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171681
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30878}
2020-03-25 09:56:07 +00:00
Patrik Höglund
b8e69efcee Write protos as binary.
We need to write protos as "wb" and not "w", otherwise we get CRLF
on Windows which corrupts the proto.

Bug: chromium:1029452
Change-Id: Iabf841405134d7bc2523ac48219ca7cb9d8214c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170320
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30772}
2020-03-12 09:43:57 +00:00
Danil Chapovalov
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
Bjorn A Mellem
7a9a092708 Delete media transport integration.
MediaTransport is deprecated and the code is unused.

No-Try: True
Bug: webrtc:9719
Change-Id: I5b864c1e74bf04df16c15f51b8fac3d407331dcd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160620
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29923}
2019-11-26 19:19:36 +00:00
Erik Språng
662678dbf7 Adds injectable trials from peerconnection down to transport controller.
This will be immediately useful to guarantee consistent state across
components referencing the pacer, but will be a net benefit overall
imo.

Bug: webrtc:10809
Change-Id: I49630696f757a832ccf2e4c8597193bf087ce53b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159885
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29859}
2019-11-21 12:41:45 +00:00
Ivo Creusen
c3d1f9b0cd Enable injection of a custom NetEqFactory into PeerConnectionFactory.
Injecting both a custom NetEqFactory and an AudioDecoderFactory is not
supported, in that case the AudioDecoderFactory should be wrapped inside
the NetEqFactory.

Bug: webrtc:11005
Change-Id: I4e311eb1bfa03c91bca587d70540e81829f881c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158720
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29673}
2019-11-01 11:30:36 +00:00
Niels Möller
ac0a4cbbd8 Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received"
This is a reland of fbde32e596

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
Mirko Bonadei
ef0627fb50 Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received"
This reverts commit fbde32e596.

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
Niels Möller
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
Danil Chapovalov
44db436e87 Propagate task queue to create test::DirectTransport by TaskQueueBase interface
actual task queue implementation for these tests is intentionally unchanged for now.

while at it, change return type of created transports to unique_ptr to note passing ownership.

Bug: webrtc:10933
Change-Id: I324597b503e647c471f43511340eb9c07ba03ee8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154743
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29335}
2019-09-30 03:23:07 +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
Niels Möller
65f17ca6b4 Move MediaTransportInterface out of the libjingle_peerconnection_api target
And move related files into api/transport/ and api/transport/media/.
The moved files are unchanged, except that
congestion_control_interface.h and datagram_transport_interface.h
no longer include media_transport_interface.h, instead, they forward
declare the few MediaTransport* types they reference.

Bug: webrtc:8733
Change-Id: I4f4000d0d111f10d15a54c99af27ec26c46ae652
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152482
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29178}
2019-09-13 10:49:56 +00:00
Yves Gerey
6516f76f9b Deprecate SingleThreadedTaskQueueForTesting class.
This class doesn't strictly follow rtc::TaskQueue semantics,
which makes it surprising and hard to use correctly.
Please use TaskQueueForTest instead.

This CL follows usual deprecation process:

1/ Rename.
% for i in `git ls-files` ; sed -i "s:SingleThreadedTaskQueueForTesting:DEPRECATED_SingleThreadedTaskQueueForTesting:" $i

2/ Annotate old name for downstream users and accidental new uses.

Bug: webrtc:10933
Change-Id: I80b4ee5a48df1f63f63a43ed0efdb50eb7fb156a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150788
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#29045}
2019-09-03 10:31:30 +00:00
Niels Möller
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
Danil Chapovalov
83bbe91398 Delete deprecated rtc_event_log header
Bug: webrtc:10206
Change-Id: I9ed3148843c647372993729b87c0e74741ab540b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147870
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28791}
2019-08-07 10:58:17 +00:00
Artem Titov
fedd625e0c Change 2g network pc audio test to more realistic network
Bug: webrtc:10138
Change-Id: I6f4b23fe702d26dbbeed05d0d09b79a9a966e40c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147728
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28752}
2019-08-02 22:28:02 +00:00
Artem Titov
cbc91efaa0 Improve low bandwidth audio test instrumentatin, fix PC test
Bug: webrtc:10138
Change-Id: I1d72fcac642064e569f6aac259fd0b6e0cf5c8b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146603
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28644}
2019-07-23 11:54:10 +00:00
Mirko Bonadei
2ab97f6f8e Migrate WebRTC test infra to ABSL_FLAG.
This is the last CL required to migrate WebRTC to ABSL_FLAG, rtc::Flag
will be removed soon after this one lands.

