408 Commits

Author SHA1 Message Date
Erik Språng
dbdd8395f7 Add ability for VideoEncoder to signal frame rate allocation.
This CL add new data to the VideoEncoder::EncoderInfo struct, indicating
how the encoder intends to allocate frames across spatial and temporal
layers.

This metadata will be used in upcoming CLs to control how the encoder's
rate controller performs.

Bug: webrtc:10155
Change-Id: Id56fae04bae5f230d1a985171097d7ca83a3be8a
Reviewed-on: https://webrtc-review.googlesource.com/c/117900
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26300}
2019-01-17 15:40:53 +00:00
Erik Språng
f93eda1705 Move some video codec constants to separate file.
kMaxSimulcastStreams, kMaxSpatialLayers and kMaxTemporalStreams don't
really beling on VideoBitrateAllocation.
common_types.h is going away and it feels dubious to requrie include
of the full VideoEncoder api to use them. Therefore moving them into a
seprate file/target.

Also includes some remaining cleanup of includes.

Bug: webrtc:9271
Change-Id: I7ded3d97a9a835ac756159700774445a2b93a697
Reviewed-on: https://webrtc-review.googlesource.com/c/117305
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26299}
2019-01-17 15:29:53 +00:00
Niels Möller
fb7a6a7f8d Delete test WebRtcVideoChannelTest.AdaptsOnOveruseAndChangeResolution
It only tests the videoadaptation in the test class FakeVideoCapturer.

Bug: webrtc:6353
Change-Id: I4766eebc5cfa7412fde9fd6e5173f1b2381a5d87
Reviewed-on: https://webrtc-review.googlesource.com/c/118042
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26297}
2019-01-17 14:23:52 +00:00
Emircan Uysaler
7c03bdc1d3 Reland "Add a high bitrate full stack test with fake codec"
In this reland, I disabled high bitrate webrtc perf test on Android32.

This is a reland of 15df2774f4e85cf8900768c1793edcf17d651dcd

Original change's description:
> This CL adds a fake codec factory  in WebRTC that can be used in tests to
> produce target bitrate output.

> We also add a high bitrate test that makes use of fake codec. This test assumes
> ideal network conditions with target bandwidth being available and exercises
> WebRTC calls with a high target bitrate(100 Mbps) end-to-end.

TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,emircan@webrtc.org,kron@webrtc.org

Bug: chromium:879723
Change-Id: I31a4b48d986bef9ca003ae71afeb567ae3e562c9
Reviewed-on: https://webrtc-review.googlesource.com/c/117980
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26285}
2019-01-16 21:03:22 +00:00
Niels Möller
dcc70297cd Simplify WebRtcVideoChannelTest.PreviousAdaptationDoesNotApplyToScreenshare
Test wiring to DegradationPreference passed to
VideoSendStream::SetSource, but not the adaptation implemented in the
test class FakeVideoCapturer.

Bug: webrtc:6353
Change-Id: Iec2ae89283fb856822ea2829db17eaa02337b467
Reviewed-on: https://webrtc-review.googlesource.com/c/117641
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26278}
2019-01-16 13:02:49 +00:00
Chen Xing
0acffb5b36 Expose jitterBufferEmittedCount in addition to the existing jitterBufferDelay for getStats().
NetEq currently only passes `jitterBufferDelay` to `getStats()`. We need its paired `jitterBufferEmittedCount` denominator stat for the calculations to be accurate.

Bug: webrtc:10192
Change-Id: I655aea629026ce9101409c2e0f18c2fa57a1c3ab
Reviewed-on: https://webrtc-review.googlesource.com/c/117320
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#26276}
2019-01-16 11:44:10 +00:00
Mirta Dvornicic
ccc1b57e32 Poll is_hardware_accelerated from VideoEncoder instead of VideoEncoderFactory.
Currently, CPU overuse settings for HW encoders are sometimes being used
even though the actual encoder is a SW encoder, e.g. in case of SW fallback
when the encoder is initialized. Polling is_hardware_accelerated after the
encoder has been created and initialized will improve choosing the correct
CPU overuse settings.

