Commit Graph

1987 Commits

Author SHA1 Message Date
82ed2e852f Cleanup: Propagating BitrateAllocationUpdate to RtpVideoSender
Bug: webrtc:9883
Change-Id: I12d342ecd5eb0cc859123fe31fc759f6f60f7c8b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156940
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29492}
2019-10-15 14:40:48 +00:00
6841d25d45 Reland "RtpRtcp modules and below: Make media, RTX and FEC SSRCs const"
This is a reland of 17608dc4592fe25c1effdd75bf856f4af251942e

Downstream test now fixed.
As a precaution, also avoid DCHECKS for non-zero SSRC.
First patch set is reland, second makes checks more lenient.

Original change's description:
> RtpRtcp modules and below: Make media, RTX and FEC SSRCs const
>
> Downstream usage of SetSsrc() / SetRtxSsrc() should now be gone. Let's
> remove them, make the members const, and remove now unnecessary locking.
>
> Bug: webrtc:10774
> Change-Id: Ie4c1b3935508cf329c5553030f740c565d32e04b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155660
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29475}

Bug: webrtc:10774
Change-Id: I540b49a31a31e98d87f02ae04083d5206e71c1b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157100
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29491}
2019-10-15 14:03:19 +00:00
e8a6bc3f25 Revert "Reland "RtpRtcp modules and below: Make media, RTX and FEC SSRCs const""
This reverts commit c9348218cfe0cff6d0d3a383f7d1d6cfce4b1262.

Reason for revert: Downstream tests are relying on incorrect behavior which this CL explicitly checks...

Original change's description:
> Reland "RtpRtcp modules and below: Make media, RTX and FEC SSRCs const"
> 
> This is a reland of 17608dc4592fe25c1effdd75bf856f4af251942e
> 
> Downstream fixed, relanding.
> 
> Original change's description:
> > RtpRtcp modules and below: Make media, RTX and FEC SSRCs const
> >
> > Downstream usage of SetSsrc() / SetRtxSsrc() should now be gone. Let's
> > remove them, make the members const, and remove now unnecessary locking.
> >
> > Bug: webrtc:10774
> > Change-Id: Ie4c1b3935508cf329c5553030f740c565d32e04b
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155660
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#29475}
> 
> TBR=nisse@webrtc.org
> 
> Bug: webrtc:10774
> Change-Id: I759bed3ff1909857696c6d1b13df595a5e552f03
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157049
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29486}

TBR=nisse@webrtc.org,sprang@webrtc.org

Change-Id: I168fb3738a04dfdbd1581ddd8c3276ede9f72322
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10774
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157080
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29488}
2019-10-15 11:54:33 +00:00
c9348218cf Reland "RtpRtcp modules and below: Make media, RTX and FEC SSRCs const"
This is a reland of 17608dc4592fe25c1effdd75bf856f4af251942e

Downstream fixed, relanding.

Original change's description:
> RtpRtcp modules and below: Make media, RTX and FEC SSRCs const
>
> Downstream usage of SetSsrc() / SetRtxSsrc() should now be gone. Let's
> remove them, make the members const, and remove now unnecessary locking.
>
> Bug: webrtc:10774
> Change-Id: Ie4c1b3935508cf329c5553030f740c565d32e04b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155660
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29475}

TBR=nisse@webrtc.org

Bug: webrtc:10774
Change-Id: I759bed3ff1909857696c6d1b13df595a5e552f03
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157049
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29486}
2019-10-15 11:42:05 +00:00
e5d0fe0dff Updated VideoStreamEncoder to destroy encoder_queue_ before encoder_switch_experiment_.
Bug: none
Change-Id: I0d72fd0b851bd3f9b5021bc9b51af5da882483dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157044
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29484}
2019-10-15 10:04:38 +00:00
4ed0b52c12 Revert "RtpRtcp modules and below: Make media, RTX and FEC SSRCs const"
This reverts commit 17608dc4592fe25c1effdd75bf856f4af251942e.