Bug: webrtc:10616
Change-Id: I2807cec39e28a2737d2c49e2dc23f2a6f98d08f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145727
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28606}
2019-07-19 06:54:04 +00:00
Artem Titov
52e240e2c1 Use 16000Hz audio in PC test when specified
Bug: webrtc:10138
Change-Id: Iea28aef07de45f244ed4d3813dc0531068f6d4b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144567
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28531}
2019-07-10 17:49:23 +00:00
Artem Titov
b1f2d60456 Reland "Fix collection of audio metrics from PC test framework for audio test"
This is a reland of 2d0880b569

To fix perf bot issue reading of perf results file was updated. Now perf
results file will be generated by each test and then returned via output to
the python script, which will get it and put into final file.

Original change's description:
> Fix collection of audio metrics from PC test framework for audio test
>
> Bug: webrtc:10138
> Change-Id: I18a8509a0cdc4ed1db6894c7540d5c0a155d6233
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144784
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28514}

Bug: webrtc:10138
Change-Id: I1347f09427736362a2d550612b48e09c06cfb1d1
No-Presubmit: True
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145201
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28527}
2019-07-10 15:21:30 +00:00
Mirko Bonadei
4876cb21c8 Revert "Fix collection of audio metrics from PC test framework for audio test"
This reverts commit 2d0880b569.

Reason for revert: Breaks perf tests (e.g. https://ci.chromium.org/p/webrtc/builders/perf/Perf%20Android32%20(L%20Nexus4)/1470).

Original change's description:
> Fix collection of audio metrics from PC test framework for audio test
> 
> Bug: webrtc:10138
> Change-Id: I18a8509a0cdc4ed1db6894c7540d5c0a155d6233
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144784
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28514}

TBR=oprypin@webrtc.org,ossu@webrtc.org,titovartem@webrtc.org

Change-Id: Id45abfbb0de47d95fe6cb396b1c29efb97a43797
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10138
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144883
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28519}
2019-07-09 20:09:14 +00:00
Artem Titov
2d0880b569 Fix collection of audio metrics from PC test framework for audio test
Bug: webrtc:10138
Change-Id: I18a8509a0cdc4ed1db6894c7540d5c0a155d6233
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144784
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28514}
2019-07-09 13:08:53 +00:00
Artem Titov
4a126e45c3 Rename tests to prevent clashing with old audio test
Bug: webrtc:10138
Change-Id: Ice785dad8646d28b2a2a7faa5d852679db41fbaa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144783
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28507}
2019-07-08 14:57:57 +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
Artem Titov
c8263e0638 Introduce PC level audio quality test.
Add PC level audio quality test as separate file but into the same
binary as low_bandwidth_audio_test.cc

Bug: webrtc:10138
Change-Id: I3bb7df4130ab62c21a82881cd9f57b936f3cc11d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144621
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28493}
2019-07-05 11:41:38 +00:00
Niels Möller
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
Danil Chapovalov
08fa953711 Reland "Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory"
This reverts commit fd5166c305.

Reason for revert: Stop using CreateTestAudioDeviceModule in downstream

Original change's description:
> Revert "Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory"
> 
> This reverts commit fc961357a7.
> 
> Reason for revert: Breaks downstream importer.
> 
> Original change's description:
> > Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory
> > 
> > Bug: webrtc:10284
> > Change-Id: Ic92f6ff31b40c48a3362745a0a81179af0595fe0
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141409
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28227}
> 
> TBR=danilchap@webrtc.org,kwiberg@webrtc.org
> 
> Change-Id: Id6d7571f48771646ddce0f05139a7ea0107759fb
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10284
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141414
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28228}

TBR=danilchap@webrtc.org,kwiberg@webrtc.org,philipel@webrtc.org

Change-Id: I42bc19793d48350ca45b751d7e1b26124ac7fbb9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10284
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141670
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28254}
2019-06-12 14:44:01 +00:00
Philip Eliasson
fd5166c305 Revert "Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory"
This reverts commit fc961357a7.

Reason for revert: Breaks downstream importer.

Original change's description:
> Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory
> 
> Bug: webrtc:10284
> Change-Id: Ic92f6ff31b40c48a3362745a0a81179af0595fe0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141409
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28227}

TBR=danilchap@webrtc.org,kwiberg@webrtc.org

Change-Id: Id6d7571f48771646ddce0f05139a7ea0107759fb
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10284
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141414
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28228}
2019-06-11 12:32:23 +00:00
Danil Chapovalov
fc961357a7 Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory
Bug: webrtc:10284
Change-Id: Ic92f6ff31b40c48a3362745a0a81179af0595fe0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141409
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28227}
2019-06-11 12:15:44 +00:00
Danil Chapovalov
b32f2c7f57 Publish rtc event log api and default factory for it in api/
Bug: webrtc:10206
Change-Id: I34194ddb6fd2b0a3d7c553fadc9ddc1ea9740da0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137500
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28023}
2019-05-22 13:38:25 +00:00
Anton Sukhanov
4f08faae82 Introduce MediaTransportConfig
Currently we pass media_transport from PeerConnection to media layers. The goal of this change is to replace media_transport with struct MediaTransportCondif, which will enable adding different transports (i.e. we plan to add DatagramTransport) as well as other media-transport related settings without changing 100s of files.

