Commit Graph

192 Commits

Author SHA1 Message Date
4ba04b7740 Delete RtcEventLogFactory factory as now unused
Bug: webrtc:10206, webrtc:10284
Change-Id: I34fa780f566b52e375ec625bf0d5d02c505d9912
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143782
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28400}
2019-06-27 10:03:22 +00:00
ef3fd9c8ad Add support for simulcast with Vp8 from caller into PC level quality tests.
Add support of negotiating simulcast offer/answer. Also fix some minor
issues around to make it finally work.

Bug: webrtc:10138
Change-Id: I382f5df04ca6ac04d8ed1e030e7b2ae5706dd10c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137425
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28274}
2019-06-13 17:27:09 +00:00
1a5fc9035b in test/pc/e2e pass TaskQueueFactory explicitly
instead of relying on factories that use GlobalTaskQueueFactory

Bug: webrtc:10284
Change-Id: Icc32ae1c159c39a6594d2aaec79c68dcc826fea8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139894
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28220}
2019-06-11 08:48:56 +00:00
5d24b16c77 Prepare for splitting the api/video:video_frames build rule.
This change is part of a change to break the dependency between "api:rtp_headers" and "api/video:video_frame". It does so by first creating an empty "api/video:video_rtp_headers" build rule so that downstream projects can be fixed before moving the source files.

Bug: webrtc:10668
Change-Id: I81aa6edfef3639b457a40aa93de048e62cbfd8ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140291
Commit-Queue: Chen Xing <chxg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28209}
2019-06-10 11:50:51 +00:00
2370242acf Enable flex fec support in PC quality test framework
Bug: webrtc:10138, webrtc:10683
Change-Id: I9235fef99d3ea857f10234fdd82e8468480f71a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138822
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28078}
2019-05-27 14:48:48 +00:00
daac58290e Remove -Wno-undef and -Wno-extra-semi.
These issues have been fixed upstream in Abseil.

Bug: webrtc:10138
Change-Id: Ic0ebd22d0ad95bbd5269c08c182a76f9bf42f3a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135571
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27883}
2019-05-08 17:42:09 +00:00
d8b9ed77cf Promote RtcEventLogOutputFile to api/
Preparation for deleting PeerConnectionInterface::StartRtcEventLog
method with a PlatformFile argument.

Bug: webrtc:6463
Change-Id: Ia9fa1d99a3d87f3bf193e73382690b782ffea65c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135285
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27879}
2019-05-08 12:29:42 +00:00
60f14ce217 Do not use absl::flat_hash_map in DefaultVideoQualityAnalyzer.
This CL removes the usage of absl::flat_hash_map because it transitively
depends on CCTZ which fails to link with lld-link after the switch to
libc++.

Since std::map doesn't support heterogeneous lookup until C++14, this
CL also stops using absl::string_view and switches to
`const std::string&`.

Bug: webrtc:10605
Change-Id: I4fc93969c6fc0cc7e7e62b4d2f801bdd27cff0f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135566
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27877}
2019-05-08 10:23:59 +00:00
f65a89b7f7 Add support of specifying concrete codec for video stream
Bug: webrtc:10138
Change-Id: I074bfccfa5c8f619ea7fa17d6ca99f9b4cbb18b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123386
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27864}
2019-05-07 11:46:57 +00:00
4487ac4a53 Reland "Add Video Bwe stats collection to DefaultVideoQualityAnalyzer."
This is a reland of 8848229234aae01ec19582ece7b748d557119d66

Original change's description:
> Add Video Bwe stats collection to DefaultVideoQualityAnalyzer.
>
> This CL adds the possibility to collect the following Video BWE stats:
> - available_send_bandwidth
> - transmission_bitrate
> - retransmission_bitrate
> - actual_encode_bitrate
> - target_encode_bitrate
>
> Example of the output:
>
> RESULT available_send_bandwidth: smoke_test/alice= {487754.33,87583.093} bytesPerSecond
> RESULT transmission_bitrate: smoke_test/alice= {465779.17,212075.5} bytesPerSecond
> RESULT retransmission_bitrate: smoke_test/alice= {20036,26326.751} bytesPerSecond
> RESULT actual_encode_bitrate: smoke_test/alice= {418779.33,200486.03} bytesPerSecond
> RESULT target_encode_bitrate: smoke_test/alice= {469491.17,77866.909} bytesPerSecond
> RESULT available_send_bandwidth: smoke_test/bob= {642924.83,168842.34} bytesPerSecond
> RESULT transmission_bitrate: smoke_test/bob= {626115.5,294783.56} bytesPerSecond
> RESULT retransmission_bitrate: smoke_test/bob= {0,0} bytesPerSecond
> RESULT actual_encode_bitrate: smoke_test/bob= {594235.33,297289.54} bytesPerSecond
> RESULT target_encode_bitrate: smoke_test/bob= {640463.5,167676.66} bytesPerSecond
>
> Bug: webrtc:10138
> Change-Id: I0414055af0010b8fb4d909297e6da86d398157c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132703
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Tommi <tommi@webrtc.org>
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@google.com>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27760}