Reason for revert: Breaks downstream build

Original change's description:
> RtpRtcp modules and below: Make media, RTX and FEC SSRCs const
> 
> Downstream usage of SetSsrc() / SetRtxSsrc() should now be gone. Let's
> remove them, make the members const, and remove now unnecessary locking.
> 
> Bug: webrtc:10774
> Change-Id: Ie4c1b3935508cf329c5553030f740c565d32e04b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155660
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29475}

TBR=nisse@webrtc.org,sprang@webrtc.org

Change-Id: Idc60f26f34dd0456a40c72375ae829e25b28621f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10774
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157046
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29483}
2019-10-15 09:43:21 +00:00
eb90e6ffe3 Merge SendTask implementation for SingleThreadedTaskQueueForTesting and TaskQueueForTest
That allows to use SingleThreadedTaskQueueForTesting via TaskQueueBase interface
but still have access to test-only SendTask function.

Bug: webrtc:10933
Change-Id: I3cc397e55ea2f1ed9e5d885d6a2ccda412beb826
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156002
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29480}
2019-10-15 09:17:36 +00:00
17608dc459 RtpRtcp modules and below: Make media, RTX and FEC SSRCs const
Downstream usage of SetSsrc() / SetRtxSsrc() should now be gone. Let's
remove them, make the members const, and remove now unnecessary locking.

Bug: webrtc:10774
Change-Id: Ie4c1b3935508cf329c5553030f740c565d32e04b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155660
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29475}
2019-10-15 07:50:59 +00:00
3f0d8e46a8 Revert "Delete methods EncodedImage::Allocate and EncodedImageBufferInterface::Realloc"
This reverts commit d6bb18479f4d9e258ae3e05427c101fb9e635373.

Reason for revert: Breaks Chromium Roll.
Example: https://ci.chromium.org/p/chromium/builders/try/android-kitkat-arm-rel/382446
Roll: https://chromium-review.googlesource.com/c/chromium/src/+/1859941

Original change's description:
> Delete methods EncodedImage::Allocate and EncodedImageBufferInterface::Realloc
> 
> Bug: webrtc:9378
> Change-Id: I3b03656769623647fcbb4f9125a3e920b7650fe9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155961
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29458}

TBR=ilnik@webrtc.org,nisse@webrtc.org,philipel@webrtc.org

Change-Id: I7209c5ae2be2d512572210cf08a4751ee2ee5bc4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9378
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157040
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29473}
2019-10-15 07:04:09 +00:00
41c650bea2 Use bitrate limits provided by encoder.
- Use minimum start bitrate to drop frame and adapt resolution in the
beginning of call.

- Use minimum bitrate to decide whether or not resolution should be
increased based on quality in MAINTAIN_FRAMERATE and BALANCED modes.
In BALANCED mode bitrate limits provided by the corresponding field
trial are prioritized over the limits provided by encoder.

Bug: webrtc:10853
Change-Id: I8257eb64565bcafa6ae9887a1af18e90f8400cac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156302
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29461}
2019-10-14 12:57:24 +00:00
d6bb18479f Delete methods EncodedImage::Allocate and EncodedImageBufferInterface::Realloc
Bug: webrtc:9378
Change-Id: I3b03656769623647fcbb4f9125a3e920b7650fe9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155961
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29458}
2019-10-14 12:13:31 +00:00
c67a4d63dd Fix WebRTC-Video-MinVideoBitrate for VP9
Make sure the experiment-derived value is used for VP9.

Bug: webrtc:11024
Change-Id: I80b6d388486f2dec793bc8ca872babe6165dcfb2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156562
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29453}
2019-10-11 17:56:51 +00:00
80f53b785b Extend WebRTC-Video-MinVideoBitrate to experiment per-codec
The experiment was extended to support per-codec minimum bitrates
for the following codecs:
 * VP8
 * VP9
 * H.264

The old semantic meaning for the field trial is retained, in that
specifying "br:" applies a minimum bitrate to all codecs. If "br:"
is not specified, the per-codec minimum config is consulted.