Bug: webrtc:10065
Change-Id: Ic6bd67630a040b5a121c13fa63dd074006973929
Reviewed-on: https://webrtc-review.googlesource.com/c/116688
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26266}
2019-01-15 14:12:12 +00:00
Niels Möller
45ccd8488e Don't set the screenshare flag on FakeVideoCapturerWithTaskQueue
A capturer with this flag was set in
WebRtcVideoChannelTest.PreviousAdaptationDoesNotApplyToScreenshare.
But the flag is used only by the VideoCapturerTrackSource class, which
isn't used in this test.

Bug: webrtc:6353
Change-Id: I58058c882c5a65b5cfa9921e302c422c8ccb20a9
Reviewed-on: https://webrtc-review.googlesource.com/c/117561
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26256}
2019-01-15 10:22:13 +00:00
Anders Carlsson
45340ca824 Remove legacy video codec factories.
Removes the deprecated video codec factories and the related flag and
helper classes.

Bug: webrtc:7925
Change-Id: I0a6d1666ece9ad074fefc79b626ba241765e1b98
Reviewed-on: https://webrtc-review.googlesource.com/c/113940
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26245}
2019-01-14 14:56:40 +00:00
Niels Möller
d0f0f68953 Refactor WebRtcVideoEngine tests to not use cricket::VideoCapturer, part 1.
Replaced with a combination of cricket::FakeFrameSource and
webrtc::test::FrameForwarder. This cl converts the first three
affected tests, the rest will follow.

Bug: webrtc:6353
Change-Id: I556f6b58f4ca81234ffae3dc6e1319f9c60a76ae
Reviewed-on: https://webrtc-review.googlesource.com/c/117260
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26239}
2019-01-14 12:34:51 +00:00
Steve Anton
aec15aa810 (5) Rename files to snake_case: install forwarding headers
Mechanically generated with this command:

tools_webrtc/do-renames.sh install all-renames.txt && git cl format

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: Ic8e99f71f2da62e5c99863c6d24a8cfe311466cd
Reviewed-on: https://webrtc-review.googlesource.com/c/115682
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26227}
2019-01-11 17:13:36 +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
1c05765831 (3) Rename files to snake_case: move the files
Mechanically generated with this command:

tools_webrtc/do-rename.sh move all-renames.txt

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I8b05b6eab9b9d18b29c2199bbea239e9add1e690
Reviewed-on: https://webrtc-review.googlesource.com/c/115481
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26225}
2019-01-11 17:05:20 +00:00
Niels Möller
4687915495 Enable use of MediaTransportInterface for video streams.
Bug: webrtc:9719
Change-Id: I8c6027b4b15ed641e42fd210b3ea87d121508a69
Reviewed-on: https://webrtc-review.googlesource.com/c/111751
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26219}
2019-01-11 14:06:15 +00:00
Sam Zackrisson
ba50223363 Make voiceengine/audio transport stop using voice_detection() interface
Configuration for AudioProcessing::voice_detection() is moving into
AudioProcessing::Config, to get rid of the pointer-to-submodule
interfaces (such as voice_detection()).

Bug: webrtc:9947
Change-Id: Ia64ae996a43d44423aa0d612a3f1185b52a3e534
Reviewed-on: https://webrtc-review.googlesource.com/c/116067
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26216}
2019-01-11 12:31:29 +00:00
Jakob Ivarsson
53eae87bf8 Add PeerConnection option to enable RTX handling in the audio jitter buffer.
Bug: webrtc:10178
Change-Id: I70abce0c7b74124d2b1978d9a5eb8216b6233d1a
Reviewed-on: https://webrtc-review.googlesource.com/c/116784
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26203}
2019-01-10 16:28:43 +00:00
Oskar Sundbom
8984cd61ca Revert "Add a high bitrate full stack test with fake codec"
This reverts commit 15df2774f4e85cf8900768c1793edcf17d651dcd.

Reason for revert: It's causing the Android perf bots to fail. E.g.: https://ci.chromium.org/buildbot/client.webrtc.perf/Android32%20Tests%20%28L%20Nexus4%29/6666

Original change's description:
> Add a high bitrate full stack test with fake codec
> 
> This CL adds a fake codec factory  in WebRTC that can be used in tests to
> produce target bitrate output.
> 
> We also add a high bitrate test that makes use of fake codec. This test assumes
> ideal network conditions with target bandwidth being available and exercises
> WebRTC calls with a high target bitrate(100 Mbps) end-to-end.
> 
> Bug: chromium:879723
> Change-Id: I981124e2087054ed72c5447e239f28aae0878e29
> Reviewed-on: https://webrtc-review.googlesource.com/c/97185
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26182}

TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,emircan@webrtc.org,kron@webrtc.org

Change-Id: I33cd01ce345d81d66543f9be99750fa100760b5c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:879723
Reviewed-on: https://webrtc-review.googlesource.com/c/116785
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26192}
2019-01-10 11:49:05 +00:00
Emircan Uysaler
15df2774f4 Add a high bitrate full stack test with fake codec
This CL adds a fake codec factory  in WebRTC that can be used in tests to
produce target bitrate output.

