Commit Graph

23894 Commits

Author SHA1 Message Date
69b03e960e Fix iOS demo H264 profile display.
Bug: none
Change-Id: I5e770f0b6b90822f4ab00432f0cbebc510d46a48
Reviewed-on: https://webrtc-review.googlesource.com/96360
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24462}
2018-08-28 08:15:07 +00:00
5cd81cbff7 AEC3: Disabling explicit handling of microphone gain changes
Disables the faster filter adaptation in the event of
microphone gain changes as it sometimes impacted transparency
negatively.

Bug: webrtc:9526,chromium:863826
Change-Id: I48fb6dd45440518aaf94b6469d6bb891247ea4ab
Reviewed-on: https://webrtc-review.googlesource.com/95143
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24461}
2018-08-28 07:26:40 +00:00
9ed9792def AEC3: Removing some old kill switches
Removing the some kill switches from the AEC3 codebase. CL is tested for
bit exactness.

Bug: webrtc:8671
Change-Id: I6ecdb1b5ccb05dca79bf0a0cd471f53d79d71d7e
Reviewed-on: https://webrtc-review.googlesource.com/96181
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24460}
2018-08-28 06:59:42 +00:00
d8111e2169 Delete unused class MockPeerConnection.
It appears to be unused since cl
https://webrtc-review.googlesource.com/46940.

Also note that there's a recently added class
MockPeerConnectionInterface, under api/test/, which may be more
suitable for new tests.

Bug: None
Change-Id: I6cd9bd2ec8847605f478663b709cd80c54895707
Reviewed-on: https://webrtc-review.googlesource.com/95421
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24459}
2018-08-27 20:02:14 +00:00
0673bc9204 Revert CLs affecting video quality toolchain.
Speculatively fixes Chromium test for cut: crbug.com/877968

Reverts CLs:
https://webrtc-review.googlesource.com/c/src/+/94772
https://webrtc-review.googlesource.com/c/src/+/95648
https://webrtc-review.googlesource.com/c/src/+/94773
https://webrtc-review.googlesource.com/c/src/+/96000
https://webrtc-review.googlesource.com/c/src/+/95949

Revert "Add Y4mFileReader"

This reverts commit 404be7f302358e6be16aadeba8bc8f8aba348c0f.

Revert "Remove SequencedTaskChecker from Y4mFileReader"

This reverts commit 1b5e5db842971340eb9128985ddbaf0225a9d0b1.

Revert "Add tool for aliging video files"

This reverts commit b2c0e8f60fad10e2786e5e131136a0da1299d883.

Revert "Reland "Update video_quality_analysis to align videos instead of using barcodes""

This reverts commit 9bb55fc09b6bfa00cba7779c37ad6c39b4206f7a.

Revert "Fix a bug in barcode_decoder.py"

This reverts commit 5c2de6b3ce079cff52c411a2c02ce6553a38dc79.

TBR=magjed@webrtc.org, phoglund@webrtc.org, phensman@webrtc.org

Bug: chromium:877968, webrtc:9642
Change-Id: I784d0598fd0370eec38d758b9fa0b38e4b3423be
Reviewed-on: https://webrtc-review.googlesource.com/96320
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24458}
2018-08-27 16:50:54 +00:00
f9fc171568 Add rtt_mult_experiment to evaluate video robustness vs. latency
Bug: webrtc:9670
Change-Id: Idb4ca130bfa652b2d0bddb5bee9ed8e34c97150a
Reviewed-on: https://webrtc-review.googlesource.com/96060
Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24457}
2018-08-27 15:51:52 +00:00
8227659075 Prevent duplicates in VP9 RTP p_diff list.
VP9 encoder wrapper duplicated p_diff values in VP9 RTP payload
descriptor for some frames. This confused ref frame finder which
invalidated such frames on receiver.

Bug: webrtc:9657
Change-Id: I5ddfed30d2908c5fbb8bce9d3dac0a519830e978
Reviewed-on: https://webrtc-review.googlesource.com/95701
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24456}
2018-08-27 14:38:59 +00:00
40f876d33e Simplify assertion.
Bug: None
Change-Id: I23830c5fe91af2e6ae736176c75b6d64270111d5
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/96241
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24455}
2018-08-27 14:36:40 +00:00
7f2eab0c7e Rename VideoQualityTestFixtureInterface::Params.pipe into config.
Also make it optional and use default value, if optional is not
specified. It is done also for next refactoring, that will introduce
ability to override network simulation layer.

