Commit Graph

505 Commits

Author SHA1 Message Date
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
85a9d91cd4 Add ability to set min/start/max bitrate on peer's PC in PC quality tests
Bug: webrtc:10138, webrtc:10692
Change-Id: I4d7ae84dc2945fef6451a6671786b3b19cd9abd8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139108
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28107}
2019-05-29 13:25:26 +00:00
232b6a16cc Propagate screenshare info into video track and it's source.
If screen share is set, then we need to tell video source, that it
is screen share source. Also video track should be aware, that it is
screen share track. It is required to choose proper video encoding
settings.

Bug: webrtc:10138
Change-Id: I5c82584ae0325a303a495554d87962a98b676694
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138278
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28098}
2019-05-29 10:14:22 +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
4d29ef063c Add periodic alive message logging to prevent test infra think, that test is dead
Bug: webrtc:10138
Change-Id: Ib39ff6df81776a7784687be2dc16ab81c500cc3e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137428
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28005}
2019-05-21 11:09:18 +00:00
7581ff7375 Add screen share support to PC level test framework
Bug: webrtc:10138
Change-Id: I1a8ac683e91f8061387f407610d7db2a6d0d4fe9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136805
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27950}
2019-05-15 14:07:00 +00:00
0379d8cfea Change a way, how receive stream is determined in DefaultAudioQualityAnalyzer.
Bug: webrtc:10138
Change-Id: I8955c30f0a5d98abeca029323396e469a2fb243b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136683
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27927}
2019-05-13 13:07:01 +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
630bd43fcf DefaultAudioQualityAnalyzer: use bytes_recv instead of packets_recv.
Bug: webrtc:10138
Change-Id: I2fa5d9da2dd7788ffc48cf6a4171eb3ce0de5423
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135461
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27866}
2019-05-07 14:59:02 +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
b93c4e622f Add propagation of test duration to PC framework user.
Add method to get real test execution time, where test execution time is
time from call setup to call terminated.

Bug: webrtc:10138
Change-Id: I7ae3995c0051ecb4fc796b895be1180c8aab77cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134302
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27822}
2019-05-02 10:20:26 +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
43f7002aff Delete DecodedImageCallback::ReceivedDecodedFrame
This was a companion method to ReceivedDecodedReferenceFrame, deleted
in https://webrtc-review.googlesource.com/c/src/+/133348.

Tbr: kwiberg@webrtc.org # Mock class update
Bug: webrtc:7408
Change-Id: I429f5f5c18f14c27136e82860297107a82c81d13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133571
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27754}
2019-04-25 08:09:29 +00:00
7e2f6b7d92 Fix reporting of metrics when there is no video in the test.
Before if there is no video in PC quality test video quaity analyzer
failed on RTC_CHECK becuase of empty counter. Now it will report no
metrics and print 0 in debug logging.

Bug: webrtc:10138
Change-Id: If6656a613465c522cac1d4b2e4dd455e409229ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133886
Reviewed-by: Artem Titarenko <artit@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27731}
2019-04-24 08:58:55 +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
2d87a50b46 Only process cross traffic simulation if added.
This avoids extra processing overhead when there's no cross traffic
simulation active.

Bug: webrtc:10365
Change-Id: I8c8ae2fb823a47af2406e62ae350ebb551169646
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133620
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27705}
2019-04-23 10:04:07 +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
2317c5ee0a Delete method DecodedImageCallback::ReceivedDecodedReferenceFrame
The code invoking it was deleted in
https://codereview.webrtc.org/2753783002

Tbr: kwiberg@webrtc.org # Change to mock class in api/test
Bug: webrtc:7408
Change-Id: I576d7aacd7dc60e42a05d2ea837fddf16594e685
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133348
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27680}
2019-04-18 08:14:40 +00:00
153056b059 Add ability to play audio in circle for TestAudioDevice wav file capturer
Also use this ability in PC smoke test.

Bug: webrtc:10138
Change-Id: I83d526344f203082a19377d9642c9e453454f7ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133163
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27649}
2019-04-16 15:33:03 +00:00
6a0aad9260 Temporary switch back to generated audio to fix test flakes
Wav file capturer won't repeat file or produce silence after file end and
WebRTC pipeline will crash in such case. In future we need to make it
possible to continue audio after file was ended to behalf in the same
way as video files capturer.

Bug: webrtc:10138
Change-Id: I35f5bd33790cd430a56002a44af0abb894a96d29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132795
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27609}
2019-04-15 09:22:28 +00:00
16cb8f5d74 Reland "Replace usage of old SetRates/SetRateAllocation methods"
This is a reland of 7ac0d5f348f0b956089c4ed65c46e65bac125508

Original change's description:
> Replace usage of old SetRates/SetRateAllocation methods
>
> This rather large CL replaces all relevant usage of the old
> VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
> API is unchanged to allow downstream projects to update without
> breakage.
>
> Bug: webrtc:10481
> Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27554}