We also add a high bitrate test that makes use of fake codec. This test assumes
ideal network conditions with target bandwidth being available and exercises
WebRTC calls with a high target bitrate(100 Mbps) end-to-end.

Bug: chromium:879723
Change-Id: I981124e2087054ed72c5447e239f28aae0878e29
Reviewed-on: https://webrtc-review.googlesource.com/c/97185
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26182}
2019-01-09 23:49:03 +00:00
Artem Titov
1ebfb6aac7 Introduce VideoFrame::id to keep track of frames inside application.
Also switch webrtc code from deprecated constructors to the builder API.

Change-Id: Ie325bf1e9b4ff1e413fef3431ced8ed9ff725107
Bug: webrtc:10138
Reviewed-on: https://webrtc-review.googlesource.com/c/114422
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26132}
2019-01-04 08:59:26 +00:00
Niels Möller
fd87da7a4b Delete WebRtcVideoCapturer and related classes.
Bug: webrtc:6353
Change-Id: Iee2d0d10bcfcb85aec81b426167d9e6a9485187f
Reviewed-on: https://webrtc-review.googlesource.com/c/115341
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26125}
2019-01-03 15:15:48 +00:00
Niels Möller
29e13fd2ca Delete rtc::PacketTime (was an alias for int64_t)
Followup to https://webrtc-review.googlesource.com/c/91860.

Bug: webrtc:9584
Change-Id: Icadf73d6c275ef32167357fc33b3c08158fa096f
Reviewed-on: https://webrtc-review.googlesource.com/c/114545
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26109}
2019-01-02 09:07:51 +00:00
Amit Hilbuch
77938e6409 Simulcast work to enable RID mux.
Rids can now be sent using rtp_sender.
Hooking up the rid values in the voice and video engine is still WIP.

Bug: webrtc:10074
Change-Id: I245c7ecb23b67fc0ba65caaa5dbb4fcfd60c81bb
Reviewed-on: https://webrtc-review.googlesource.com/c/114505
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26092}
2018-12-21 20:59:23 +00:00
Erik Språng
f5fc537c4b Allow H264 simulcast be default
Bug: chromium:908099
Change-Id: If50d59e01ea008ce29c8225c24f5189d34817506
Reviewed-on: https://webrtc-review.googlesource.com/c/115042
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26059}
2018-12-19 17:48:26 +00:00
Niels Möller
31d8b52075 Delete unneeded includes of rtc_base/stringutils.h.
Also delete corresponding dependencies on rtc_base:stringutils.

Bug: webrtc:6424
Change-Id: I2be5e021292eea2d788c76a63cc0e4f7cefd927d
Reviewed-on: https://webrtc-review.googlesource.com/c/114544
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26057}
2018-12-19 11:04:27 +00:00
Florent Castelli
73f2da9fa7 Remove VP8EncoderSimulcastProxy
The class has been renamed to EncoderSimulcastProxy.

Bug: webrtc:10069
Change-Id: Ief03cfb27145798ac46692d9e51371d2e119eeb0
Reviewed-on: https://webrtc-review.googlesource.com/c/114551
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26031}
2018-12-17 15:29:20 +00:00
Henrik Grunell
e1301a8b3a Revert "Implement read-only codecPayloadType in RtpParameters"
This reverts commit 806e06d1366b58878ced05cdd8d1d56394982fe6.