TBR=tommi@webrtc.org

Bug: webrtc:10138
Change-Id: Ib76dfeca741134d6f18ae0eb436920ead42a1d42
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134543
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27856}
2019-05-06 06:32:48 +00:00
e680c83a41 Revert "Add Video Bwe stats collection to DefaultVideoQualityAnalyzer."
This reverts commit 8848229234aae01ec19582ece7b748d557119d66.

Reason for revert: break chromium compilation on iOS
https://logs.chromium.org/logs/chromium/buildbucket/cr-buildbucket.appspot.com/8915214519549611184/+/steps/compile/0/stdout

Original change's description:
> Add Video Bwe stats collection to DefaultVideoQualityAnalyzer.
> 
> This CL adds the possibility to collect the following Video BWE stats:
> - available_send_bandwidth
> - transmission_bitrate
> - retransmission_bitrate
> - actual_encode_bitrate
> - target_encode_bitrate
> 
> Example of the output:
> 
> RESULT available_send_bandwidth: smoke_test/alice= {487754.33,87583.093} bytesPerSecond
> RESULT transmission_bitrate: smoke_test/alice= {465779.17,212075.5} bytesPerSecond
> RESULT retransmission_bitrate: smoke_test/alice= {20036,26326.751} bytesPerSecond
> RESULT actual_encode_bitrate: smoke_test/alice= {418779.33,200486.03} bytesPerSecond
> RESULT target_encode_bitrate: smoke_test/alice= {469491.17,77866.909} bytesPerSecond
> RESULT available_send_bandwidth: smoke_test/bob= {642924.83,168842.34} bytesPerSecond
> RESULT transmission_bitrate: smoke_test/bob= {626115.5,294783.56} bytesPerSecond
> RESULT retransmission_bitrate: smoke_test/bob= {0,0} bytesPerSecond
> RESULT actual_encode_bitrate: smoke_test/bob= {594235.33,297289.54} bytesPerSecond
> RESULT target_encode_bitrate: smoke_test/bob= {640463.5,167676.66} bytesPerSecond
> 
> Bug: webrtc:10138
> Change-Id: I0414055af0010b8fb4d909297e6da86d398157c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132703
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Tommi <tommi@webrtc.org>
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@google.com>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27760}

TBR=mbonadei@webrtc.org,mbonadei@google.com,ilnik@webrtc.org,tommi@webrtc.org,titovartem@webrtc.org

Change-Id: Ib0ef94331410d9b22b6425e4da412b75360fa2d9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10138
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134210
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27771}
2019-04-25 13:39:04 +00:00
8848229234 Add Video Bwe stats collection to DefaultVideoQualityAnalyzer.
This CL adds the possibility to collect the following Video BWE stats:
- available_send_bandwidth
- transmission_bitrate
- retransmission_bitrate
- actual_encode_bitrate
- target_encode_bitrate

Example of the output:

RESULT available_send_bandwidth: smoke_test/alice= {487754.33,87583.093} bytesPerSecond
RESULT transmission_bitrate: smoke_test/alice= {465779.17,212075.5} bytesPerSecond
RESULT retransmission_bitrate: smoke_test/alice= {20036,26326.751} bytesPerSecond
RESULT actual_encode_bitrate: smoke_test/alice= {418779.33,200486.03} bytesPerSecond
RESULT target_encode_bitrate: smoke_test/alice= {469491.17,77866.909} bytesPerSecond
RESULT available_send_bandwidth: smoke_test/bob= {642924.83,168842.34} bytesPerSecond
RESULT transmission_bitrate: smoke_test/bob= {626115.5,294783.56} bytesPerSecond
RESULT retransmission_bitrate: smoke_test/bob= {0,0} bytesPerSecond
RESULT actual_encode_bitrate: smoke_test/bob= {594235.33,297289.54} bytesPerSecond
RESULT target_encode_bitrate: smoke_test/bob= {640463.5,167676.66} bytesPerSecond

Bug: webrtc:10138
Change-Id: I0414055af0010b8fb4d909297e6da86d398157c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132703
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27760}
2019-04-25 09:37:54 +00:00
1845922d5a Introduce QualityMetricsReporter and implement network stats gathering
QualityMetricsReporter helps to keep network emulation framework and
peer connection level test framework separated. Also it provides
ability to gather statistics from any component around with
correlation with call start and end.

