Commit Graph

1023 Commits

Author SHA1 Message Date
ade0dc9860 Make FrameBuffer be able to signal if it's trivially convertible to I420
Bug: chromium:930186,webrtc:10310
Change-Id: I7857c33d3616ac58738b22816f9c78fe9e6d1d3c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134206
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27798}
2019-04-29 09:55:47 +00:00
299c4e6846 Piping audio interruption metrics to API layer
Bug: webrtc:10549
Change-Id: Ie6abe5819c5b73dc5f5f89bdc375bad77f44ce97
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134303
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27788}
2019-04-26 13:32:34 +00:00
da5aa4ddf5 Use CodecBufferUsage to determine dependencies.
In this CL:
 - Assign frame IDs so that simulcast streams share one frame ID space.
 - Added a CodecBufferUsage class that represent how a particular buffer
   was used (updated, referenced or both).
 - Calculate frame dependencies based on the CodecBufferUsage information.

Bug: webrtc:10342
Change-Id: I4ed5ad703f9376a7d995c04bb757c7d214865ddb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131287
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27784}
2019-04-26 12:13:28 +00:00
544dece6c1 Allow Vp8FrameBufferController to override resilience mode
Bug: webrtc:10382
Change-Id: I626d616d7a1b50a696f5378345d026f6dce5b97f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134207
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27777}
2019-04-25 18:30:08 +00:00
44c21f48ee Encoder side of Multistream Opus.
Follows https://webrtc-review.googlesource.com/c/src/+/129768 closely.
Adds an ENCODER and sets it up to parse SDP config for multistream
opus.

E.g. this is the new SDP syntax for 6.1 surround sound:
"multiopus/48000/6 channel_mapping=0,4,1,2,3,5 num_streams=4 coupled_streams=2"


Bug: webrtc:8649
Change-Id: I3fc341e76f5c41dab0243cf65f6461e4c3d9d67d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132001
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27775}
2019-04-25 15:07:38 +00:00
7ccaf8969d Cleanup of network controller handling in Scenario tests.
Removing functionality to choose congestion controller implementation,
using injection instead. Also cleaning up some related functionality
that's no longer needed, such as the injection of event logs into the
factory.

Bug: webrtc:9883
Change-Id: Ia528005625430ae31a15bc88881e2d4ac6ad1d42
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133890
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27768}
2019-04-25 12:40:00 +00:00
1845922d5a Introduce QualityMetricsReporter and implement network stats gathering
QualityMetricsReporter helps to keep network emulation framework and
peer connection level test framework separated. Also it provides
ability to gather statistics from any component around with
correlation with call start and end.

Bug: webrtc:10138
Change-Id: Ib3330a8d35481fde77fcf77d2271d6cfcf188fec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132718
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27759}
2019-04-25 09:36:50 +00:00
43f7002aff Delete DecodedImageCallback::ReceivedDecodedFrame
This was a companion method to ReceivedDecodedReferenceFrame, deleted
in https://webrtc-review.googlesource.com/c/src/+/133348.

Tbr: kwiberg@webrtc.org # Mock class update
Bug: webrtc:7408
Change-Id: I429f5f5c18f14c27136e82860297107a82c81d13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133571
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27754}
2019-04-25 08:09:29 +00:00
5d34dcfe60 Reland "Delete deprecated variant of VideoDecoder::Decode"
This is a reland of 3a86d9520c9692abe4c6d2a73faf4fb7a0cb80b2

Original change's description:
> Delete deprecated variant of VideoDecoder::Decode
> 
> Bug: webrtc:10379
> Change-Id: I4dd8b503625a9ea2a71177165238e128ac3e49bb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132554
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27712}

Bug: webrtc:10379
Change-Id: I7206756eb5cdbeb320fae74f286a97852fa4368b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133889
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27750}
2019-04-25 07:07:28 +00:00
70a596386d AEC3: Stationary init: fixing a typo when reading the json configuration
Bug: webrtc:10554
Change-Id: I8c05b7c654117359ff294c7ae4604d7913505650
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133181
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27726}
2019-04-24 07:42:37 +00:00
162a03979b Followup to Rename "UpdateLayerConfig" to "NextFrameConfig"
CL #133202 missed some documentation in vp8_frame_buffer_controller.h.
TBR=sprang@webrtc.org