TODO: In the future we should consider also adding rtp_transport in the same config, but it will require a bit more work, so I did not include it in the same change.


Bug: webrtc:9719
Change-Id: Ie31e1faa3ed9e6beefe30a3da208130509ce00cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137181
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28016}
2019-05-21 18:58:33 +00:00
Ying Wang
0810a7c25a Add base class NetworkPredictor and NetworkPredictorFactory and wire up.
Add base class NetworkPredictor and NetworkPredictorFactory in /api, make it possible to inject customized NetworkPredictor in PeerConnectionFactory level. The NetworkPredictor object will be pass down to GoogCCNetworkControl and DelayBasedBwe.

Bug: webrtc:10492
Change-Id: Iceeadbe1c9388b11ce4ac01ee56554cb0bf64d04
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130201
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27543}
2019-04-10 12:38:58 +00:00
Mirko Bonadei
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
Danil Chapovalov
31660fdfea Avoid using global task queue factory in audio/ unittests
in particular replace rtc::TaskQueue with TaskQueueForTest class since
latter uses DefaultTaskQueueFactory() directly instead of through
GlobalTaskQueueFactory

Bug: webrtc:10284
Change-Id: I1a52c5942626e3e2256b3d78975d2740e9facb1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128880
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27245}
2019-03-22 15:53:28 +00:00
Oleksandr Iakovenko
53de7255b9 Fix outdated android sdk path in tests.
Bug: chromium:943507
Change-Id: Iffdf18a66485a98f08b2a556c1b3fa1e817fafba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128607
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Oleksandr Iakovenko <iakovenko@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27224}
2019-03-21 12:18:19 +00:00
Benjamin Wright
17b050f8f8 Fixes ClangTidy errors in audio/
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: I1b46653b91bce012afabfa0f2d249718e6de2df8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127626
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27139}
2019-03-15 01:55:52 +00:00
Danil Chapovalov
471783fc87 Remove rtc::QueuedTask alias, use webrtc::QueuedTask directly
Use absl::WrapUnique/absl::make_unique to create the queued tasks.

Bug: webrtc:10191
Change-Id: I8f47a60cb326b0fc361c7f0e338b25373d39937c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126525
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27063}
2019-03-11 16:49:21 +00:00
Danil Chapovalov
ad89528051 Reland "Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current"
This reverts commit 42d8c93ec3.

Reason for revert: Got Aliby for FEC test flakes

Original change's description:
> Revert "Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current"
> 
> This reverts commit 304e9d2df3.
> 
> Reason for revert: Breaks downstream projects.
> Seems to make VideoSendStreamTest.SupportsFlexfecSimulcastVp8 flaky.
> 
> Original change's description:
> > Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current
> > 
> > Bug: webrtc:10191
> > Change-Id: I506cc50a90c73a6a4f6a3de36de0999cca72f5ba
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126230
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27035}
> 
> TBR=danilchap@webrtc.org,kwiberg@webrtc.org
> 
> Change-Id: If98324f88e4b3d18bf2fe33597dfb9711867c243
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10191
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126484
> Reviewed-by: Yves Gerey <yvesg@webrtc.org>
> Commit-Queue: Yves Gerey <yvesg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27041}

TBR=danilchap@webrtc.org,kwiberg@webrtc.org,yvesg@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:10191
Change-Id: Id87a17ae415142b8e0b11ba03ae7bad84a473fb0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126720
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27056}
2019-03-11 12:32:49 +00:00
Yves Gerey
42d8c93ec3 Revert "Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current"
This reverts commit 304e9d2df3.

Reason for revert: Breaks downstream projects.
Seems to make VideoSendStreamTest.SupportsFlexfecSimulcastVp8 flaky.

Original change's description:
> Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current
> 
> Bug: webrtc:10191
> Change-Id: I506cc50a90c73a6a4f6a3de36de0999cca72f5ba
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126230
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27035}

TBR=danilchap@webrtc.org,kwiberg@webrtc.org

Change-Id: If98324f88e4b3d18bf2fe33597dfb9711867c243
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10191
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126484
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27041}
2019-03-08 16:14:54 +00:00
Sebastian Jansson
44dd9f29c7 Adds ChannelSend specific encoder task queue.
Before this change the encoder tasks runs on a shared worker queue.
That makes the destruction require synchronization to avoid races.
By keeping a separate encode queue to ChannelSend, we can safely
destruct the object without worrying for left over tasks, as they
will be stopped when the task queue is destroyed.

For TaskQueue implementations using one thread per TaskQueue this
will increase the thread count by the number of AudioSendStreams,
which typically is just one.