Bug: webrtc:9630
Change-Id: I88cf1f9c70857f3299b5c3e9580a98570768e129
Reviewed-on: https://webrtc-review.googlesource.com/96121
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24454}
2018-08-27 14:17:32 +00:00
c8e202f3fc Minor improve od documentation for network simulation.
Bug: webrtc:9630
Change-Id: I03827b890ab73662117864c16c59f15a9ae3aac8
Reviewed-on: https://webrtc-review.googlesource.com/96200
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24453}
2018-08-27 14:16:27 +00:00
657f2e6c3e AEC3: audioproc_f: adding the read of the parameter fixed_capture_delay_samples
Bug: webrtc:8671
Change-Id: Ibbf1a725c1ec3a26879ab4feb2a655ed1460b359
Reviewed-on: https://webrtc-review.googlesource.com/96220
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24452}
2018-08-27 14:15:22 +00:00
82ec0faf72 Limiter reset when fixed gain controller gain set.
When FixedGainController::SetGain() is called first on a large value (e.g., 40 dB)
and afterwards on a smaller one (e.g., 0 dB), the limiter used by FixedGainController
takes time (about 10-20 seconds) to converge. During that period, the audio is not
audible and the volume slowly increases.

Even if switching from 40 dB to 0 dB is unlikely, this behavior can be corrected by
resetting the limiter every time that FixedGainController::SetGain() is called.
This eliminates the undesired effect described above even when the transient is short.

Bug: webrtc:7494
Change-Id: I419b8986d2181448b4671cdbbd1c256dfb460216
Reviewed-on: https://webrtc-review.googlesource.com/94902
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24451}
2018-08-27 14:06:32 +00:00
b0588e6368 Reland "Move FakeCodec to separate target and behave like real encoder."
Reland after fixes for ramp-up-tests

original reviewed on: https://webrtc-review.googlesource.com/95182

TBR=mbonadei@webrtc.org,ilnik@webrtc.org,sprang@webrtc.org,srte@webrtc.org,perkj@webrtc.org,titovartem@webrtc.org

Bug: none
Change-Id: I4f9af0c98a0341dd4fadd5184bb85d7511efbdc0
Reviewed-on: https://webrtc-review.googlesource.com/96120
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24450}
2018-08-27 13:09:37 +00:00
bf2b620cc8 Convert VP8 descriptor to generic descriptor.
Also adds a running picture id for the old generic format when
kVideoCodecGeneric is used (behind "WebRTC-GenericPictureId" field trial).

Bug: webrtc:9361
Change-Id: I6f232a2663bb60257c97ed3473eb07044d325b90
Reviewed-on: https://webrtc-review.googlesource.com/94842
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24449}
2018-08-27 13:03:20 +00:00
fde4aa9909 AEC3: Adaptive handling of echo path with strong high-frequency gain
This CL adds adaptive handling of platforms where the echo path has
a strong gain above 10 kHz. A configurable offset is adaptively applied
depending on the amount of echo and mode of the echo suppressor.

Bug: webrtc:9663
Change-Id: I27dde6dc23b04a76a3be8c49d7fc9e226b9137e6
Reviewed-on: https://webrtc-review.googlesource.com/95947
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24448}
2018-08-27 12:49:28 +00:00
01a89904c0 Remove deprecated type alias for RtpVideoCodecTypes.
First phase of this removal landed with cl https://webrtc-review.googlesource.com/79561

Bug: webrtc:8995
Change-Id: I9dc152e2f1bac17e2959af7e18106760ca5435c8
Reviewed-on: https://webrtc-review.googlesource.com/95720
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24447}
2018-08-27 12:09:57 +00:00
638d4d375f AEC3: No ERLE uncertainty with diverged filter
Disable the use of ERLE uncertainty with a diverged filter as it has
been shown to make transparency worse.

