Commit Graph

28139 Commits

Author SHA1 Message Date
443b7ee635 Destroy existing encoder instance before creating a new one.
Before this change, an attempt to recreate video encoder would fail if
video encoder factory supports only single instance of an encoder.

Added tracking of max number of existed simultaneously encoder
instances to VideoEncoderProxyFactory.

Bug: webrtc:10776
Change-Id: I317cbdf1af94dfb4c72bf99c5cd4ce7b454188fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144044
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28457}
2019-07-03 10:15:57 +00:00
2c5af4f6dc Add * and / operator into SamplesStatsCounter.
Add * and / operator into SamplesStatsCounter to simplify convertion
between samples units.

Bug: webrtc:10138
Change-Id: I985b60bc63c071ceaf0fddcdbcd3e6476353c0fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142171
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28456}
2019-07-03 09:19:36 +00:00
1d46f9c599 Add RtpPacket::IsExtensionReserved().
This is a small utility method to check whether an extension has been
reserved, so that can be checked before attempting to set an extension
without the need to actually try setting it and potentially failing
with warning loggins as a result.

Bug: webrtc:10633
Change-Id: Ie6f2c4f3f5e94a30dbf60aec6290ebee72681d9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144461
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28455}
2019-07-03 08:13:41 +00:00
6038926379 Roll chromium_revision 35be8751a4..ba17fd6b36 (674036:674288)
Change log: 35be8751a4..ba17fd6b36
Full diff: 35be8751a4..ba17fd6b36

Changed dependencies
* src/base: 5eac8b6a4a..c97a005aa2
* src/build: c4c0a07646..6c0140b448
* src/ios: 5a37f1d06a..9c68e43388
* src/testing: cff8e64fbe..5f46e6546d
* src/third_party: ed7c465048..39b1633438
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/2c752374f1..63f7fcc05a
* src/third_party/libvpx/source/libvpx: 30e7f9d856..cd9f1763c8
* src/third_party/libyuv: b36c86fdfe..09cfb2bbd6
* src/tools: 9d7bd5d9cc..143f7f54fb
DEPS diff: 35be8751a4..ba17fd6b36/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,marpan@webrtc.org, jianj@chromium.org,
BUG=None

Change-Id: If1a4181b12d223ebcfd841b42febd463b6f0decc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144500
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#28454}
2019-07-03 01:01:28 +00:00
02d7d353a9 Revert "Add ability to set ssrcs of RtpSender at construction time"
This reverts commit e9d6e658c307fc0241d622756703d5c0d5388d80.

Reason for revert: breaks downstream project

Original change's description:
> Add ability to set ssrcs of RtpSender at construction time
> 
> Bug: webrtc:10774
> Change-Id: I7147a75ccbcd1093dcd2e08047da8900843fdd8d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144037
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28447}

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

Change-Id: I8b0cba0836e7d86ae1718055196c2c89860b97ff
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10774
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144368
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28453}
2019-07-02 21:05:07 +00:00
c4421979fe Check the rid direction matches the direction in simulcast description
Section 5.2 of draft-ietf-mmusic-sdp-simulcast-14:
The direction for an rid-id MUST be aligned with the direction
specified for the corresponding RTP stream identifier on
the "a=rid" line.

Bug: webrtc:10785
Change-Id: I1fc70706511ae17c821c5ec4d90a0b854171454f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144245
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28452}
2019-07-02 19:17:06 +00:00
238aab9948 Fix bug in use_datagram_transport configuration.
Currently, use_datagram_transport's non-default value is never used.
Instead of reading configuration.use_datagram_transport,
PeerConnection::Initialize reads the local configuration's
use_datagram_transport.  This hasn't been set yet, and so it always
falls back to the default value.

Bug: webrtc:9719
Change-Id: I028ed537c7d88ee3421b6bd92fc7d5e3c6970529
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144441
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28451}
2019-07-02 18:45:46 +00:00
b073f1c27f Only set the RtcEventLog for media transport when it's used for media.
Currently, GenerateOrGetLastMediaTransportOffer() creates a media
transport that has an RtcEventLog, regardless of whether the media
transport is used for media or data channels.  It should only set the
RtcEventLog when used for media.