Bug: None
Change-Id: I634f965c13a1cb179bcdc38f371e1c6e9380e4b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133909
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27721}
2019-04-23 15:20:27 +00:00
c1d209258a AEC3: Corrected the reading/creation of the json parameter file
Bug: webrtc:8671
Change-Id: Ide8693b3be023e56d3acca38e703c95ba0e9b69b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133623
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27714}
2019-04-23 13:29:53 +00:00
d8bf2d4986 Revert "Delete deprecated variant of VideoDecoder::Decode"
This reverts commit 3a86d9520c9692abe4c6d2a73faf4fb7a0cb80b2.

Reason for revert: breaks downstream project

Original change's description:
> Delete deprecated variant of VideoDecoder::Decode
> 
> Bug: webrtc:10379
> Change-Id: I4dd8b503625a9ea2a71177165238e128ac3e49bb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132554
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27712}

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

Change-Id: Ie971fd821f4de9e4b68e1608d7074835bdf2ed16
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10379
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133907
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27713}
2019-04-23 12:52:25 +00:00
3a86d9520c Delete deprecated variant of VideoDecoder::Decode
Bug: webrtc:10379
Change-Id: I4dd8b503625a9ea2a71177165238e128ac3e49bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132554
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27712}
2019-04-23 12:38:33 +00:00
38ac7bf0ab Add Vp8FrameConfig::IntraFrame()
Simple function to check if the frame described by Vp8FrameConfig
will be an intra frame.

Bug: None
Change-Id: I0ba0476762a152e1be3711352024fc6e1bd35f12
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133560
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27709}
2019-04-23 11:58:18 +00:00
428dcb2517 Remove SetLatency/GetLatency from MediaSourceInterface API level
Bug: webrtc:10287
Change-Id: I74fad31db98b75791085688438064f9510b0b6fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133165
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27692}
2019-04-18 19:11:31 +00:00
cab77fd1be Inject network state predictor into video quality test.
Bug: webrtc:10492
Change-Id: Ia2ae5de1019ac4ab89e54e261b1d94a482334ee9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133161
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27681}
2019-04-18 08:19:10 +00:00
2317c5ee0a Delete method DecodedImageCallback::ReceivedDecodedReferenceFrame
The code invoking it was deleted in
https://codereview.webrtc.org/2753783002

Tbr: kwiberg@webrtc.org # Change to mock class in api/test
Bug: webrtc:7408
Change-Id: I576d7aacd7dc60e42a05d2ea837fddf16594e685
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133348
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27680}
2019-04-18 08:14:40 +00:00
40889f35fc Removes TimeMicros interface from ThreadProcessingFakeClock.
Bug: webrtc:9883
Change-Id: Ib48872f81f734b09e3ffa4d9d26da79177b02303
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133341
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27668}
2019-04-17 15:37:48 +00:00
cf96e0f87d Implement RTCOutboundRtpStreamStats.retransmitted[Bytes/Packets]Sent.
Spec: https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedpacketssent

These are already existed in StreamDataCounters. This CL takes care of
the plumbing of these values to the standard stats collector.

TBR=solenberg@webrtc.org

Bug: webrtc:10447
Change-Id: I27d6c3ee3ab627d306303e6ee67e586ddf31cc81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132012
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27663}
2019-04-17 13:04:50 +00:00
979c4426a4 Rename "UpdateLayerConfig" to "NextFrameConfig"
Rename "UpdateLayerConfig" to the more appropriate "NextFrameConfig".
Also update some comments in vp8_frame_buffer_controller.h.

Bug: None
Change-Id: Iba8227f84e33e5ebd28d2eeb10fe03e776036603
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133202
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27660}
2019-04-17 11:59:07 +00:00
123ee9be8f OnLossNotification() receives references
A typo in a previous CL made OnLossNotification() accept its
single argument as a const-value, rather than a const-reference.

Bug: webrtc:10501
Change-Id: I5e6f9c79f15205b75ec90a53d3fccf3dd9927e33
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133343
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27659}
2019-04-17 11:47:25 +00:00
a3aa9bd75b Make VideoBitrateAllocatorFactory injectable.
This patch makes VideoBitrateAllocatorFactory injectable
by adding to PeerConnectionDependencies instead of allowing it to be
overridden using MediaEngine (on PeerConnectionFactory).