This is partly a reland of 9b9344742b

Original change's description:
> Removes lock from ChannelSend.
>
> The lock isn't really needed as encoder_queue_is_active_ can be checked
> on the task queue to provide synchronization.
>
> There is one behavioral change due to this: We will not cancel any currently
> pending encoding tasks when we stop sending, they will be allowed to finish.
>
> Bug: webrtc:10365
> Change-Id: I2b4897dde8d49bc7ee5d2d69694616aee8aaea38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125096
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26963}

Bug: webrtc:10365
Change-Id: Iafb84e25d90ec8639359be80fad65763d08e5719
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125740
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27038}
2019-03-08 15:53:12 +00:00
Danil Chapovalov
304e9d2df3 Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current
Bug: webrtc:10191
Change-Id: I506cc50a90c73a6a4f6a3de36de0999cca72f5ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126230
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27035}
2019-03-08 13:17:46 +00:00
Sebastian Jansson
0b69826ffb Don't inject worker queue into send streams.
This prepares for making AudioSendStream use its own task queue. In the
future more of the functionality that depends on running on the task
queue is planned to be moved directly into RtpTransportControllerSend.

They should instead get it from the transport controller. This affects
the media transport tests which previously assumed that the transport
controller could be missing. However, this is not something that is used
in production, so this is an improvement of the tests as they will
behave more like production code.

Bug: webrtc:9883
Change-Id: Ie32f4c2f6433ec37ac16a08d531ceb690ea9c0b5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126000
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27010}
2019-03-07 09:42:26 +00:00
Sebastian Jansson
977b3351b9 Injecting Clock into audio streams.
Bug: webrtc:10365
Change-Id: Ia47fd806b84d94fd90b734c87c5e338e36fb695a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125191
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26969}
2019-03-05 10:49:46 +00:00
Elad Alon
d8d3248d95 Reland "Delete test/constants.h"
This reverts commit 4f36b7a478.

Reason for revert: Failing tests fixed.

Original change's description:
> Revert "Delete test/constants.h"
>
> This reverts commit 389b1672a3.
>
> Reason for revert: Causes failure (and empty result list) in CallPerfTest.PadsToMinTransmitBitrate
>
> Original change's description:
> > Delete test/constants.h
> >
> > It's not possible to use constants.h for all RTP extensions
> > after the number of extensions exceeds 14, which is the maximum
> > number of one-byte RTP extensions. This is because some extensions
> > would have to be assigned a number greater than 14, even if the
> > test only involves 14 extensions or less.
> >
> > For uniformity's sake, this CL also edits some files to use an
> > enum as the files involved in this CL, rather than free-floating
> > const-ints.
> >
> > Bug: webrtc:10288
> > Change-Id: Ib5e58ad72c4d3756f4c4f6521f140ec59617f3f5
> > Reviewed-on: https://webrtc-review.googlesource.com/c/123048
> > Commit-Queue: Elad Alon <eladalon@webrtc.org>
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#26728}
>
> TBR=danilchap@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org
>
> Bug: webrtc:10288, chromium:933127
> Change-Id: If1de0bd8992137c52bf0b877b3cb0a2bafc809d4
> Reviewed-on: https://webrtc-review.googlesource.com/c/123381
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26744}

TBR=danilchap@webrtc.org,oprypin@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org

Change-Id: I65e391325d3a6df6db3c0739185e2002e70fb954
Bug: webrtc:10288, chromium:933127
Reviewed-on: https://webrtc-review.googlesource.com/c/123384
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26750}
2019-02-19 08:51:20 +00:00
Oleh Prypin
4f36b7a478 Revert "Delete test/constants.h"
This reverts commit 389b1672a3.

Reason for revert: Causes failure (and empty result list) in CallPerfTest.PadsToMinTransmitBitrate

Original change's description:
> Delete test/constants.h
>
> It's not possible to use constants.h for all RTP extensions
> after the number of extensions exceeds 14, which is the maximum
> number of one-byte RTP extensions. This is because some extensions
> would have to be assigned a number greater than 14, even if the
> test only involves 14 extensions or less.
>
> For uniformity's sake, this CL also edits some files to use an
> enum as the files involved in this CL, rather than free-floating
> const-ints.
>
> Bug: webrtc:10288
> Change-Id: Ib5e58ad72c4d3756f4c4f6521f140ec59617f3f5
> Reviewed-on: https://webrtc-review.googlesource.com/c/123048
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26728}

TBR=danilchap@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org

No-Presubmit: True
Bug: webrtc:10288, chromium:933127
Change-Id: If1de0bd8992137c52bf0b877b3cb0a2bafc809d4
Reviewed-on: https://webrtc-review.googlesource.com/c/123381
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26744}
2019-02-18 18:09:22 +00:00
Elad Alon
389b1672a3 Delete test/constants.h
It's not possible to use constants.h for all RTP extensions
after the number of extensions exceeds 14, which is the maximum
number of one-byte RTP extensions. This is because some extensions
would have to be assigned a number greater than 14, even if the
test only involves 14 extensions or less.