Bug: webrtc:9668
Change-Id: I5e23665def187c0d1cf47a029c4ebc950e79bb44
Reviewed-on: https://webrtc-review.googlesource.com/96140
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24446}
2018-08-27 12:06:43 +00:00
40b7050ac6 Add copy and move constructors to RateStatistics.
Bug: none
Change-Id: I589a7f202ee1c4b8c06e8f44aa570c47d066ab72
Reviewed-on: https://webrtc-review.googlesource.com/95647
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24445}
2018-08-27 11:28:37 +00:00
ec4a060a55 Revert "Delete leftover includes and declarations for MediaConstraintsInterface"
This reverts commit a1e4ae23715867eca58488be307759ffa5901463.

Reason for revert: Breakage in downstream code still using constraints.

Original change's description:
> Delete leftover includes and declarations for MediaConstraintsInterface
> 
> Bug: webrtc:9239
> Change-Id: I1f49d6847572faecd6022a44222d6271302fe443
> Reviewed-on: https://webrtc-review.googlesource.com/95721
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24442}

TBR=kwiberg@webrtc.org,nisse@webrtc.org,hta@webrtc.org

Change-Id: Idbef4c57a0d3b82e94a431c5407a86c9fcd4be41
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9239
Reviewed-on: https://webrtc-review.googlesource.com/96160
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24444}
2018-08-27 11:26:42 +00:00
b615d1af90 Add lock annotations in StreamStatisticianImpl
Also consolidate lock operations to public methods only, moving one
CritScope out of UpdateCounters (private) up to IncomingPacket
(public).

Bug: webrtc:7135
Change-Id: I458857d3cfa49961fa34196ffe02cdefd83cec10
Reviewed-on: https://webrtc-review.googlesource.com/96122
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24443}
2018-08-27 11:06:40 +00:00
a1e4ae2371 Delete leftover includes and declarations for MediaConstraintsInterface
Bug: webrtc:9239
Change-Id: I1f49d6847572faecd6022a44222d6271302fe443
Reviewed-on: https://webrtc-review.googlesource.com/95721
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24442}
2018-08-27 10:41:57 +00:00
c30a13147c Add experiment for boosted qp at lowest stream for screenshare
Bug: webrtc:9659
Change-Id: I2320afc393d6a78ae03a4f447f0e3333dd5748c4
Reviewed-on: https://webrtc-review.googlesource.com/95943
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24441}
2018-08-27 10:04:50 +00:00
764c14c87d Delete unused AudioCodingModule methods.
Methods deleted: IsCodecValid (static), QueryEncoder, SendFrequency.

Bug: None
Change-Id: Id63ea7cdc364583e896d3301d04fa9caae1e4d94
Reviewed-on: https://webrtc-review.googlesource.com/95486
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24440}
2018-08-27 10:00:59 +00:00
18f1adc0da Delete AudioCodingModule::LeastRequiredDelayMs and related NetEq code.
Bug: None
Change-Id: I2f68502d19415899b3694f7bf5da523da831b223
Reviewed-on: https://webrtc-review.googlesource.com/95640
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24439}
2018-08-27 09:58:19 +00:00
8d92e8d323 Revert "Reland "Move FakeCodec to separate target and behave like real encoder.""
This reverts commit f2a8287cc5bbe982cc008d0550df83533623b780,
original reviewed on: https://webrtc-review.googlesource.com/95182

Reason for revert: Breaks ramp-up tests

TBR=mbonadei@webrtc.org,ilnik@webrtc.org,sprang@webrtc.org,srte@webrtc.org,perkj@webrtc.org,titovartem@webrtc.org

Bug: none
Change-Id: I11ddf8619c33cf93825088fd293bcdf11e8cedab
Reviewed-on: https://webrtc-review.googlesource.com/96083
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24438}
2018-08-27 09:19:33 +00:00
c3af97d68c Add a list of allowed and disallowed Abseil things
Bug: webrtc:8821
Change-Id: Ifb3bacd3403bbf823c78fff47571a83159f1da73
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/95880
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24437}
2018-08-27 08:37:20 +00:00
2afd281ec4 Adds debug printing of units in unit tests.
Bug: webrtc:9656
Change-Id: I643b79bc214643f47b2b64967ce713665dbef5c9
Reviewed-on: https://webrtc-review.googlesource.com/95652
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24436}
2018-08-27 08:28:12 +00:00
1946a3f0fe Add frame rate parameter to SpatialLayer struct.
This will allow us to configure VP9 encoder to produce spatial layers
with different frame rates.