Bug: webrtc:10138
Change-Id: Ib3330a8d35481fde77fcf77d2271d6cfcf188fec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132718
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27759}
2019-04-25 09:36:50 +00:00
d0e0ed82af Use explicit TaskQueueFactory for FrameGeneratorCapturer in test/pc/e2e
This replaces the implicit usage of GlobalTaskQueueFactory

Bug: webrtc:10284
Change-Id: I04f04208c852d9e4917a9b1dc555c72bfc28fb7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133577
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27707}
2019-04-23 10:26:06 +00:00
76723ae836 Add API to get raw stats value from DefaultAudioQualityAnalyzer
Bug: webrtc:10138
Change-Id: I60601a47c8dd8f669297d91825fe057f2b3da634
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133565
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27685}
2019-04-18 12:04:06 +00:00
70f80e5962 Add support for creation of AEC dump during the test with PC framework.
Also add conversational speech into PC smoke test (with resource files).

Bug: webrtc:10138
Change-Id: I415a5565bc9146821476ffc60f57f47ed51f89c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132323
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27592}
2019-04-12 13:09:12 +00:00
f948eb66aa Implement DefaultAudioQualityAnalyzer.
The DefaultAudioQualityAnalyzer will read stats reports (temporarily
using the old PeerConnectionInterface::GetStats) and for each audio
stream it will collect some NetEq related stats.

When DefaultAudioQualityAnalyzer::Stop is invoked by the framework,
it will report the following metrics:
- expand_rate
- accelerate_rate
- preemptive_rate
- speech_expand_rate
- preferred_buffer_size_ms

Bug: webrtc:10138
Change-Id: Ie493456fcb9ed86455b12dabdab98a317387ef46
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125980
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27474}
2019-04-07 14:32:33 +00:00
07122bc87e Use TaskQueueForTest instead or TaskQueue in unittests
To avoid hidden dependency on GlobalTaskQueueFactory used to construct TaskQueue

Bug: webrtc:10284
Change-Id: Iaa08be2827198e16aeb5538ea188d54cab60c1d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128879
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27291}
2019-03-26 14:42:49 +00:00
8ea8dcbae6 Rename create_network_emulation_manager_api into create_network_emulation_manager
Bug: webrtc:10138
Change-Id: I2feaa1009739556ca87fe4d081d808fed3957479
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128877
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27288}
2019-03-26 13:09:22 +00:00
d57628fed4 Move API for PC e2e test framework to the public API folder
Bug: webrtc:10138
Change-Id: If60019c9a7afe4760f4292e722cbc5aa229f437b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127891
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27247}
2019-03-22 16:52:16 +00:00
94b57c044e Cleanup BUILD.gn files from imports like foo:foo
Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn
files.

Done with Sublime regex replace.
Find: \b([-a-zA-Z0-9_]+):+\1\b
In: *.gn
Replace with: \1

Bug: None
Change-Id: I40aba1b14face687a595b852ffe443cb20197611
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:05:28 +00:00
533a9fec55 Clean BUILD.gn files: remove extra :memory
Use //third_party/abseil-cpp/absl/memory instead of
//third_party/abseil-cpp/absl/memory:memory in BUILD.gn files.