Reason for revert: Breaks WebRTC roll to Chromium. https://chromium-review.googlesource.com/c/chromium/src/+/1375538

02:52:35.346 7748   [6936:11248:1213/025234.206:ERROR:mediaengine.cc(80)] Attempted to set RtpParameters with modified codecPayloadType (INVALID_MODIFICATION)

Original change's description:
> Implement read-only codecPayloadType in RtpParameters
> 
> Bug: webrtc:7580
> Change-Id: I6d901afa97262b6c6d9fe6c7366df465ec77bfb3
> Reviewed-on: https://webrtc-review.googlesource.com/c/113944
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25993}

TBR=steveanton@webrtc.org,sakal@webrtc.org,andersc@webrtc.org,shampson@webrtc.org,orphis@webrtc.org

Change-Id: I157f9a79ae7133395431891e15e2c053559d359b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7580
Reviewed-on: https://webrtc-review.googlesource.com/c/114300
Reviewed-by: Henrik Grunell <henrikg@webrtc.org>
Commit-Queue: Henrik Grunell <henrikg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26000}
2018-12-13 12:13:30 +00:00
Florent Castelli
806e06d136 Implement read-only codecPayloadType in RtpParameters
Bug: webrtc:7580
Change-Id: I6d901afa97262b6c6d9fe6c7366df465ec77bfb3
Reviewed-on: https://webrtc-review.googlesource.com/c/113944
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25993}
2018-12-12 16:24:29 +00:00
Niels Möller
1d8307d706 Delete VideoCodec::targetBitrate
This member is unused by encoders.

Bug: None
Change-Id: I867013bfdb89f48782e84842de05bb57648e0b64
Reviewed-on: https://webrtc-review.googlesource.com/c/113882
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25988}
2018-12-12 12:48:15 +00:00
Anders Carlsson
01092957f5 Mark functions using old factory classes as deprecated.
The flag rtc_use_builtin_sw_codecs will be removed in a later CL and
this marks usage of the various entry points using the old video factory
API as deprecated.

Bug: webrtc:7925, webrtc:10044
Change-Id: I5c75516a41b0666e77539c028808cc2b173ed4bd
Reviewed-on: https://webrtc-review.googlesource.com/c/113061
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25970}
2018-12-11 16:17:33 +00:00
Benjamin Wright
514f084c26 New statistic added to VideoReceiveStream to determine latency to first decode.
This change introduces a new measurement into the VideoReceiveStream::Stats
structure to measure the latency between the first frame being received and
the first frame being decoded in WebRTC. The goal here is to measure the latency
difference when a FrameEncryptor is attached and not attached.

Change-Id: I0f0178aff73b66f25dbc6617098033e226da2958
Bug: webrtc:10105
Reviewed-on: https://webrtc-review.googlesource.com/c/113328
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25956}
2018-12-10 18:49:34 +00:00
Sergey Silkin
8b9b5f98db Activate/deactivate VP9 spatial layers.
* Stop encoding spatial layers S(n >= N) if application deactivates
spatial layer N by setting RTCRtpEncodingParameters.active = false.

* Move calculation of padding bitrate to SvcRateAllocator class.

* Pad up to minimum required bitrate of base layer if ALR probing is
enabled.

Bug: webrtc:9350
Change-Id: I398284c943d43348def535c83263fc234c9767fa
Reviewed-on: https://webrtc-review.googlesource.com/c/113240
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25945}
2018-12-10 12:55:51 +00:00
Mirta Dvornicic
1ec2a16121 Revert "Replace VideoEncoderFactory::QueryVideoEncoder with VideoEncoder::GetEncoderInfo"
This reverts commit cdc5eb0de179dcc866ef770ea303879c64466879.

Reason for revert: Causes wrong CPU adaptation to be used for some HW codecs since GetEncoderInfo() is polled before InitEncode().

Original change's description:
> Replace VideoEncoderFactory::QueryVideoEncoder with VideoEncoder::GetEncoderInfo
> 
> Make implementation of VideoEncoderFactory::QueryVideoEncoder optional
> until it is removed downstream and remove all implementations of it.
> 
> Bug: webrtc:10065
> Change-Id: Ibb1f9612234e536651ce53f05ee048a5d172a41f
> Reviewed-on: https://webrtc-review.googlesource.com/c/113065
> Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25924}

TBR=brandtr@webrtc.org,sakal@webrtc.org,kthelgason@webrtc.org,sprang@webrtc.org,srte@webrtc.org,perkj@webrtc.org,mirtad@webrtc.org

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

Bug: webrtc:10065
Change-Id: Idaa452e1d8c1c58cdb4ec69b88fce9042589cc3c
Reviewed-on: https://webrtc-review.googlesource.com/c/113800
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25943}
2018-12-10 10:36:00 +00:00
Mirta Dvornicic
cdc5eb0de1 Replace VideoEncoderFactory::QueryVideoEncoder with VideoEncoder::GetEncoderInfo
Make implementation of VideoEncoderFactory::QueryVideoEncoder optional
until it is removed downstream and remove all implementations of it.