Bug: webrtc:9650
Change-Id: I3a9c58072003b8a8da681d5291d8f7ede7f52fa4
Reviewed-on: https://webrtc-review.googlesource.com/95427
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24435}
2018-08-26 19:19:36 +00:00
524e878121 AEC3: Add state-specific suppressor behaviors
This CL allows selecting an echo suppressor behavior which is specific
for whether the nearend is dominant, or the echo is dominant.

The changes in this CL are bitexact.

Bug: webrtc:9660
Change-Id: Ie32e65efe47e692de6d6a22a7ad3b469d745fd6b
Reviewed-on: https://webrtc-review.googlesource.com/95725
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24434}
2018-08-24 21:43:36 +00:00
90ab76dd19 Adds support for JSON config in video_replay.
This change adds the ability to optionally provide configuration to the
video_replay binary through JSON. This is a merge of the changes provided by
philipel on this bug: https://bugs.chromium.org/p/chromium/issues/detail?id=840536#c1

However it has been updated to pull all the json parsing into the example binary
itself instead of it being integrated into the core library. Writing test cases
out to JSON configuration will be handled in a different CL. Most likely there
will be a utility class added that takes a Config and converts it to JSON that is
decoupled from the actual implementation.

Bug: webrtc:9609
Change-Id: Icc5900063d7f704825f224240e4b3787c06ca074
Reviewed-on: https://webrtc-review.googlesource.com/95320
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24433}
2018-08-24 18:33:00 +00:00
e583174d1e Optionally disable digital adaptive AGC2.
The AGC2 is enabled by flipping
AudioProcessing::Config::GainController2::enabled. The flag enables
both AdaptiveAgc and FixedGainController. Before this CL, there was no
way(*) to only enable the FixedGainController. After this CL, it's
also possible to flip the setting
|AudioProcessing::Config::GainController2::adaptive_digital_mode|. The
default is |true|, which is the previous behavior.

* Except for instantiating and setting it up outside of the APM like
  it's done in the AudioMixer.

Bug: webrtc:7494
Change-Id: I506e93b6687221ac467f083fa8db3d45c98c1b83
Reviewed-on: https://webrtc-review.googlesource.com/95426
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24432}
2018-08-24 15:54:43 +00:00
5c2de6b3ce Fix a bug in barcode_decoder.py
When converting from a .y4m file, it's illegal to pass a video_size
option since the resolution is already contained in the .y4m file.

TBR=phoglund@webrtc.org
NOTRY=TRUE

Bug: webrtc:9642
Change-Id: Iee7d2ba1332c45a1669af0fba43b0c3e7ce5846b
Reviewed-on: https://webrtc-review.googlesource.com/95949
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24431}
2018-08-24 15:36:46 +00:00
85f20cbe4a Add VideoEncoder::Settings::numberOfSimulcastStreams.
This helps Java encoders take action if simulcast is enabled or not.

Bug: webrtc:9646
Change-Id: Iad967e237bdc790ff2af111bdec1319f3e661ff7
Reviewed-on: https://webrtc-review.googlesource.com/95651
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24430}
2018-08-24 14:40:04 +00:00
9bb55fc09b Reland "Update video_quality_analysis to align videos instead of using barcodes"
This is a reland of d65e143801a7aaa9affdb939ea836aec1955cdcc

The binary for frame_analyzer.cpp is precompiled and stored in the cloud, so it
won't automatically pick up change to the source file. Therefore, restore all
old code to be backwards compatible.

Original change's description:
> Update video_quality_analysis to align videos instead of using barcodes
>
> This CL is a follow-up to the previous CL
> https://webrtc-review.googlesource.com/c/src/+/94773 that added generic
> logic for aligning videos. This will allow us to easily extend
> video_quality_analysis with new sophisticated video quality metrics.
> Also, we can use any kind of video that does not necessarily need to
> contain bar codes. Removing the need to decode barcodes also leads to a
> big speedup for the tests.
>
> Bug: webrtc:9642
> Change-Id: I74b0d630b3e1ed44781ad024115ded3143e28f50
> Reviewed-on: https://webrtc-review.googlesource.com/94845
> Reviewed-by: Paulina Hensman <phensman@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24423}

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

Bug: webrtc:9642
Change-Id: Id8d129ce103284504c67690f8363c03eaae3eee7
Reviewed-on: https://webrtc-review.googlesource.com/96000
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24429}
2018-08-24 13:21:18 +00:00
3e169ac18c Revert "Update video_quality_analysis to align videos instead of using barcodes"
This reverts commit d65e143801a7aaa9affdb939ea836aec1955cdcc.