Bug: None
Change-Id: I47c915f0847b102b37c5b38009c91b315cd3a1b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128615
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27222}
2019-03-21 12:09:50 +00:00
98aa44859b Move Params, InjectableComponents and classes around on the top level
Bug: webrtc:10138
Change-Id: I3ee489c5558f9acad30587dc774ed240e115640e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128608
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27210}
2019-03-20 14:45:05 +00:00
ba82e0020d Add API to schedule environment changing actions during test in PC E2E framework
Bug: webrtc:10138
Change-Id: Ieebeec823829eb9dcaba4c31e7e9e998814982e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126463
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27153}
2019-03-15 20:44:01 +00:00
7bf8c7f8cc Add public API for NetworkEmulationManager
Bug: webrtc:10138
Change-Id: Ib5f8e95761813bd117a5e29adbc6822a5c6c73bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126122
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27146}
2019-03-15 14:50:59 +00:00
208634763a Move creation of rtc::NetworkManager into network emulation layer
Bug: webrtc:10138
Change-Id: I64271fab46a8dccb09f255eb14a4404b0bccdea3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127285
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27097}
2019-03-13 12:52:49 +00:00
d36c08623d Add support for simulcast streams in QualityAnalyzingVideoDecoder.
In QualityAnalyzingVideoEncoder all encoded images that belongs to
unrelated simulcast streams will be marked as to be discarded. So
to support simulcast streams QualityAnalyzingVideoDecoder have to return
black frames when all encoded images in received concatenated encoded
image are marked as to be discarded. Also QualityAnalyzingVideoDecoder
shouldn't pass such encoded image into VideoQualityAnalyzerInterface.

Bug: webrtc:10138
Change-Id: I0f793a7dc04b5d6b10949479bd074b2db86c5c6f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125460
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26973}
2019-03-05 16:33:14 +00:00
859abef68c Use DefaultVideoQualityAnalyzer as default, cleanup headers.
Bug: webrtc:10138
Change-Id: I2435b22e4e2cc2d2bfe6fd537494bdba539bb367
Reviewed-on: https://webrtc-review.googlesource.com/c/125092
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26920}
2019-03-01 10:42:22 +00:00
fc52b912a3 Implicitly suppress //build/config/clang:find_bad_constructs.
Since there is no way to enable/disable these diagnostics at runtime,
this CL moves the suppression into the rtc_* templates in order to
remove the need to explicitly add the snippet of code needed to
suppress it (currently copy/pasted in 144 locations).

The diagnostic that causes the most problems is the one about "complex
class/struct explicit ctor/dtor" [1] because WebRTC doesn't find
it useful enough.

Other diagnostics are good (for example the one that warns about
using "virtual" instead of "override", but that will be covered by
this clang-tidy check [2]) while others are Chromium related so
they have never triggered.

[1] - https://cs.chromium.org/chromium/src/tools/clang/plugins/FindBadConstructsConsumer.cpp?l=147-167&rcl=b4bebe1aa15dba7ca5fcc6456a81a55665327c3a
[2] - https://clang.llvm.org/extra/clang-tidy/checks/modernize-use-override.html

Bug: webrtc:163
Change-Id: Icbf27efa5b369100a31e6a32df1a0913729b3b34
Reviewed-on: https://webrtc-review.googlesource.com/c/125088
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26918}
2019-03-01 10:18:17 +00:00
3830d9b143 Fix peerconnection_quality_test #includes and deps.
Bug: webrtc:10138
Change-Id: I84413260dcda0e0c9e0790e13c5da35af706dd3d
Reviewed-on: https://webrtc-review.googlesource.com/c/124987
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26917}
2019-03-01 09:11:58 +00:00
12ae4f4d50 Introduce possibility to poll stats and notify analyzers.
This CL introduces the possibility to poll the 2 peer connections
at constant intervals.

It also introduces a dummy AudioQualityAnalyzer that will have to
be implemented in a follow-up CL and it moves every type of the
test framework inside the webrtc::test namespace.

Bug: webrtc:10138
Change-Id: I40acf7894bd67ea5229baba2d2cf18cd8ef65e67
Reviewed-on: https://webrtc-review.googlesource.com/c/123441
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26854}
2019-02-26 14:43:31 +00:00
ce7a4fb67b Adding possibility to save an RTCEventLog of the call.
This CL introduces the possibility to save an RTCEventLogs from the
call in order to do further analysis and call debugging.

Bug: webrtc:10138
Change-Id: If95ef66ecf52218b34ce01a4bcf8ab7991b04e5b
Reviewed-on: https://webrtc-review.googlesource.com/c/123881
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26838}
2019-02-25 11:38:19 +00:00
32232e92f3 Add spatial layers support to video analyze pipeline.
To support analyze of spatial layers we will continue sending them
into the network on encoder side, but will mark which should be then
discarded and which should be processed. On decoder side we will drop
layers, if they should be discarded and decode only parts, that
should be processed.