Bug: webrtc:11024
Change-Id: I89630262c7710771d5e25d039fe35f0bd217b58a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156171
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29450}
2019-10-11 15:34:33 +00:00
51bf200294 Reduce number of RTPVideoSender::SendVideo parameters
use frame_type from the RTPVideoHeader instead of as an extra parameter
merge payload data and payload size into single argument
pass RTPVideoHeader by value (relying on copy elision)

Bug: None
Change-Id: Ie7970af3b198b83b723d84c7a8b047219c4b38c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156400
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29445}
2019-10-11 10:59:21 +00:00
3b819f3d8b Move video_sources_.clear() call to CallTest::DestroyStreams
When one of the sources is a FrameGeneratorCapturer, this implies that
its TaskQueue is stopped. Before this change, the FrameGeneratorCapturer
was destroyed later, by the CallTest destructor, which led to a
use-after-free race on the Clock object passed to the capturer.

Bug: webrtc:11018
Change-Id: I3e53f95a725b6fb53b13e182ecd2caf03ea15bc4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156170
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29443}
2019-10-11 07:56:52 +00:00
5963c7cf0a Count disabled due to low bw streams or layers as bw limited quality in GetStats
Bug: webrtc:11015
Change-Id: I65cd890706f765366d89ded8c21fa7507797fc23
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155964
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29421}
2019-10-09 16:58:34 +00:00
7c06777ab0 Cleanup includes in modules/include/module_common_types.h
Add missing includes to files that were transactivly depending on removed includes.

Bug: None
Change-Id: Id5923bb8dc3e1d8fbb664e460278ad3e5993be7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155963
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29396}
2019-10-07 16:06:26 +00:00
24d251f796 Add 100 ms network delay to the SupportsFlexFEC* tests.
Some of the tests are currently flaky because FEC is disabled if the
RTT is <200 ms, and the simulated network is configured to use 100 ms
for the send transport, but nothing is configured for the receive
transport. This CL configures the receive transport to 100 ms delay.

Bug: webrtc:10920
Change-Id: I79995693ba73683406fa9ced92a7918e6c05473f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154571
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29394}
2019-10-07 13:01:05 +00:00
608083b66e Reset QP sum when QP is not reported on decoded frame.
To avoid incorrect QP sum in the reported stats and to avoid log spam
when switching from a decoder that reports QP to a decoder that does
not report QP.

Bug: None
Change-Id: Ib5ef4d6227344b0d03c3d75596b4a07ef427ae1c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155444
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29373}
2019-10-03 16:17:00 +00:00
7c079f650d Reland "Fix minor regression caused by a8336d3"
This is a reland of 809198edfff416fce8d75b574a43afab5e67b1cd

A fix was made in https://webrtc-review.googlesource.com/c/src/+/154343
which fixed the regression issues caused by the original patch.

Original change's description:
> Fix minor regression caused by a8336d3
>
> VideoEncoder::SetRates was being called unnessesarily when the fields
> appended to RateControlParameters were changed. Since SetRates only
> cares about RateControlParameters, it should have only been called if
> the RateControlParameters themselves were actually changed.
>
> Bug: webrtc:10126
> Change-Id: Ic47d67e642a3043307fec950e5fba970d9f95167
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152829
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Evan Shrubsole <eshr@google.com>
> Cr-Commit-Position: refs/heads/master@{#29208}

Bug: webrtc:10126
Change-Id: Iecc3ab6a5cd1193a1fa8e824dcf4f0b8165f9bf8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154359
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29356}
2019-10-01 11:49:38 +00:00
09f119598e Always pass arguments to INSTANTIATE_TEST_SUITE_P.
Passing an empty arg is working at the moment but it is not
guaranteed to continue to work in the future.