Bug: webrtc:10789
Change-Id: Id91c16973deec89bbc8c6518c4c9f1039f1265fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144367
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28450}
2019-07-02 17:30:32 +00:00
73bfc0ea8d Roll chromium_revision 6f0434662d..35be8751a4 (673926:674036)
Change log: 6f0434662d..35be8751a4
Full diff: 6f0434662d..35be8751a4

Changed dependencies
* src/base: 3d13a7aa98..5eac8b6a4a
* src/build: 3fa1f27670..c4c0a07646
* src/ios: dfb6071255..5a37f1d06a
* src/testing: 45f0246c12..cff8e64fbe
* src/third_party: 13250e00d5..ed7c465048
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/15ad0b0b62..2c752374f1
* src/tools: a27a29d288..9d7bd5d9cc
DEPS diff: 6f0434662d..35be8751a4/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: Ie1fb5149871037abddb2f064fe537e751c958a34
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144365
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#28449}
2019-07-02 15:04:12 +00:00
ed56cf44b0 Remove deprecated version of Vp8FrameBufferControllerFactory::Create
Bug: webrtc:10769
Change-Id: I31de2d21d5325eb4b175815739c2477352842425
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144381
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28448}
2019-07-02 14:40:20 +00:00
e9d6e658c3 Add ability to set ssrcs of RtpSender at construction time
Bug: webrtc:10774
Change-Id: I7147a75ccbcd1093dcd2e08047da8900843fdd8d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144037
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28447}
2019-07-02 13:03:25 +00:00
5ee6967c4e Don't reset encoder on max/min bitrate change.
- Don't reset encoder if max/min bitrate changed.
- Removed min/max bitrate DCHECKs from encoder wrappers.
- Reset encoder if start_bitrate changed. Only do this if encoding
  has not yet started.
- Updated ReconfigureBitratesSetsEncoderBitratesCorrectly test.
- Removed EncoderSetupPropagatesCommonEncoderConfigValues test since it
was a subset of ReconfigureBitratesSetsEncoderBitratesCorrectly.

Bug: webrtc:10773
Change-Id: Id9cbb2ea229232fd95967819e2a937b26948de9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144028
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28446}
2019-07-02 12:52:55 +00:00
bc70b6164e Switch rnn_vad_tool to ABSL_FLAG.
Tested:
$ ./out/Debug/rnn_vad_tool --i ./data/voice_engine/audio_tiny8.wav \
  --o /tmp/o.prob --f /tmp/o.feat
(rnn_vad_tool.cc:47): Input sample rate: 8000
(rnn_vad_tool.cc:105): VAD probabilities written to /tmp/o.prob
(rnn_vad_tool.cc:108): features written to /tmp/o.feat

Bug: webrtc:10616
Change-Id: Ied33d9425bc1621d084bb04d9acf12ea9602a88b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144048
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28445}
2019-07-02 12:27:15 +00:00
f1a7bb1b09 Stop using unnecessary gclient vars
My understanding is that this purely cargo-culting. We never actually use the variables, nor do we need the "Three lines of non-changing comments", because we roll everything at once anyway.

I also remove the 'chromium_git' variable because the indirection is not helpful and makes the file harder to follow.

I went even further and concatenated all strings. Again, perhaps in the past this was used to more easily update the file with automated tools, but that is also not necessary now.

Verified:
* `gclient revinfo` output remains exactly the same.
* roll_deps.py keeps working.

Change-Id: Ib8fc33b0f11f53fab0fed9b9b578db964fb70bdc
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144400
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28444}
2019-07-02 11:16:45 +00:00
45befc5f1f Pass FecControllerOverride to Vp8FrameBufferControllerFactory::Create
Previously, FecControllerOverride was passed to
Vp8FrameBufferController::SetFecControllerOverride. Passing to
the factory is a more elegant way, since it's only used when
the controller is constructed.