Bug: webrtc:10138
Change-Id: Ic8b8fe7787674c0ec49b879fcc29e54e8e3d787f
Reviewed-on: https://webrtc-review.googlesource.com/c/123185
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26784}
2019-02-20 22:05:15 +00:00
0b2150c884 Add a task queue into pc e2e fixture implementation
Bug: webrtc:10138
Change-Id: I0337df78c601cac2b5f2749e15369bd87221134d
Reviewed-on: https://webrtc-review.googlesource.com/c/123446
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26762}
2019-02-20 10:12:48 +00:00
713188010b Don't block the signaling thread during the call.
Since WebRTC stats are collected on the signaling thread, this CL moves
the wait from the signaling thread to the main thread.

Bug: webrtc:10138
Change-Id: I0e554fe82e3a4afe66b45e53032b06d533f54a39
Reviewed-on: https://webrtc-review.googlesource.com/c/123228
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26746}
2019-02-18 18:21:52 +00:00
6b88a8f161 Introduce default video quality analyzer
This implementation won't support spatial layers and simulcast. It will
be added in next CLs.

Bug: webrtc:10138
Change-Id: I08baef36fb15b8d2d2fa222c761d40508de7ff61
Reviewed-on: https://webrtc-review.googlesource.com/c/121944
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26676}
2019-02-14 09:08:31 +00:00
a6a273db11 Introduce PeerConnectionE2EQualityTestFixture implementation.
Introduce PeerConnectionE2EQualityTestFixture implementation with
example test.

Bug: webrtc:10138
Change-Id: Iec1d135f1b43863a3fa6f0723b579d2b7ff44807
Reviewed-on: https://webrtc-review.googlesource.com/c/120810
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26589}
2019-02-07 17:07:39 +00:00
840b05587f Introduce TestPeer.
TestPeer represent single participant in the call and will own most
required for call objects.

TestPeer::CreateTestPeer is responsible for full setup of TestPeer and
allow to correctly inject media analyzers into call.

Bug: webrtc:10138
Change-Id: Ide7062004b0dc113b9c05181d8144797a3cc27a8
Reviewed-on: https://webrtc-review.googlesource.com/c/119941
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26464}
2019-01-30 10:05:54 +00:00
f50c6c2fb4 Introduce VideoQualityAnalyzerInjectionHelper.
VideoQualityAnalyzerInjectionHelper will be used to provide all required
entities to inject video quality analyzer into peer connection pipeline.

Bug: webrtc:10138
Change-Id: Iea7cf453311d809619839d5cf94b78a020ce9167
Reviewed-on: https://webrtc-review.googlesource.com/c/119642
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26395}
2019-01-24 17:11:21 +00:00
fc2175da73 Introduce QualityAnalyzingVideoEncoder and QualityAnalyzingVideoDecoder.
This encoder will be used to inject VideoQualityAnalyzerInterface into
VideoEncoder, so it will be able to measure its metrics and also trace
frames from capturing on one peer side to rendering on another peer side.
The decoder will be used for the same purpose but in VideoDecoder pert.

Bug: webrtc:10138
Change-Id: Idf719753e3c0b3b1369ff206365bf0558705eb98
Reviewed-on: https://webrtc-review.googlesource.com/c/117363
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26381}
2019-01-24 11:15:12 +00:00
24a164bdcf Introduce ExampleVideoQualityAnalyzer.
This analyzer will be used in implementatino of peer connection level
test framework before main analyzer will be implemented.

Bug: webrtc:10138
Change-Id: Ibb7c5cd94b0f07c6fc5a2415f04b0f0ae7ae75e2
Reviewed-on: https://webrtc-review.googlesource.com/c/117221
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26301}
2019-01-17 16:50:38 +00:00
4895b45703 Introduce EncodedImageIdInjector.
EncodedImageIdInjector is responsible for injection of frame id into
encoded image before it will be sent to the transport layer. It will
help to track video frame from capturing on 1st peer side to rendering
on 2nd peer side and will make it possible to calculate video quality
stats between these frames.

This CL also introduces two different implementations for injector:
  1. DefaultEncodedImageIdInjector will prepend all encoded images with
     extra data and then will restore them on another side. This injector
     can work even if peers are running on different devices.
  2. SingleProcessEncodedImageIdInjector can work only when all peers
     are running in the same process, but won't use any extra data
     to propagate frame id between peers, so it won't affect any
     transport level metrics and bitrate estimator.

This CL is first part of new video quality analyzer for end-2-end
peer connection level test framework.

Bug: webrtc:10138
Change-Id: I77defc8e8c95cb244a695a9732980a47bd7a2e9b
Reviewed-on: https://webrtc-review.googlesource.com/c/116682
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26251}
2019-01-14 17:59:42 +00:00