TBR=brandtr@webrtc.org,sakal@webrtc.org,perkj@webrtc.org

Bug: webrtc:10481
Change-Id: I2978d5c527a18e885b7845c4e53a2424e8ad5b4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132551
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27593}
2019-04-12 13:37:32 +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
7aacdd9515 Reland "Delete CodecSpecificInfo argument from VideoDecoder::Decode"
This is a reland of 39d3a7de02d63894d12e7332322e1d80cd7c0d40

Original change's description:
> Delete CodecSpecificInfo argument from VideoDecoder::Decode
>
> Bug: webrtc:10379
> Change-Id: I079b419604bf4e9c1994fe203d7db131a0ccddb6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125920
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27022}

Tbr: kwiberg@webrtc.org
Bug: webrtc:10379
Change-Id: I8197bebd2ae7dc460644a98795b8257b033c27c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126480
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27565}
2019-04-11 13:03:52 +00:00
7ddef1af88 Revert "Replace usage of old SetRates/SetRateAllocation methods"
This reverts commit 7ac0d5f348f0b956089c4ed65c46e65bac125508.

Reason for revert: <INSERT REASONING HERE>

Original change's description:
> Replace usage of old SetRates/SetRateAllocation methods
> 
> This rather large CL replaces all relevant usage of the old
> VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
> API is unchanged to allow downstream projects to update without
> breakage.
> 
> Bug: webrtc:10481
> Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27554}

TBR=brandtr@webrtc.org,sakal@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,perkj@webrtc.org

Change-Id: I576760b584e3f258013b0279c0c173c895bbb37e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10481
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132561
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27559}
2019-04-11 10:50:29 +00:00
7ac0d5f348 Replace usage of old SetRates/SetRateAllocation methods
This rather large CL replaces all relevant usage of the old
VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
API is unchanged to allow downstream projects to update without
breakage.

Bug: webrtc:10481
Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27554}
2019-04-11 07:46:09 +00:00
8b9f51132a Add stream labels into PeerConnection level smoke test.
Bug: webrtc:10138
Change-Id: I779a8eb0516471273ddb313df3c2a34e66dff869
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131396
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27479}
2019-04-08 09:58:51 +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
1770ddb7a4 Remove obsolete TODO in default_encoded_image_data_injector.h
Bug: webrtc:10138
Change-Id: I401dea607feb2f1ea967dd2473bf4f0ba7a7d43e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131280
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27446}
2019-04-04 09:31:16 +00:00
ade945d834 Add ability to specify encoder bitrate multiplier in PC level tests
Bug: webrtc:10138
Change-Id: I40b42e83ccec7b08226606d2770f3afa80e3fcc6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130241
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@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@{#27440}
2019-04-03 12:09:42 +00:00
e5cc85b5c5 Introduce dynamic endpoints
Bug: webrtc:10138
Change-Id: I7f6922adb93680cada6bea014539fc3089735834
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128480
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27336}
2019-03-28 12:27:41 +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
89cc7d43e6 Add logging of internal stats into default video quality analyzer
Bug: webrtc:10138
Change-Id: I2ce0837baee4719bb571a989a850003e6521cfca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128874
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27286}
2019-03-26 12:50:16 +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
741daaf039 Move rtc::FunctionView to the public API
Bug: webrtc:10138
Change-Id: Icc25a2a277a9608701aaddd546882366739991ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127898
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27227}
2019-03-21 15:23:05 +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
d09bc55d3b Introduce new API for runnig PC e2e test fixture
Bug: webrtc:10138
Change-Id: I704f09843e5b8a05de4a1d25a4baa44c683a5552
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128402
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@{#27204}
2019-03-20 11:23:29 +00:00
0b44314b76 Move PC e2e test framework into its own namespace
Bug: webrtc:10138
Change-Id: I7fc02967058d3c53da73e280a7a1533a0860ba4c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128403
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27203}
2019-03-20 11:12:59 +00:00
82b7ff5797 Don't store last rendered frame in DefaultVideoQualityAnalyzer
Bug: webrtc:10138
Change-Id: I1aec9b8e5abbc202cafcf7db74aba89182612308
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128578
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27183}
2019-03-19 16:39:07 +00:00
4fa9eded8e Refactor DefaultEncodedImageDataInjector to let EncodedImage own the data.
Bug: webrtc:9378
Change-Id: I930935b6d4759dfbdb03a4ca2728a8b637997045
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128577
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27181}
2019-03-19 16:13:17 +00:00
80cfd81483 Move PeerConnectionComponents when creating PeerConnectionDependencies.
Move PeerConnectionComponents when creating PeerConnectionDependencies
instead of passing them by pointer in test_peer.cc in PC e2e test
framework

Bug: webrtc:10138
Change-Id: I490f576c6af3eab42df04ba597945e66a87880e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128579
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27180}
2019-03-19 16:04:27 +00:00
276cdfcf6a Rename resolution_of_encoded_image into resolution_of_rendered_frame.
Rename resolution_of_encoded_image into resolution_of_rendered_frame in
DefaultVideoQualityAnalyzer to make it consistent with the way, how it
is calculated.

Bug: webrtc:10138
Change-Id: Ibf89f08ac0646b57b4a6b8316cec1ed73bad02a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128576
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27179}
2019-03-19 15:41:22 +00:00