TBR=kwiberg@webrtc.org

Bug: webrtc:10769
Change-Id: Iae599889e7ca9003e3200c2911239cbb763ee65a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144380
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28443}
2019-07-02 10:55:55 +00:00
14be7993c6 Switch neteq tools to ABSL_FLAG.
Bug: webrtc:10616
Change-Id: I2aa688f0976d5618347e402f25d8701b0cf5a360
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144027
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28442}
2019-07-02 10:54:06 +00:00
e731a2ed98 Remove check on supported profile in favor of expilict disabling
Bug: webrtc:10138
Change-Id: I7a258ac992f774e4453092175e51e0033b29d4e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144244
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28441}
2019-07-02 09:07:57 +00:00
bfd343b9be Add totalDecodeTime to RTCInboundRTPStreamStats
Pull request to WebRTC stats specification:
https://github.com/w3c/webrtc-stats/pull/450

Bug: webrtc:10775
Change-Id: Id032cb324724329fee284ebc84595b9c39208ab8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144035
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: 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@{#28440}
2019-07-02 08:28:06 +00:00
419aae2ed3 Remove android_tools deps
Bug: chromium:428426
Change-Id: If99814585df0ef2dd0f1433ca1e71184ecd94baf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144340
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28439}
2019-07-02 06:48:30 +00:00
ebdf9f8163 Roll chromium_revision 097ffaa18d..6f0434662d (673789:673926)
Change log: 097ffaa18d..6f0434662d
Full diff: 097ffaa18d..6f0434662d

Changed dependencies
* src/base: 5413464fb7..3d13a7aa98
* src/build: b1f7a47cfa..3fa1f27670
* src/ios: 07244c207c..dfb6071255
* src/testing: 971f346545..45f0246c12
* src/third_party: ad26420900..13250e00d5
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/0384e1a460..15ad0b0b62
* src/tools: 3e04aeaa49..a27a29d288
DEPS diff: 097ffaa18d..6f0434662d/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I05b6a5cae838906bd6e7fdcd6792cd77af96fc13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144265
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#28438}
2019-07-02 02:55:51 +00:00
6fdfec13b9 Add overload to CreateIceTransport that takes additional dependencies
Bug: chromium:927579
Change-Id: I99c77af69b529ebd38c9f5a83882dfe6a2a1ffe9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144320
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28437}
2019-07-01 21:37:54 +00:00
1cf9470d92 Roll chromium_revision 05067e74f0..097ffaa18d (673689:673789)
Change log: 05067e74f0..097ffaa18d
Full diff: 05067e74f0..097ffaa18d

Changed dependencies
* src/build: e41944eb14..b1f7a47cfa
* src/ios: 9b22391709..07244c207c
* src/testing: 7054f41840..971f346545
* src/third_party: bde0ca49d8..ad26420900
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/88a8d7ec3b..0384e1a460
* src/tools: 8ec21a77bc..3e04aeaa49
DEPS diff: 05067e74f0..097ffaa18d/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I3e77179b563ee642669a17059c233281a64f0c4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144300
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#28436}
2019-07-01 20:41:27 +00:00
5985a0481e Add a field trial to control datagram transport use.
First, the existing configuration parameter (use_datagram_transport) is
now optional.

The new field trial has two flag values:
 1. Whether to enable the datagram transport (enabled)
 2. Whether to use the datagram transport by default (default_value)

The first is a kill-switch.  It disables the datagram transport, even
for applications which inject a datagram transport factory and specify
use_datagram_transport = true.  This allows applications which hard-code
a datagram transport to switch it off via field trials.

This flag defaults to true, to avoid breaking downstream projects which
already inject and configure a datagram transport.  It may be changed to
false after updating downstream to set this field trial flag to true
when required.

The second provides a default value to be used in case the
aforementioned use_datagram_transport parameter is unset.  Applications
which explicitly set use_datagram_transport will use that value.
Applications which do not explicitly specify whether or not to use the
datagram transport will use it (or not) according to the default_value
flag.

One goal of this flag is to simplify rollout in applications which
already set field trials based on configuration, but require code
changes for new RTCConfiguration parameters.  A second goal is to
provide platforms with a knob to control whether datagram transport is
"opt-in" or "opt-out".