Bug: webrtc:10065
Change-Id: Ibb1f9612234e536651ce53f05ee048a5d172a41f
Reviewed-on: https://webrtc-review.googlesource.com/c/113065
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25924}
2018-12-06 15:24:45 +00:00
Florent Castelli
e7862cc6b5 Copy VP8EncoderSimulcastProxy to EncoderSimulcastProxy
Use the new class internally where appropriate too.

The objective is to rename it, but due to some external dependency,
it is better to copy, update dependencies and remove.

Bug: webrtc:10069
Change-Id: I8477ce5a2982933db27513cc9509f51558dafaf3
Reviewed-on: https://webrtc-review.googlesource.com/c/113265
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25920}
2018-12-06 13:24:07 +00:00
Ilya Nikolaevskiy
5546aef682 Vp9 flexible mode fixes
- Enable vp9 flexible mode in VideoEngine if 3 spatial layers are set.
- Enable flexible mode in loopback tools and quality tests.
- Reset first active spatial layer on keyframe in encoder.
- Ensure duplicate references are not set by the sender in video header.
- Set references manually for flexible mode in vp9 encoder.
- Delay new activated layers until next base layer frame.
- On receive side put each spatial layer as a separate frame to FrameBuffer
  and return several frames combined from FrameBuffer.

Bug: webrtc:10049,webrtc:9794,webrtc:9784
Change-Id: I01e69f134cc145deba666ccc92deb1d37a324ede
Reviewed-on: https://webrtc-review.googlesource.com/c/112289
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25895}
2018-12-04 15:36:28 +00:00
Johannes Kron
d0b69a8c50 Send and receive color space information if available
Bug: webrtc:8651
Change-Id: I244647cb1ccbda66fce83ae925cf4273c5a6568b
Reviewed-on: https://webrtc-review.googlesource.com/c/112383
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25884}
2018-12-03 21:07:45 +00:00
Mirta Dvornicic
897a991618 Add metadata from VideoEncoderFactory::CodecInfo to VideoEncoder::EncoderInfo
This is the first step in moving the metadata and eventually replacing
VideoEncoderFactory::QueryVideoEncoder with VideoEncoder::GetEncoderInfo.

Bug: webrtc:10065
Change-Id: If925b895718e1b1225d2cf49bede1adb3ff281b8
Reviewed-on: https://webrtc-review.googlesource.com/c/112285
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25856}
2018-11-30 12:58:53 +00:00
Erik Språng
72e52ee619 Make simulcast screenshare default-on
Bug: chromium:690537
Change-Id: I7380f5e7b3faa20ba60bebee8b8b4d74db885faf
Reviewed-on: https://webrtc-review.googlesource.com/c/112381
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25840}
2018-11-29 14:17:41 +00:00
Yves Gerey
3e70781361 [Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
This is a follow-up to
https://webrtc-review.googlesource.com/c/src/+/106280.
This time the whole code base is covered.
Some files may have not been fixed though, whenever the IWYU tool
was breaking the build.

Bug: webrtc:8311
Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef
Reviewed-on: https://webrtc-review.googlesource.com/c/111965
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 18:25:07 +00:00
Jakob Ivarsson
10403ae87c Add PeerConnection option to configure minimum audio jitter buffer delay.
Note that this value will override the minimum delay that is used for audio/video sync.

Bug: webrtc:10053
Change-Id: Ia129f6c9ee9da5d00a3d955afaaa6e8f0c2bee33
Reviewed-on: https://webrtc-review.googlesource.com/c/112121
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25805}
2018-11-27 19:49:48 +00:00
Jakob Ivarsson
352ce5c419 Expose delayed packet outage as a cumulative metric of samples in the new getStats API.
The stat will be exposed through origin trial described in:
https://docs.google.com/document/d/1stYIZhEmDZ7NJF9gjjsM66eLFJUdc-14a3QutrFbIwI