Reason for revert: Breaks perf bots. frame_analyzer is a prebuilt binary, so it won't automatically pick up changes in the .cc file.

Original change's description:
> Update video_quality_analysis to align videos instead of using barcodes
> 
> This CL is a follow-up to the previous CL
> https://webrtc-review.googlesource.com/c/src/+/94773 that added generic
> logic for aligning videos. This will allow us to easily extend
> video_quality_analysis with new sophisticated video quality metrics.
> Also, we can use any kind of video that does not necessarily need to
> contain bar codes. Removing the need to decode barcodes also leads to a
> big speedup for the tests.
> 
> Bug: webrtc:9642
> Change-Id: I74b0d630b3e1ed44781ad024115ded3143e28f50
> Reviewed-on: https://webrtc-review.googlesource.com/94845
> Reviewed-by: Paulina Hensman <phensman@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24423}

TBR=phoglund@webrtc.org,magjed@webrtc.org,phensman@webrtc.org

Change-Id: Ia590b465687b861fe37ed1b14756d4607ca90da1
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9642
Reviewed-on: https://webrtc-review.googlesource.com/95946
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24428}
2018-08-24 12:45:13 +00:00
a1cceca02c Remove unused include of <android/log.h>.
Bug: None
Change-Id: I7e9673a1ac455cbab77e07fac47b93f4b96942cb
Reviewed-on: https://webrtc-review.googlesource.com/95646
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24427}
2018-08-24 12:10:22 +00:00
3288168e4f Enable video adaptation for all screenshare content
Since screenshare uses "Maintain resolution" degradation preference,
adapting should be enabled to reduce framerate if encoder can't keep up.

Bug: chromium:690537
Change-Id: I1f4418b7b7b4faa13f34d5353e3c625a59975c05
Reviewed-on: https://webrtc-review.googlesource.com/95460
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24426}
2018-08-24 12:02:03 +00:00
41dd22b15d AEC3: Removing more dead code from the suppressor
This CL removes the UpdateGainIncrease code that is not used anymore.
The CL has been tested for bit exactness.

Bug: webrtc:8671
Change-Id: I4fcf26c3b4b5bba760ee139416ddefac86a36c2e
Reviewed-on: https://webrtc-review.googlesource.com/95940
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24425}
2018-08-24 10:25:00 +00:00
fc1acd2364 Add support for enabling simulcast in "Plan B" using MediaConstraints.
BUG=webrtc:9655

Change-Id: Ieb5fe5d97b6d4381608a51593bca5423979d1b9f
Reviewed-on: https://webrtc-review.googlesource.com/95481
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24424}
2018-08-24 09:55:59 +00:00
d65e143801 Update video_quality_analysis to align videos instead of using barcodes
This CL is a follow-up to the previous CL
https://webrtc-review.googlesource.com/c/src/+/94773 that added generic
logic for aligning videos. This will allow us to easily extend
video_quality_analysis with new sophisticated video quality metrics.
Also, we can use any kind of video that does not necessarily need to
contain bar codes. Removing the need to decode barcodes also leads to a
big speedup for the tests.

Bug: webrtc:9642
Change-Id: I74b0d630b3e1ed44781ad024115ded3143e28f50
Reviewed-on: https://webrtc-review.googlesource.com/94845
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24423}
2018-08-24 09:26:14 +00:00
523b4c4330 Add unlimited retransmission experiment for screenshare
Bug: webrtc:9659
Change-Id: Idcdc647c112ed2c7c027a7a0056b145ce8f45788
Reviewed-on: https://webrtc-review.googlesource.com/95724
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24422}
2018-08-24 08:48:30 +00:00
cd634ce933 Remove client.webrtc.branches.
The associated master has been turned down.