This flag defaults to false, to prevent downstream projects from
unintentionally enabling the datagram tranpsort.

Bug: webrtc:9719
Change-Id: I521a5fa61c992e76e5081118678a1812a261d672
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144184
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28435}
2019-07-01 20:03:05 +00:00
3e8ef940fe Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker.
This change adds the plumbing of RtpPacketInfo from ChannelReceive::OnRtpPacket() to ChannelReceive::GetAudioFrameWithInfo() for audio. It is a step towards replacing the non-spec compliant ContributingSources that updates itself at packet-receive time, with the spec-compliant SourceTracker that will update itself at frame-delivery-to-track time.

Bug: webrtc:10668
Change-Id: I03385d6865bbc7bfbef7634f88de820a934f787a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139890
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28434}
2019-07-01 15:56:40 +00:00
62eb89d221 Fixing possible overflow in NetEq buffle level filter.
Bug: chromium:979281
Change-Id: Ieb3a8f9dc03114b76b13d1f8c529e9f759804da9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144240
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28433}
2019-07-01 15:17:29 +00:00
5983087656 Forced vp8 sw encoder fallback: only use min bitrate config if codec type is vp8.
Bug: none
Change-Id: Ifd99d2628e7cb1f4b38f028d26c8698ed8ff0ea1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144055
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28432}
2019-07-01 13:32:35 +00:00
5b2ce12bc1 Roll chromium_revision ce6d12c81b..05067e74f0 (673457:673689)
Change log: ce6d12c81b..05067e74f0
Full diff: ce6d12c81b..05067e74f0

Changed dependencies
* src/base: b8d7dd92d6..5413464fb7
* src/build: fe39ce9fd1..e41944eb14
* src/ios: 8c6bafc3be..9b22391709
* src/testing: 5398243e97..7054f41840
* src/third_party: 68164bde6b..bde0ca49d8
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/85e0cba51e..88a8d7ec3b
* src/third_party/depot_tools: 788d9e0de8..dc37feb99d
* src/third_party/icu: 9f0f47b1e4..fd97d4326f
* src/tools: a596745760..8ec21a77bc
DEPS diff: ce6d12c81b..05067e74f0/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: Ia541aa44b8f6dffd4261e70bbfc21468a7359420
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144260
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#28431}
2019-07-01 12:58:40 +00:00
ea95c37b0c Report freeze_time_ms from PC test framework
Bug: webrtc:10138
Change-Id: If6bc8f4e4b03bc18b44c8625e241837a47e96ae2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144243
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28430}
2019-07-01 12:41:20 +00:00
754c952dfa Don't do ToI420() for each frame while checking is it dummy
Bug: webrtc:10138
Change-Id: I7dba58324afd65ce046a0a255277de7134ebd0b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144242
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28429}
2019-07-01 12:23:49 +00:00
6d9f0010f0 Fix regression in PC quality test.
After adding support of simulcast for Vp8 in PC test framework the bug
was intorduced: when ulp FEC is enabled by user, it actualy was disabled
because of typo in FilterVideoCodecCapabilities. This CL will restore
the right behavior.

Bug: webrtc:10138, chromium:976690
Change-Id: Ia977f6d903af5a6b0ed9d2c65b75973bd65f5000
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144241
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28428}
2019-07-01 12:18:16 +00:00
a6cb1507cc Use Default instead of GlobalTaskQueueFactory to create AudioDeviceBuffer for ios
Bug: webrtc:10284
Change-Id: Ibeaf3c79335abe9ac32522156b8e20a6e2266c49
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144034
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28427}
2019-07-01 11:20:27 +00:00
3e3a6e59dc Remove obsolete deps
BUG=None