For uniformity's sake, this CL also edits some files to use an
enum as the files involved in this CL, rather than free-floating
const-ints.

Bug: webrtc:10288
Change-Id: Ib5e58ad72c4d3756f4c4f6521f140ec59617f3f5
Reviewed-on: https://webrtc-review.googlesource.com/c/123048
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26728}
2019-02-17 21:47:41 +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
Steve Anton
40d55331d7 Include absl/memory/memory.h if absl::make_unique is used
Tbr: kwiberg@webrtc.org
Bug: None
Change-Id: Iaf4533d2ce0e80b351a8a664ef8cf7ba0e5ec583
Reviewed-on: https://webrtc-review.googlesource.com/c/115746
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@google.com>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26168}
2019-01-08 20:08:32 +00:00
Sam Zackrisson
ff0581672e Delete the WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds metric
It never saw much use, and is blocking refactoring.

Histograms.xml-side cleanup:
https://chromium-review.googlesource.com/c/chromium/src/+/1344141

Bug: webrtc:7882
Change-Id: I112232a573fcd218dc7a51bfcdd7898847d14f18
Reviewed-on: https://webrtc-review.googlesource.com/c/111506
Commit-Queue: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25780}
2018-11-26 09:32:35 +00:00
Bjorn Mellem
273d0296a4 Implement data channel methods in LoopbackMediaTransport.
This enables PeerConnection tests to use LoopbackMediaTransport to test
data-channel-over-media-transport code.

Also changes LoopbackMediaTransport to invoke callbacks asynchronously.
This is more accurate, as these callbacks are triggered by network
events.  The caller should not block while the callback executes.

Since LoopbackMediaTransport is used for testing, it provides a
FlushAsyncInvokes() method which may be used to ensure that callbacks
occur deterministically (eg. before checking that data has been
received).

Bug: webrtc:9719
Change-Id: Ib8ea9aebf4a0ad3d5934a6fe4ab33432c68523fd
Tbr: stefan@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/c/109060
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25489}
2018-11-02 16:15:32 +00:00
Niels Möller
7d76a31f3d Use MediaTransportInterface, for audio streams.
Bug: webrtc:9719
Change-Id: I6d3db66b781173b207de51d84193fbd34a7f3239
Reviewed-on: https://webrtc-review.googlesource.com/c/104642
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25385}
2018-10-26 11:40:57 +00:00
Mirko Bonadei
2dfa998be2 Reland "Prefix flag macros with WEBRTC_."
This is a reland of 5ccdc1331f

Original change's description:
> Prefix flag macros with WEBRTC_.
>
> Macros defined in rtc_base/flags.h are intended to be used to define
> flags in WebRTC's binaries (e.g. tests).
>
> They are currently not prefixed and this could cause problems with
> downstream clients since these names are quite common.
>
> This CL adds the 'WEBRTC_' prefix to them.
>
> Generated with:
>
> for x in DECLARE DEFINE; do
>   for y in bool int float string FLAG; do
>     git grep -l "\b$x\_$y\b" | \
>     xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
>   done
> done
> git cl format
>
> Bug: webrtc:9884
> Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
> Reviewed-on: https://webrtc-review.googlesource.com/c/106682
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25270}

TBR=kwiberg@webrtc.org

Bug: webrtc:9884
Change-Id: I5ba5368a231a334d135ed5e6fd7a279629ced8a3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/107161
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25277}
2018-10-19 15:06:43 +00:00
Mirko Bonadei
c538fc77b0 Revert "Prefix flag macros with WEBRTC_."
This reverts commit 5ccdc1331f.

Reason for revert: Breaks downstream project.

Original change's description:
> Prefix flag macros with WEBRTC_.
> 
> Macros defined in rtc_base/flags.h are intended to be used to define
> flags in WebRTC's binaries (e.g. tests).
> 
> They are currently not prefixed and this could cause problems with
> downstream clients since these names are quite common.
> 
> This CL adds the 'WEBRTC_' prefix to them.
> 
> Generated with:
> 
> for x in DECLARE DEFINE; do
>   for y in bool int float string FLAG; do
>     git grep -l "\b$x\_$y\b" | \
>     xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
>   done
> done
> git cl format
> 
> Bug: webrtc:9884
> Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
> Reviewed-on: https://webrtc-review.googlesource.com/c/106682
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25270}

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org

Change-Id: Ia79cd6066ecfd1511c34f1b30fd423e560ed6854
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9884
Reviewed-on: https://webrtc-review.googlesource.com/c/107160
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25276}
2018-10-19 15:04:13 +00:00
Mirko Bonadei
5ccdc1331f Prefix flag macros with WEBRTC_.
Macros defined in rtc_base/flags.h are intended to be used to define
flags in WebRTC's binaries (e.g. tests).