Change-Id: Ib191a11c6bd9e617abbe9dd82239b0c5b4e6b4e0
Bug: webrtc:10043
Reviewed-on: https://webrtc-review.googlesource.com/c/111922
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25802}
2018-11-27 15:10:09 +00:00
Ruslan Burakov
8af8896596 Expose jitter buffer flushes metric in new getStats api.
Origin trial experiment proposal (new statistic part):
https://docs.google.com/document/d/1stYIZhEmDZ7NJF9gjjsM66eLFJUdc-14a3QutrFbIwI/edit?ts=5bf5535c#

Bug: chromium:907113
Change-Id: I1d005291f9b47665f70c26148dbdcbb55564bef8
Reviewed-on: https://webrtc-review.googlesource.com/c/111505
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Cr-Commit-Position: refs/heads/master@{#25768}
2018-11-23 11:41:43 +00:00
Fredrik Solenberg
c69a56ef04 Remove more unneeded things from ChannelSend
- SetNACKStatus() - only affects NetEq and RTP receiver
- GetRtpTimestampRateHz() - never used.
- ResendPackets() - never used.

Bug: webrtc:9801
Change-Id: I280b620723eb6917624f30f503eb8b8c88144e6d
Reviewed-on: https://webrtc-review.googlesource.com/c/111460
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25721}
2018-11-21 09:04:07 +00:00
Niels Möller
2222a80e79 Delete unneeded includes of common_types.h and gn deps on webrtc_common.
Bug: webrtc:5876
Change-Id: Iae14e5f1679067a5a5e0584ca830aee0870c8807
Reviewed-on: https://webrtc-review.googlesource.com/c/111463
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25715}
2018-11-20 16:28:39 +00:00
Florent Castelli
38332cdcb1 Add RTCP and simulcast support for RTCRtpReceiver::getParameters()
Bug: webrtc:9989
Change-Id: I1235789cd485750937a427199f9d32ed6180145e
Reviewed-on: https://webrtc-review.googlesource.com/c/110616
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25714}
2018-11-20 14:03:18 +00:00
Niels Möller
22b70ff1d4 Move VideoCodecType from common_types.h to api/video/video_codec_type.h
Bug: webrtc:7660
Change-Id: I9381364a64113dbb622b26acbf2b71228c3c4b96
Reviewed-on: https://webrtc-review.googlesource.com/c/111480
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25710}
2018-11-20 13:12:57 +00:00
Mirko Bonadei
8ef57932b1 Switch from RTC_DISABLE_VP9 to RTC_ENABLE_VP9.
RTC_ENABLE_VP9 is more natural to deal with then RTC_DISABLE_VP9.
In all the places this macro is used, WebRTC needs to do more things
so it is easier to "do more if RTC_ENABLE_VP9 is defined" than
"do more if RTC_DISABLE_VP9 is not defined".

Bug: None
Change-Id: If992e5c554173e6af3f030f6e0fd21bd82acf9eb
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/c/111242
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25679}
2018-11-19 08:30:55 +00:00
Sebastian Jansson
fa0aa39fba Removes templating from CompositeMediaEngine.
Usage of templates makes it harder for tooling to help the user. This
can be experienced when trying to investigate compile failures and using
editor tools to browse the code.

This CL replaces usage of templates with injection of unique pointers to
interfaces that implements the behavior that previously was assumed by
the templated implementation.

Bug: webrtc:9883
Change-Id: Ica17af9646f68a9b063988f9e85d6acc8ca37c10
Reviewed-on: https://webrtc-review.googlesource.com/c/106703
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25668}
2018-11-16 10:31:57 +00:00
Sebastian Jansson
84848f26b5 Adds interfaces for audio and video engines.
This makes the currently implicit interfaces explicit and
prepares for making CompositeMediaEngine non-templated.

Bug: webrtc:9883
Change-Id: I57452acc9ada60a801f6d624894440a942c12ded
Reviewed-on: https://webrtc-review.googlesource.com/c/106940
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25667}
2018-11-16 10:10:36 +00:00
Piotr (Peter) Slatala
cc8e8bb73f Pass the media transport from JsepTransportController to Call.
Add TargetRateObservers for media transport in the call object.



Bug: webrtc:9719
Change-Id: I5448d05359cf09b8cd2a678b2ac876aa8f8970e7
Reviewed-on: https://webrtc-review.googlesource.com/c/110622
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25662}
2018-11-15 17:36:48 +00:00