With this patch VideoBitrateAllocatorFactory is owned
by the PeerConnection.

WANT_LGTM (examples) : sakal@
WANT_LGTM (api/pc) : steveanton@

Bug: webrtc:10547
Change-Id: I768d400a621f2b7a98795eb7f410adb48651bfd6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132706
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27654}
2019-04-17 06:17:34 +00:00
01738c63aa Wire up RTCInboundRtpStreamStats.lastPacketReceivedTimestamp.
This collects this metric for both audio and video streams.
https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-lastpacketreceivedtimestamp

This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/130479
which calculated this metric. This CL is purely plumbing from
"StreamDataCounters::last_packet_received_timestamp_ms" to
RTCInboundRtpStreamStats.


Bug: webrtc:10449
Change-Id: I757ad19b5b8e84553da5edd4a75efa3e1fe30b56
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131397
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27628}
2019-04-15 16:06:01 +00:00
df88cc014a Allow injection of network estimator into GoogCC.
Bug: webrtc:10498
Change-Id: Ie9225411db201dfcfa0a37a3c40992acbdc215bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132402
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27624}
2019-04-15 14:12:08 +00:00
6796ec2289 Add OnFrameDropped() to Vp8FrameBufferController
Prior to this CL, this was indicated by passing |size_bytes| = 0
to the method.

Bug: webrtc:10501
Change-Id: Icff3bb83344834dc62d62bde5ec5d05096a08e11
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132712
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27620}
2019-04-15 12:35:45 +00:00
16cb8f5d74 Reland "Replace usage of old SetRates/SetRateAllocation methods"
This is a reland of 7ac0d5f348f0b956089c4ed65c46e65bac125508

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

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

Bug: webrtc:10481
Change-Id: I2978d5c527a18e885b7845c4e53a2424e8ad5b4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132551
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27593}
2019-04-12 13:37:32 +00:00
70f80e5962 Add support for creation of AEC dump during the test with PC framework.
Also add conversational speech into PC smoke test (with resource files).

Bug: webrtc:10138
Change-Id: I415a5565bc9146821476ffc60f57f47ed51f89c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132323
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27592}
2019-04-12 13:09:12 +00:00
806299e09b Introduce network emulation layer stats API.
Bug: webrtc:10138
Change-Id: I32133cd14c7a1933dcbeaa37d4c9ce6748274ebe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131383
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27588}
2019-04-12 12:08:06 +00:00
114871b1ba Expose SSL ciphersuite on the webrtc::DtlsTransport interface
This allows the ciphersuite to be accessed from Blink code, which is
useful if we want to emit deprecation messages.

Bug: none
Change-Id: Idcf1f5402948406e4cf7c6e54b67a622a1a403ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132004
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27581}
2019-04-12 08:29:59 +00:00
b113862ccd Allow log print of data units.
Bug: webrtc:9709
Change-Id: I5987a9779e645115dc1893944302a73d540bcf2f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125680
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27578}
2019-04-12 07:36:49 +00:00
4d7160e41d Code cleanup: Make JsepSessionDescription::Initialize take std::unique_ptr
Bug: none
Change-Id: I5e03ff6a6f16bd2e67fa4830e218b510b702d1d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132321
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27577}
2019-04-12 05:42:46 +00:00
70c961f965 Delete unused members of VideoCodecH264.
profile-level-id for H.264 comes in through the SdpVideoFormat,
rather than through these members.

Bug: None
Change-Id: I9c4ea8873346ca16174aecf5f90a649cbaf913dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132545
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27571}
2019-04-11 15:32:48 +00:00
52caa0ef58 AEC3: Configuration parameter for disabling linear filter
The configuration parameter filter.use_linear_filter can be used to
disable the linear filtering. Disabling the linear filter is equivalent
to runing in non-linear mode.