They are currently not prefixed and this could cause problems with
downstream clients since these names are quite common.

This CL adds the 'WEBRTC_' prefix to them.

Generated with:

for x in DECLARE DEFINE; do
  for y in bool int float string FLAG; do
    git grep -l "\b$x\_$y\b" | \
    xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
  done
done
git cl format

Bug: webrtc:9884
Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
Reviewed-on: https://webrtc-review.googlesource.com/c/106682
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25270}
2018-10-19 10:55:20 +00:00
Artem Titov
75e3647a76 Switch usages of DefaultNetworkSimulationConfig to BuiltInNetworkBehaviorConfig
Bug: webrtc:9630
Change-Id: Ia0e0b5b4e1e3a8e687d1e7fe3bb600dbdda09efa
Reviewed-on: https://webrtc-review.googlesource.com/c/104561
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25045}
2018-10-08 12:19:31 +00:00
Stefan Holmer
64be7fa7d8 Move FecController to RtpVideoSender.
This also moves the packet feedback tracking to RtpVideoSender.

Bug: webrtc:9517
Change-Id: Ifb1ff85051730108a0b0d1dd30f6f8595ad2af6e
Reviewed-on: https://webrtc-review.googlesource.com/c/95920
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25019}
2018-10-05 14:39:01 +00:00
Oleh Prypin
637b0b5d38 Make Python-based performance tests output an empty result output.json
TBR: phoglund@webrtc.org
Bug: webrtc:9767
Change-Id: I2e51e33ae2fd13a1e09f641dd4f2819f5901b15b
Reviewed-on: https://webrtc-review.googlesource.com/101360
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24784}
2018-09-21 15:45:38 +00:00
Artem Titov
4e199e9f08 Mark DirectTransport subclasses ctors as deprecated and switch from them
Bug: webrtc:9630
Change-Id: I6e7bf898fd95ef76758458e759d9f9aa381f89e1
Reviewed-on: https://webrtc-review.googlesource.com/94843
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24345}
2018-08-20 12:05:05 +00:00
Artem Titov
46c4e60939 Introduce SimulatedNetworkReceiverInterface.
Introduce SimulatedNetworkReceiverInterface and switch DirectTransport
on this interface. Also switch part of related users on
DefaultNetworkSimulationConfig.

This two changes united into single CL to prevent work duplication.
Most changes were done because of stop including fake_network_pipe.h
into direct_transport.h, so splitting this into 2 CLs will require
first fix all imports of fake_network_pipe.h and then replace them
on new API imports again.

Bug: webrtc:9630
Change-Id: I87d4a6ff1bab72d04a9871a40441f4fbe028f4e6
Reviewed-on: https://webrtc-review.googlesource.com/94762
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24336}
2018-08-20 07:23:41 +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
Yves Gerey
665174fdbb Reformat the WebRTC code base
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}
2018-06-19 14:00:39 +00:00
Edward Lesmes
9599fd4414 Make num-retries default a string.
TBR=phoglund@webrtc.org

Bug: chromium:755660
Change-Id: I770a79a78721a312b603aec40d23689245a48001
Reviewed-on: https://webrtc-review.googlesource.com/61343
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22392}
2018-03-12 21:19:51 +00:00
Edward Lesmes
5b9c6840b1 Add num-retries flag to Android perf tests.
Add a flag to Android perf tests, so we can specify the number of
retries.

Bug: chromium:755660
Change-Id: Ic498373421b7e0fdf779a4659a0c79d47a59fbde
Reviewed-on: https://webrtc-review.googlesource.com/61103
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22390}
2018-03-12 19:51:09 +00:00
Artem Titov
3faa832247 Separate test/fake_audio_device on API and implementation. Step 2.
Switch WebRTC internal usage of FakeAudioDevice on TestAudioDeviceModule.

Bug: webrtc:8946
Change-Id: I96b8b5d3b475d2197662e9007f836bd71f8ed04d
Reviewed-on: https://webrtc-review.googlesource.com/60521
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22388}
2018-03-12 16:14:39 +00:00
Artem Titov
6723cdc8a4 Revert "Separate test/fake_audio_device on API and implementation."
This reverts commit 8ea5f9ae5b.