Bug: chromium:877279
Change-Id: I39c3a43680288b17094d1300926ec7a8bd427509
Reviewed-on: https://webrtc-review.googlesource.com/95882
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24421}
2018-08-24 08:19:37 +00:00
e8a2e6c82f Remove assumption that all video codecs are known.
Bug: webrtc:9516
Change-Id: I810e9bfe556e6d2ccfeb7a35f7c6785c9909a0e2
Reviewed-on: https://webrtc-review.googlesource.com/94512
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24420}
2018-08-24 08:01:03 +00:00
a765c8208a Change some pointers to std::unique_ptr in rtp_rtcp tests.
Bug: none
Change-Id: Ia4e69e44bbda7b5b633b8be1779d105649f44930
Reviewed-on: https://webrtc-review.googlesource.com/94844
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24419}
2018-08-24 07:26:04 +00:00
e89dda7cb9 Roll chromium_revision c86aa801eb..ca3a5e1cbb (585622:585726)
Change log: c86aa801eb..ca3a5e1cbb
Full diff: c86aa801eb..ca3a5e1cbb

Changed dependencies:
* src/base: f0d28f609f..7bf1a620f7
* src/build: 96082b33ed..53a2dfe471
* src/ios: 2c47c67b49..59b5f34e52
* src/testing: 78b6a1338b..7484a964e5
* src/third_party: 14ff0932ca..a3a6c8af12
* src/third_party/depot_tools: d06cc78ec8..7b7eb8800b
* src/tools: 3e95bda3b3..5592370ea3
DEPS diff: c86aa801eb..ca3a5e1cbb/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: I708c452abe0c7ece70380a2bc2367d91d14fce7a
Reviewed-on: https://webrtc-review.googlesource.com/95904
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24418}
2018-08-24 07:05:57 +00:00
ecb2d5670d AEC3: Removing old suppressor logic
This CL removes some of the unused code in the suppressor. The CL has
been tested for bit exactness.

Bug: webrtc:8671
Change-Id: I960f9445dfd109cf1d5790debb8758872b5b8d0d
Reviewed-on: https://webrtc-review.googlesource.com/95682
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24417}
2018-08-24 06:34:42 +00:00
5a72a5ef2b Adding quiet mode for audioproc_f
This CL adds a quiet mode for audioproc_f and hooks up the verbose
output of the AEC3 settings read from the JSON input file to that.

Bug: webrtc:8671
Change-Id: I93bbd1efc6502649da7b2b3e9f7557e9c184b0ed
Reviewed-on: https://webrtc-review.googlesource.com/95700
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24416}
2018-08-24 05:52:43 +00:00
6fcdc2f708 Support domain name ICE candidates
Bug: webrtc:4165
Change-Id: Icc06bb13120080635cb722b8a8720e7d25426e2d
Reviewed-on: https://webrtc-review.googlesource.com/85540
Commit-Queue: Zach Stein <zstein@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24415}
2018-08-24 04:54:43 +00:00
c34731b1ef Roll chromium_revision 607c2083aa..c86aa801eb (585516:585622)
Change log: 607c2083aa..c86aa801eb
Full diff: 607c2083aa..c86aa801eb

Changed dependencies:
* src/base: 0752189994..f0d28f609f
* src/build: ce2f1999ca..96082b33ed
* src/ios: e6f6199fe8..2c47c67b49
* src/testing: c82106c804..78b6a1338b
* src/third_party: d9451e7310..14ff0932ca
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/a7fb87b3a3..7a1ed44d24
* src/third_party/depot_tools: 81db1d5032..d06cc78ec8
* src/tools: 47670771a0..3e95bda3b3
DEPS diff: 607c2083aa..c86aa801eb/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: Ie13899528934960def3eedbf2dcc0a0996265600
Reviewed-on: https://webrtc-review.googlesource.com/95840
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24414}
2018-08-23 23:20:50 +00:00
e4749c2bf0 The default logic for creating video bitrate allocator.
It is a mirror of `VideoCodecInitializer::CreateBitrateAllocator`

Bug: webrtc:9513
Change-Id: Ib2e83e9f757387a2f6f6101d5d21512f1d507a95
Reviewed-on: https://webrtc-review.googlesource.com/92320
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24413}
2018-08-23 20:50:32 +00:00