Bug: b/130016532
Change-Id: I8ffdf474822888b9915444bba6cc1c25ec1efe5a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132552
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27566}
2019-04-11 13:19:42 +00:00
7aacdd9515 Reland "Delete CodecSpecificInfo argument from VideoDecoder::Decode"
This is a reland of 39d3a7de02d63894d12e7332322e1d80cd7c0d40

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

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

Reason for revert: <INSERT REASONING HERE>

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

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

Change-Id: I576760b584e3f258013b0279c0c173c895bbb37e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10481
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132561
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27559}
2019-04-11 10:50:29 +00:00
7061e51b48 Expose DtlsTransport::remote_ssl_certificates
Bug: chromium:907849
Change-Id: If990d541099edb9a327230e1d78a03b406269885
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131951
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27558}
2019-04-11 09:47:33 +00:00
7ac0d5f348 Replace usage of old SetRates/SetRateAllocation methods
This rather large CL replaces all relevant usage of the old
VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
API is unchanged to allow downstream projects to update without
breakage.

Bug: webrtc:10481
Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27554}
2019-04-11 07:46:09 +00:00
b6ef99bb33 Translate loss notifications and pass to encoder
Translate LossNotification RTCP messages (sequence number to
timestamp and additional information), then send the translted
message onwards to the encoder.

Bug: webrtc:10501
Change-Id: If2fd943f75c36cf813a83120318d8eefc8c595d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131950
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27545}
2019-04-10 15:31:53 +00:00
0810a7c25a Add base class NetworkPredictor and NetworkPredictorFactory and wire up.
Add base class NetworkPredictor and NetworkPredictorFactory in /api, make it possible to inject customized NetworkPredictor in PeerConnectionFactory level. The NetworkPredictor object will be pass down to GoogCCNetworkControl and DelayBasedBwe.

Bug: webrtc:10492
Change-Id: Iceeadbe1c9388b11ce4ac01ee56554cb0bf64d04
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130201
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27543}
2019-04-10 12:38:58 +00:00
2af5dcbe9e Reland "Refactor FrameDecryptorInterface::Decrypt to use new API."
This reverts commit 7dd83e2bf73a7f1746c5ee976939bf52e19fa8be.

Reason for revert: This wasn't the cause of the break. 

Original change's description:
> Revert "Refactor FrameDecryptorInterface::Decrypt to use new API."
> 
> This reverts commit 642aa81f7d5cc55d5b99e2abc51327eed9d40195.
> 
> Reason for revert: Speculative revert. The chromium roll is failing:
> https://ci.chromium.org/p/chromium/builders/try/linux-rel/64388
> But I can't figure out exactly what is failing, this looks suspecious.
> 
> Original change's description:
> > Refactor FrameDecryptorInterface::Decrypt to use new API.
> > 
> > This change refactors the FrameDecryptorInterface to use the new API. The new
> > API surface simply moves bytes_written to the return type and implements a
> > simple Status type.
> > 
> > Bug: webrtc:10512
> > Change-Id: I622c5d344d58e618853c94c2f691cf7c8fb73a36
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131460
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27497}
> 
> TBR=brandtr@webrtc.org,steveanton@webrtc.org,solenberg@webrtc.org,ossu@webrtc.org,stefan@webrtc.org,benwright@webrtc.org
> 
> Change-Id: Ia9ec70263762c34671af13f0d519e636eb8473cd
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10512
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132013
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27510}

TBR=brandtr@webrtc.org,steveanton@webrtc.org,solenberg@webrtc.org,hbos@webrtc.org,ossu@webrtc.org,stefan@webrtc.org,benwright@webrtc.org

Change-Id: I8e4b7965cf1d1a1554c3b46e6245f5ad0d2dcbb4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10512
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131982
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27529}
2019-04-09 20:08:56 +00:00
6a489f22c7 Fully qualify googletest symbols.
Semi-automatically created with:

git grep -l " testing::" | xargs sed -i "s/ testing::/ ::testing::/g"
git grep -l "(testing::" | xargs sed -i "s/(testing::/(::testing::/g"
git cl format

After this, two .cc files failed to compile and I have fixed them
manually.

Bug: webrtc:10523
Change-Id: I4741d3bcedc831b6c5fdc04485678617eb4ce031
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132018
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27526}
2019-04-09 17:18:20 +00:00
cf267052b3 Field trial to control inter-layer prediction.
This adds WebRTC-Vp9InterLayerPred field trial that allows to control
inter-layer prediction mode in VP9 encoder.