Reason for revert: breaks downstream project

Original change's description:
> Separate test/fake_audio_device on API and implementation.
> 
> Adding ability of injecting audio in end to end tests, that are using
> WebRTC. For this purpose as a 1st step test/fake_audio_device will
> be moved to production part of WebRTC source code and renamed to
> test_audio_device_module. Old header is replaced with alias to the
> new one and will be deleted after a while.
> 
> Change-Id: I5284d1dd46ce9bf86cf43268e855520606fa8c5c
> 
> Bug: webrtc:8946
> Change-Id: I5284d1dd46ce9bf86cf43268e855520606fa8c5c
> Reviewed-on: https://webrtc-review.googlesource.com/58086
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22289}

TBR=kwiberg@webrtc.org,titovartem@webrtc.org

Change-Id: I88d9c4f09cc576ed7c9182dcf0a873d25a8bab54
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8946
Reviewed-on: https://webrtc-review.googlesource.com/59720
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22291}
2018-03-05 15:36:23 +00:00
Artem Titov
8ea5f9ae5b Separate test/fake_audio_device on API and implementation.
Adding ability of injecting audio in end to end tests, that are using
WebRTC. For this purpose as a 1st step test/fake_audio_device will
be moved to production part of WebRTC source code and renamed to
test_audio_device_module. Old header is replaced with alias to the
new one and will be deleted after a while.

Change-Id: I5284d1dd46ce9bf86cf43268e855520606fa8c5c

Bug: webrtc:8946
Change-Id: I5284d1dd46ce9bf86cf43268e855520606fa8c5c
Reviewed-on: https://webrtc-review.googlesource.com/58086
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22289}
2018-03-05 14:30:42 +00:00
Edward Lemur
ed7b4ff9e3 Use isolated-script-test-perf-output on low_bandwidth_audio_test.
Instead of chartjson-result-file, since that's the flag passed by the recipe.

TBR=phoglund@webrtc.org

Bug: chromium:807737
Change-Id: I3a679ab7e5c0a446e675d0f4647344cc4194b357
Reviewed-on: https://webrtc-review.googlesource.com/46541
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21858}
2018-02-01 20:35:24 +00:00
Edward Lemur
d8b041cc54 Ignore extra arguments in low_bandwidth_audio_test.
R=phoglund@webrtc.org
TBR=solenberg@webrtc.org

Bug: chromium:755660
Change-Id: I39488f6905875bdc08b006619b972d3f1af2fce1
Reviewed-on: https://webrtc-review.googlesource.com/39924
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21641}
2018-01-16 14:07:54 +00:00
Edward Lemur
b4017712c5 Store JSON perf results for low_bandwidth_audio_test.
Bug: chromium:755660
Change-Id: I87c19f8dde14772e75f93c723f11408702ca8c8e
Reviewed-on: https://webrtc-review.googlesource.com/39515
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21615}
2018-01-15 15:24:09 +00:00
Ivo Creusen
56d460902e Use the new AudioProcessing statistics everywhere.
The new interface uses optionals instead of default values, and only values that are actually used are included. To make it easy to add/remove stats in the future, the struct itself is copied around on all layers, instead of copying the values one by one. This CL also fixes a bug which caused several APM statistics to get stuck at a fixed level when there are no more receive streams (after some period where there were receive streams). Since APM doesn't know this happens, an argument was added to the GetStats call to pass this information down to APM.

Bug: webrtc:8563, b/67926135
Change-Id: I96cc008353355bb520c4523f5c5379860f73ee24
Reviewed-on: https://webrtc-review.googlesource.com/25621
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20877}
2017-11-24 18:17:39 +00:00
Karl Wiberg
e40468ba3d Move some numeric utility code from rtc_base/ to rtc_base/numerics/
Specifically, I'm moving

  safe_compare.h
  safe_conversions.h
  safe_minmax.h

They shouldn't be part of the API, and moving them to an appropriate
subdirectory of rtc_base/ is a good way to keep track of that.

BUG=webrtc:8445

Change-Id: I458531aeb30bcf4291c4bec3bf22a2fffbf054ff
Reviewed-on: https://webrtc-review.googlesource.com/20860
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20829}
2017-11-22 11:21:47 +00:00
Oskar Sundbom
2707fb2782 Optional: Use nullopt and implicit construction in /audio
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.

This CL was uploaded by git cl split.

R=solenberg@webrtc.org

Bug: None
Change-Id: I03562600978bdedb9dc93a34aeb0561c66f54aae
Reviewed-on: https://webrtc-review.googlesource.com/23617
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20731}
2017-11-17 15:56:17 +00:00
Edward Lemur
88b23f6662 Fix flag name in low_bandwidth_audio_test.py
TBR=kjellander@webrtc.org, oprypin@webrtc.org

No-Try: true
Bug: chromium:755660
Change-Id: I6bf5dda5374cae16da54aa10e77b136c638e1975
Reviewed-on: https://webrtc-review.googlesource.com/6442
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20148}
2017-10-04 17:26:14 +00:00
Edward Lemur
7e3b5697d9 Ignore swarming arguments in low_bandwidth_audio_test.py
Needed because swarming adds --isolated-script-test-output and --isolated-script-test-perf-output