Bug: None
Change-Id: I975bc8779bac9700854de411301415338dcaf9f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154820
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29343}
2019-09-30 12:52:07 +00:00
45b176fc22 Downgrade fps in same step as resolution in initial drop due to size.
Bug: none
Change-Id: If0943ee291a029fa81035c72607873995ba8ab8b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154742
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29342}
2019-09-30 12:28:26 +00:00
fbf75a7891 Video: Log scalability configuration on encoder reconfigure
Lately there were 2 separate bugs, where seeing this information in the
log could help immediately figuring out the problem.

Bug: none
Change-Id: I3f2b2d5864106cdb231715e1702edee3b9b05caa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154566
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29338}
2019-09-30 09:08:20 +00:00
fc3587418d Use new RtpFrameObject ctor for unittests.
Bug: webrtc:10979
Change-Id: I63f501b3a4538d65a73aae226f2006de191dbbec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154565
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29337}
2019-09-30 08:28:45 +00:00
44db436e87 Propagate task queue to create test::DirectTransport by TaskQueueBase interface
actual task queue implementation for these tests is intentionally unchanged for now.

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

Bug: webrtc:10933
Change-Id: I324597b503e647c471f43511340eb9c07ba03ee8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154743
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29335}
2019-09-30 03:23:07 +00:00
abd9e0f5c2 VideoStreamEncoder: Don't crop and scales frames for external encoders
Bug: none
Change-Id: I1edc0314450793c0ae3fb471aa8936a102773a96
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154424
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29319}
2019-09-26 11:49:09 +00:00
7acc4a4a3a Reset |reference_finder_| on codec switch.
In this CL:
 - Moved critical section out of RtpFrameReferenceFinder.
 - RtpFrameReferenceFinder can now assign picture ids with an offset.
 - RtpVideoStreamReceiver will now reset the |reference_finder_| in case
   of a codec switch.

Bug: webrtc:10795, webrtc:10828
Change-Id: I22631c121a465c434de24af5ce8be2a647fe3556
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154353
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29317}
2019-09-26 11:05:59 +00:00
ccfb3401ee Update balanced tests to use BalancedDegradationTest fixture.
Bug: none
Change-Id: If10f28ff1da40a379b4a5b26c7bdf638913b48c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153351
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29312}
2019-09-26 06:59:54 +00:00
e32ae4f8fb Invalidate encoder rates on VideoStreamEncoder::ReconfigureEncoder
Without invalidation the codec configuration may not get updated. This
is prone to happen when the bandwidth is below the min or above the max
in which case the last_encoder_rate_settings_ may have not changed.

This led to a regression in b6a45dd, where last_encoder_rate_settings_
would not change the allocation or frame rate, but the frame rate would
have been set by the video adapter. Thus the frame rates were set
incorrectly, leading to lower values in the regression tests.

I have re-run this scenario against some of the metric drops and the
regression appears to be fixed.


Bug: webrtc:10126
Change-Id: I0fa6c9b71e7aff5dd80e53119db109d97eed98b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154343
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29301}
2019-09-25 11:37:52 +00:00
01dd88505c Moves contents of bitrate_controller to goog_cc
This CL moves send_side_bandwidth_estimation.cc/h and
loss_based_bandwidth_estimation.cc/h from modules/bitrate_controller
to modules/congestion_controller/goog_cc.

Bug: webrtc:9883
Change-Id: Ibb2c2ba3762007e7e5114f39042ee96431b73776
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154346
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29297}
2019-09-25 08:43:24 +00:00
b67c44c3f5 Add unit tests for balanced degradation settings.
Bug: none
Change-Id: I159965b931f0ab734b84cb68d5bfb7b5bd9a81a4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153348
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29295}
2019-09-25 08:41:14 +00:00
70bc753cc6 Add comments to MultiCodecReceiveTest.
Follow up to https://webrtc-review.googlesource.com/c/src/+/153880

Bug: none
Change-Id: If52e2ba638cc463f55330d5d5db1e1e566231562
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154349
Reviewed-by: Yves Gerey <yvesg@google.com>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29294}
2019-09-25 08:15:20 +00:00
1b575417b3 Always pass arguments to INSTANTIATE_TEST_SUITE_P.
Passing an empty arg is working at the moment but it is not
guaranteed to continue to work in the future.