Bug: chromium:949536
Change-Id: Iea03db07fd21f28ab58382c5fdaac68acacc701c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131322
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27521}
2019-04-09 13:40:55 +00:00
2e06926c95 Implement RTC[In/Out]boundRtpStreamStats.contentType.
Spec: https://henbos.github.io/webrtc-provisional-stats/#dom-rtcinboundrtpstreamstats-contenttype

This already exists as a goog-stat. This CL only plumbs the value to the
new stats collector.

Note: There is currently no distinction between the extension being
missing and it being present but the value being "unspecified". Until
https://crbug.com/webrtc/10529 is fixed, this metric is only exposed if
SCREENSHARE was present.

Bug: webrtc:10452
Change-Id: Ic8723f4d0efb43ab72a560e954676facd3b90659
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131946
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27520}
2019-04-09 13:02:03 +00:00
20789e477d Name changes inside VideoEncoder
Change names of two members of VideoEncoder to enhance clarity.

Bug: webrtc:10501
Change-Id: I6bcf64a5936f828bc585e54c429f24676ccfb0ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131941
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27514}
2019-04-09 11:27:10 +00:00
7dd83e2bf7 Revert "Refactor FrameDecryptorInterface::Decrypt to use new API."
This reverts commit 642aa81f7d5cc55d5b99e2abc51327eed9d40195.

Reason for revert: Speculative revert. The chromium roll is failing:
https://ci.chromium.org/p/chromium/builders/try/linux-rel/64388
But I can't figure out exactly what is failing, this looks suspecious.

Original change's description:
> Refactor FrameDecryptorInterface::Decrypt to use new API.
> 
> This change refactors the FrameDecryptorInterface to use the new API. The new
> API surface simply moves bytes_written to the return type and implements a
> simple Status type.
> 
> Bug: webrtc:10512
> Change-Id: I622c5d344d58e618853c94c2f691cf7c8fb73a36
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131460
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27497}

TBR=brandtr@webrtc.org,steveanton@webrtc.org,solenberg@webrtc.org,ossu@webrtc.org,stefan@webrtc.org,benwright@webrtc.org

Change-Id: Ia9ec70263762c34671af13f0d519e636eb8473cd
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10512
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132013
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27510}
2019-04-09 10:36:48 +00:00
f3736ed3d8 Datachannel: Use absl::optional for maxRetransmits and maxRetransmitTime.
These parameters are nullable in the JS API.
This allows cleaner handling of "unset" vs "set" in Chrome.

Backwards compatibility note: Behavior should not change, even for users
who set the values explicitly to -1 in the DataChannelInit struct.
Those who try to read back the value will get a compile-time error.

Bug: chromium:854385
Change-Id: Ib488ca5f70bc24ba8b4a3f71b506434c4d2c60b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131381
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27507}
2019-04-09 08:32:43 +00:00
b703db9bce Fix and test CreateVideoStreamDecoder
create TaskQueue using provided factory instead of through constructor that uses GlobalTaskQueueFactory
Rename interface to avoid collision with class in video/video_stream_decoder.h
Add simple unittest to avoid future breakages.

Bug: webrtc:10284, webrtc:10521
Change-Id: I647b31cc99a2b6beb79b936f05f482788861f1cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131398
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27503}
2019-04-09 07:57:18 +00:00
f71362f0cf Wire up RTCOutboundRtpStreamStats.totalEncodeTime.
This is a follow-up to
https://webrtc-review.googlesource.com/c/src/+/130517 that calculated
this metric.

This CL is purely plumbing, exposing
VideoSendStream::total_encode_time_ms in standard getStats() as
RTCOutboundRtpStreamStats.totalEncodeTime (in seconds):
https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalencodetime

Bug: webrtc:10448
Change-Id: I715f1ef937e441169dee55b5e8d4fbf98811c5f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131940
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27501}
2019-04-09 07:34:38 +00:00
642aa81f7d Refactor FrameDecryptorInterface::Decrypt to use new API.
This change refactors the FrameDecryptorInterface to use the new API. The new
API surface simply moves bytes_written to the return type and implements a
simple Status type.

Bug: webrtc:10512
Change-Id: I622c5d344d58e618853c94c2f691cf7c8fb73a36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131460
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27497}
2019-04-08 20:45:09 +00:00