Change-Id: I6627b5d50df44e60e23b6d25ff81f689b7efc38a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144220
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28426}
2019-07-01 10:29:36 +00:00
896f4b666c Use Default instead of GlobalTaskQueueFactory to create AudioDeviceBuffer for android
Bug: webrtc:10284
Change-Id: I979eab78e1841e2b6900d7729159ee69274af8e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144031
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28425}
2019-07-01 09:43:06 +00:00
a63aede22b Make VideoStreamEncoderInterface::SetFecControllerOverride pure virtual
Clean-up after CL #143962

Bug: webrtc:10769
Change-Id: Ibf574b7b012c40846a88aaf2f1e7a6e2526a25ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143963
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28424}
2019-07-01 09:03:31 +00:00
65764e4ed7 Add missing overrides in VideoEncoder proxies/adapters
Add:
1. OnPacketLossRateUpdate
2. OnRttUpdate
3. OnLossNotification

Add them to:
1. VideoEncoderSoftwareFallbackWrapper
2. SimulcastEncoderAdapter
3. MultiplexEncoderAdapter

Bug: None
Change-Id: I4b0799f7d8c19211741f48da87106daccd39af95
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144030
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28423}
2019-06-28 22:45:53 +00:00
2ce1da5328 Roll chromium_revision dfe5e91525..ce6d12c81b (673350:673457)
Change log: dfe5e91525..ce6d12c81b
Full diff: dfe5e91525..ce6d12c81b

Changed dependencies
* src/base: bba591b64b..b8d7dd92d6
* src/build: e16f9dfa9e..fe39ce9fd1
* src/ios: 766dd2581e..8c6bafc3be
* src/testing: 40bd8e04ee..5398243e97
* src/third_party: 6e20b1cb4e..68164bde6b
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/6d47875c4c..85e0cba51e
* src/third_party/libvpx/source/libvpx: 7d9288f5f8..30e7f9d856
* src/tools: e3e8d61384..a596745760
DEPS diff: dfe5e91525..ce6d12c81b/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,marpan@webrtc.org, jianj@chromium.org,
BUG=None

Change-Id: I3ca03d11fb4a5d32fbab62226386a5683ee19f2a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144183
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#28422}
2019-06-28 21:28:23 +00:00
1e00dbccc2 Stun server should return XOR-MAPPED-ADDRESS/MAPPED-ADDRESS correctly
* Return xor mapped address for RFC5389 compatible client
* fix a typo in function name
* update stunserver unitest case
* update author

Bug: webrtc:10764
Change-Id: I466799744a343508233c18b7c477d2212680392a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143841
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28421}
2019-06-28 19:12:14 +00:00
67daf71689 Implement RtpVideoSender::SetFecAllowed()
Bug: webrtc:10769
Change-Id: I7214b2eaad828c59fd9836e85a3ecd8e737fe5f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143966
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28420}
2019-06-28 18:24:16 +00:00
099b02a366 Get rid of deprecated version of NackSender::SendNack [2/2]
[1/2] - Make new version pure-virtual, and deprecated version non-pure.
        This will allow deleting the deprecated version from downstream
        projects.
[2/2] - Remove deprecated version.

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

Bug: webrtc:10336
Change-Id: I3904da12ec471980adfb22f2e61304d42de4ec66
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144043
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28419}
2019-06-28 17:58:38 +00:00
7e00c679a5 Pass FecControllerOverride to Vp8FrameBufferController
Bug: webrtc:10769
Change-Id: I06d875f5afdc7ebf290ad70934b6632e20ddf065
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143964
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28418}
2019-06-28 17:48:08 +00:00
22896d4e78 Roll chromium_revision 27c2f87cf5..dfe5e91525 (672459:673350)
Change log: 27c2f87cf5..dfe5e91525
Full diff: 27c2f87cf5..dfe5e91525