This CL has been generated with:
git grep -l "INSTANTIATE_TEST_SUITE_P(," | xargs sed -i \
    "s/INSTANTIATE_TEST_SUITE_P(,/INSTANTIATE_TEST_SUITE_P(All,/g"

Bug: None
Change-Id: Icd2fb9d9d29aed5d692a234124bd990d0f097db4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153890
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29282}
2019-09-24 08:56:24 +00:00
82ce384801 Add improvement directions to PC and Call framework metrics
Bug: webrtc:10138
Change-Id: Ib957950df6e7490a15da0345fcd73e037c1a5b19
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153892
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29278}
2019-09-24 08:25:44 +00:00
834a554962 Include module_common_types.h only where needed
Bug: None
Change-Id: I73d493f8f186b429c7be808f4dfac0398f150931
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153891
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29277}
2019-09-24 08:22:38 +00:00
bf5ee00f8d Disable prerender smoothing in MultiCodecReceiveTest.
Avoids frame dropping in render queue.


Bug: webrtc:10828
Change-Id: I9e09fc2faee4626c8d60c152840b4208dbb89dc4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153880
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29276}
2019-09-24 08:12:09 +00:00
f7457e55fe Store PacketBuffer by value instead of as reference counted object
Bug: None
Change-Id: I5a594972e8a8dad731c927a1a374301e549f5d71
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153887
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29267}
2019-09-23 13:28:09 +00:00
04fd21513b Cleanup passing rtp packet to ulpfec receiver.
Pass RtpPacket class of header and raw packet separately

Bug: None
Change-Id: Id6d107db0e3751ff3dec87321ce6f850da0ee33a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153740
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29254}
2019-09-20 11:09:11 +00:00
0cff4fce55 Removed unused frame_size param from RtpFrameObject ctor.
Bug: webrtc:10979
Change-Id: Idde493dc7f5165e3ca173d5a38861b444b5904a4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153668
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29253}
2019-09-20 10:56:01 +00:00
b5e4785464 RtpFrameObject now takes an EncodedImageBuffer in its ctor.
Bug: webrtc:10979
Change-Id: Ibc8b4a524ca95b5faa8850a41df8f2f0136a2969
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153666
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29251}
2019-09-20 10:15:01 +00:00
f0be5b5380 Make GetBitstream non-virtual since it is no longer needed for testing.
Bug: webrtc:10979
Change-Id: Id313c7fddbec40b9f19dae95f736379b872e3082
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153663
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29242}
2019-09-19 14:04:09 +00:00
b6a45dda4c Revert "Fix minor regression caused by a8336d3"
This reverts commit 809198edfff416fce8d75b574a43afab5e67b1cd.

Reason for revert: Performance regressions that need to be addressed.

Original change's description:
> Fix minor regression caused by a8336d3
> 
> VideoEncoder::SetRates was being called unnessesarily when the fields
> appended to RateControlParameters were changed. Since SetRates only
> cares about RateControlParameters, it should have only been called if
> the RateControlParameters themselves were actually changed.
> 
> Bug: webrtc:10126
> Change-Id: Ic47d67e642a3043307fec950e5fba970d9f95167
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152829
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Evan Shrubsole <eshr@google.com>
> Cr-Commit-Position: refs/heads/master@{#29208}