See for example:
https://chromium-swarm.appspot.com/task?id=39006c763bebf710

No-Try: true
Bug: chromium:755660
Change-Id: Iff9fb3441200f760c511a67211fbc4a1272717b4
Reviewed-on: https://webrtc-review.googlesource.com/6362
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20146}
2017-10-04 16:19:44 +00:00
Edward Lemur
b0250f0504 Reland "Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script."
This is a reland of f4898a6509
Original change's description:
> Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script.
> 
> They should've been downloaded already.
> 
> NOTRY=True
> 
> Bug: chromium:755660
> Change-Id: I8ecb355f06026a38bd9377633e2be6c55d7c6452
> Reviewed-on: https://webrtc-review.googlesource.com/5620
> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20109}

No-Try: true
Bug: chromium:755660
Change-Id: I391130545eee5d4928101f87ac4a4e0945d665a1
Reviewed-on: https://webrtc-review.googlesource.com/6380
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20140}
2017-10-04 13:03:24 +00:00
Edward Lemur
45a0b36d3f Revert "Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script."
Reason for revert: Breaks windows bot.
https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/3804

TBR=kjellander@webrtc.org,phoglund@webrtc.org,ehmaldonado@webrtc.org,oprypin@webrtc.org

Change-Id: I0f2221b66c4f7dcf0a6f03004e5acc24c77ba8b0
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:755660
Reviewed-on: https://webrtc-review.googlesource.com/6001
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20117}
2017-10-03 14:00:15 +00:00
Edward Lemur
f4898a6509 Reland "Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script."
This reverts commit bb1222f3ad.

Reason for revert: Fixing

Original change's description:
> Revert "Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script."
> 
> This reverts commit 2019698455.
> 
> Reason for revert: Fails on Mac
> https://build.chromium.org/p/client.webrtc.perf/builders/Mac%2010.11/builds/4070
> 
> Original change's description:
> > Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script.
> > 
> > They should've been downloaded already.
> > 
> > NOTRY=True
> > 
> > Bug: chromium:755660
> > Change-Id: I8ecb355f06026a38bd9377633e2be6c55d7c6452
> > Reviewed-on: https://webrtc-review.googlesource.com/5620
> > Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
> > Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> > Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#20109}
> 
> TBR=kjellander@webrtc.org,phoglund@webrtc.org,ehmaldonado@webrtc.org,oprypin@webrtc.org
> 
> Change-Id: I0cfc1d0b398587a023af528536e7d995c6de1413
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:755660
> Reviewed-on: https://webrtc-review.googlesource.com/5940
> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20112}

No-Try: true
Bug: chromium:755660
Change-Id: Idbf30003c78c4093afcaf99dc0c5bb3468cdcf6d
Reviewed-on: https://webrtc-review.googlesource.com/5941
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20113}
2017-10-03 12:49:51 +00:00
Edward Lemur
bb1222f3ad Revert "Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script."
This reverts commit 2019698455.

Reason for revert: Fails on Mac
https://build.chromium.org/p/client.webrtc.perf/builders/Mac%2010.11/builds/4070

Original change's description:
> Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script.
> 
> They should've been downloaded already.
> 
> NOTRY=True
> 
> Bug: chromium:755660
> Change-Id: I8ecb355f06026a38bd9377633e2be6c55d7c6452
> Reviewed-on: https://webrtc-review.googlesource.com/5620
> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20109}

TBR=kjellander@webrtc.org,phoglund@webrtc.org,ehmaldonado@webrtc.org,oprypin@webrtc.org

Change-Id: I0cfc1d0b398587a023af528536e7d995c6de1413
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:755660
Reviewed-on: https://webrtc-review.googlesource.com/5940
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20112}
2017-10-03 12:33:52 +00:00
Edward Lemur
2019698455 Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script.
They should've been downloaded already.

NOTRY=True

Bug: chromium:755660
Change-Id: I8ecb355f06026a38bd9377633e2be6c55d7c6452
Reviewed-on: https://webrtc-review.googlesource.com/5620
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20109}
2017-10-03 11:25:10 +00:00
solenberg
4652e86c0c Disable flaky AudioStats.NoLoss test.
BUG=none

Review-Url: https://codereview.webrtc.org/3013783002
Cr-Commit-Position: refs/heads/master@{#19928}
2017-09-22 13:07:56 +00:00
Henrik Kjellander
5a6aa4f05d Fix path to root in low_bandwidth_audio_test.py
BUG=chromium:611808
TBR=solenberg@webrtc.org
NOTRY=True
NOPRESUBMIT=True
NOTREECHECKS=True

Change-Id: Iba2b0851ee99916b9809231b4b27046315fd8565
Reviewed-on: https://webrtc-review.googlesource.com/1569
Commit-Queue: Henrik Kjellander <kjellander@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19849}
2017-09-15 08:29:11 +00:00