Changed dependencies
* src/base: cfd4d4295e..bba591b64b
* src/build: aa80214bc0..e16f9dfa9e
* src/ios: 7fa66b1103..766dd2581e
* src/testing: cabcd61b51..40bd8e04ee
* src/third_party: 7355a06201..6e20b1cb4e
* src/third_party/android_build_tools/aapt2: j6U3mv7-KG3PSDtVvTwycWzjwvFR1_sSdA540AYxpucC..version:3.6.0-alpha03-5516695-cr0
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/92b7c89e6e..cfcb0060e8
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/d556f63347..6d47875c4c
* src/third_party/depot_tools: d9ae33420a..788d9e0de8
* src/third_party/freetype/src: 81445c034a..7b1c7585d7
* src/third_party/harfbuzz-ng/src: 659eeddb2d..7185bd6ffb
* src/tools: 957d279bd5..e3e8d61384
DEPS diff: 27c2f87cf5..dfe5e91525/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: Ia68c6f9410de941298364d5ae093442b0ab96a75
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144181
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#28417}
2019-06-28 17:34:59 +00:00
8f01c4e1b6 Define FecControllerOverride and plumb it down to VideoEncoder
The purpose of this interface is to allow VideoEncoder to override
the bandwidth allocation set by FecController in RtpVideoSender.

This CL defines the interface and sends it down to VideoSender.
Two upcoming CLs will:
1. Make LibvpxVp8Encoder pass it on to the (injectable)
   FrameBufferController, where it might be put to good use.
2. Modify RtpVideoSender to respond to the message sent to it
   via this API.

TBR=kwiberg@webrtc.org

Bug: webrtc:10769
Change-Id: I2ef82f0ddcde7fd078e32d8aabf6efe43e0f7f8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143962
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28416}
2019-06-28 15:57:22 +00:00
52e5242593 Add trait to Build/Parse DependencyDescriptor rtp header extension
TBR=aleloi@webrtc.org

Bug: webrtc:10342
Change-Id: I9d321ec47ed748ccfac2be6793923c46d0a88d16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144032
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28415}
2019-06-28 14:21:21 +00:00
225842ced8 Initialize signal processing function pointers statically
The last run-time logic for selecting function pointers was removed in
May 2016, here: https://codereview.webrtc.org/1955413003

It would be even better if we could eliminate the function pointers
entirely and just have different implementations that we select at
compile time; I've left a TODO asking for this.

Bug: webrtc:9553
Change-Id: Ica71d71e19759da00967168f6479b7eb8b46c590
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144053
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28414}
2019-06-28 14:20:03 +00:00
a47ba4119f Get rid of deprecated version of NackSender::SendNack [1/2]
[1/2] - Make new version pure-virtual, and deprecated version non-pure.
        This will allow deleting the deprecated version from downstream
	projects.
[2/2] - Remove deprecated version.

TBR=stefan@webrtc.org

Bug: webrtc:10336
Change-Id: Ia132ef071b1f379fc74834178e75e981ca908125
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144042
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28413}
2019-06-28 13:04:48 +00:00
a09484940b RateControlSettings: add option to set min pixels per frame for libvpx vp8.
Bug: none
Change-Id: I09aa1bcea2f4a9cd65ffeef1df1d9656e4604def
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144029
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28412}
2019-06-28 10:54:23 +00:00
60bfb3d4e3 NetEQ: BackgroundNoise::Update returns true when the filter is updated
Bug: webrtc:10690
Change-Id: I17ff7dc1cffc8c46987d0a9ff8c6633ce9dcc8d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144040
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28411}
2019-06-28 10:11:33 +00:00
825aad13dc Delete almost all includes of platform_file.h
Only remaining user is WavReader. Demote its constructor
accepting a PlatformFile to private, to refactor implementation
in a later cl.

Bug: webrtc:6463
Change-Id: I7b950be6f02073cb135dd0fab1190b9dc0de1fba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144025
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28410}
2019-06-28 07:30:15 +00:00
767efaba10 Delete method ReadableWav::Eof, which was used incorrectly.
Bug: webrtc:6463
Change-Id: I160ff01c16045eba438f928b429eacdb5342e5a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144026
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28409}
2019-06-28 07:29:10 +00:00
71809c67ce WindowCapturerWin: properly check return value of GetClassName
In debug mode I hit the assert so this function can return 0. So
just skip the window in that case like for other desktop elements

Bug: None
Change-Id: I92abf2a1f450b677632f5eb4332ca218cfd850ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143860
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28408}
2019-06-27 21:11:41 +00:00