TBR=sprang@webrtc.org,eshr@google.com

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

Bug: webrtc:10126
Change-Id: I133cbe5d8cb894ed944ae8a2d0f63a78bbed72ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153484
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29221}
2019-09-18 14:38:15 +00:00
738bfa7bab Remove api/bitrate_constraints.h.
Bug: webrtc:8733
Change-Id: Iaeb26e07d399f25dc18b0c4af38ed400577a5d3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153220
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29217}
2019-09-18 06:37:58 +00:00
ee5ec9a93a Replacing local closure classes with C++14 moving capture lambdas.
Bug: webrtc:10945
Change-Id: I569b9495cae98f204065911e13c37c31f35da372
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153241
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29214}
2019-09-17 19:43:05 +00:00
86314cfb5d Cleaning up C++14 move into lambda TODOs.
Bug: webrtc:10945
Change-Id: I4d2f358b0e33b37e4b4f7bfcf3f6cd55e8d46bf9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153240
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29212}
2019-09-17 19:18:26 +00:00
317a1f09ed Use std::make_unique instead of absl::make_unique.
WebRTC is now using C++14 so there is no need to use the Abseil version
of std::make_unique.

This CL has been created with the following steps:

git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt
git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt
git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt

diff --new-line-format="" --unchanged-line-format="" \
  /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \
  uniq > /tmp/only_make_unique.txt
diff --new-line-format="" --unchanged-line-format="" \
  /tmp/only_make_unique.txt /tmp/memory.txt | \
  xargs grep -l "absl/memory" > /tmp/add-memory.txt

git grep -l "\babsl::make_unique\b" | \
  xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g"

git checkout PRESUBMIT.py abseil-in-webrtc.md

cat /tmp/add-memory.txt | \
  xargs sed -i \
  's/#include "absl\/memory\/memory.h"/#include <memory>/g'
git cl format
# Manual fix order of the new inserted #include <memory>

cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \
  xargs sed -i '/#include "absl\/memory\/memory.h"/d'

git ls-files | grep BUILD.gn | \
  xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d'

python tools_webrtc/gn_check_autofix.py \
  -m tryserver.webrtc -b linux_rel

# Repead the gn_check_autofix step for other platforms

git ls-files | grep BUILD.gn | \
  xargs sed -i 's/absl\/memory:memory/absl\/memory/g'
git cl format

Bug: webrtc:10945
Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 15:47:29 +00:00
809198edff Fix minor regression caused by a8336d3
VideoEncoder::SetRates was being called unnessesarily when the fields
appended to RateControlParameters were changed. Since SetRates only
cares about RateControlParameters, it should have only been called if
the RateControlParameters themselves were actually changed.

Bug: webrtc:10126
Change-Id: Ic47d67e642a3043307fec950e5fba970d9f95167
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152829
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#29208}
2019-09-17 13:34:18 +00:00
7262fc29a0 Refactor Rtp Receivers to accept SSRC 0.
Changes Rtp Receivers to use a null value of ssrc to mean a default
receive stream.

Bug: webrtc:8694
Change-Id: I835199345f7add993b9078c8b0e7988d5cdd6646
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152425
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Saurav Das <dinosaurav@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29201}
2019-09-16 21:29:58 +00:00
d9cc8c08dc Encoder switching based on network and/or resolution conditions.
In this CL:
 - Renamed EncoderFailureCallback to EncoderSwitchRequestCallback. An encoder
   switch request can now also be made with a configuration that specifies which
   codec/implementation to switch to.
 - Added "WebRTC-NetworkCondition-EncoderSwitch" field trial that specifies
   switching conditions and desired codec to switch to.
 - Added checks to trigger the switch based on these conditions.

Bug: webrtc:10795
Change-Id: I9d3a9a39a7c4827915a40bdceed10b581d70b90a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151900
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29196}
2019-09-16 13:43:29 +00:00
a740142398 Refactor LossNotificationController to not use VCMPacket
Bug: None
Change-Id: I15e1b3405c6538dd22aaeb125751c158c069478a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152384
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29193}
2019-09-16 11:25:45 +00:00