2582 Commits

Author SHA1 Message Date
Artem Titov
089758dbc5 Allow creation of TestVideoTrackSource not on the signaling thread
Bug: b/272350185
Change-Id: I1bc18f35e2d0b36791966a5954eb28886c569a9b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299261
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39704}
2023-03-28 13:33:35 +00:00
Harald Alvestrand
d32e5b3078 Deprecate non-refcount CreateVideoTrack
Bug: webrtc:15017
Change-Id: I978a14dcb2fac7777a12c3f89b1a7207dd896b37
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299075
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39700}
2023-03-28 11:58:36 +00:00
Danil Chapovalov
e14abcb20b Cleanup FieldTrialView
Delete alias WebRtcKeyValueConfig as unused
Replace .find() == 0 with absl::StartsWith per clang-tidy recommendation
https://clang.llvm.org/extra/clang-tidy/checks/abseil/string-find-startswith.html

Bug: webrtc:10335
Change-Id: I1f09c262844c0678a8d8c0d0d3274df3d947737c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299181
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39690}
2023-03-27 17:06:33 +00:00
Jeremy Leconte
a3f7b54518 [DVQA] Don't check if peer exists on Pause/Resume.
Change-Id: I0f26444c6a420017caaa4c27520e75e4146ecfd4
Bug: webrtc:14995
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299077
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39689}
2023-03-27 16:23:52 +00:00
Artem Titov
6fd5f33d45 Extend TestVideoTrackSource API
Bug: b/272350185
Change-Id: Ibc53e7a9ee8f572475d86fc78de1c1ed71078910
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299140
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39678}
2023-03-25 21:35:23 +00:00
Jakob Ivarsson
94b51210f8 Include packet waiting time in concealment decision.
This is to be more robust to packet loss during DTX and paused streams.

Without it, we can wait to decode an available packet when in CNG or
PLC mode until more packets arrive, which for DTX and paused streams
can take a long time.

We already include the waiting time if the last packet in the buffer
is a DTX packet.

Bug: webrtc:13322
Change-Id: Iaf5b3894500140d6f83377ba2cd65b44e0cdac05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299009
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39667}
2023-03-24 13:18:58 +00:00
Tommi
c848268ab1 Use SequenceChecker(SequenceChecker::kDetached) in a few places.
This CL is partly a test to see if there's an impact on binary size:
- Not a big difference for binaries (decrease): -776b to -4Kb
- For libraries (libwebrtc.a) it actually increases the size: +40Kb

Secondarily this CL is basically to introduce this pattern to the
code base. In terms of LOC, this makes things slightly more compact.

From:

  class Foo {
   public:
     Foo() {
       checker_.Detach();
     }
   private:
    SequenceChecker checker_;
  };

To:

  class Foo {
   public:
     Foo() = default;
   private:
    SequenceChecker checker_{SequenceChecker::kDetached};
  };

Bug: none
Change-Id: I59fc34ccea10847e13455a349851ce9a0af458e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299020
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39664}
2023-03-24 07:44:18 +00:00
Henrik Boström
4baea5b07f Make VP9 simulcast behave like singlecast for single active layer cases.
Various "if streams == 1" cases are updated to "if
IsSinglecastOrAllNonFirstLayersInactive()" in order not to cause subtle
differences between VP9 {active} and VP9 {active,inactive,inactive}.

This CL also affects a line that conditionally sets
`simulcastStream[0].active = codec_active` so it seemed fitting to
improve the test coverage of "if all streams are inactive, don't send".

Bug: webrtc:15028
Change-Id: I8872dc8be0f2dfc1d8914bdba5e6433f9ba8cbfd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298881
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39656}
2023-03-23 14:49:22 +00:00
Per K
7effd7657b Change visibility on target mock_network_control
This is to allow external tests to depend on it.

Bug: none
Change-Id: Ic8e2f864041d959f673e7f2c18eb563a13274dcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298745
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39646}
2023-03-22 19:53:04 +00:00
Per K
452d94047b Add mock for NetworkControllerInterface
Bug: none
Change-Id: Ibdd72011932a36348a4382caa5d0bf0ab2c02dd9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298742
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39641}
2023-03-22 15:49:32 +00:00
Harald Alvestrand
041ecb87f5 New PeerConnectionFactory::CreateVideoTrack with refcounted source
Bug: webrtc:15017
Change-Id: I04c794d8959583bb4cc5c3898f4175783ec49f16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249363
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39635}
2023-03-22 09:10:27 +00:00
Philipp Hancke
f0f435e983 Remove deprecated RTCStatsReport(int64) and timestamp_us
BUG=webrtc:14813

Change-Id: I80c2ba8f57354ef63cf2cc7b767d1f64dd0dd766
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298444
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39633}
2023-03-22 08:00:53 +00:00
Sergey Silkin
ebb5383fd8 Dump codec input
Add functionality for dumping encoder and decoder input to file in video codec test.

Bug: b/261160916, webrtc:14852
Change-Id: I49a84a886d87903c601cf5c35bd723b6393c2a75
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298051
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39626}
2023-03-21 16:54:19 +00:00
Henrik Boström
62dc65b537 Add test that attempting HW is still possible after SW fallback.
Based on previous discussions I would have thought that this test would
fail, but it turns out that it passes. See referenced bug for context.

Bug: webrtc:15021
Change-Id: I845b48f688fb25942e3b770d50cafbf8a0bafe94
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298562
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39625}
2023-03-21 15:02:34 +00:00
Artem Titov
6a78e93346 [PCLF] Introduce test video source and make it more controllable
Bug: b/272350185
Change-Id: I15572b7e4d0cb0ce41da676a4eedbc1e138510fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298047
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39621}
2023-03-21 14:15:24 +00:00
Jeremy Leconte
2148f8ed71 [DVQA] Change API to pause and resume all streams from a sender.
Also make it possible to pause an already paused stream by making it a no-op.

Change-Id: Id10f74a4c6464067ae63208162194f020c6470eb
Bug: b/271542055
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298202
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39620}
2023-03-21 14:05:03 +00:00
Jonas Oreland
122d777943 Add new stun attribute GOOG_DELTA_SYNC_REQ
Assigned by IANA: https://www.iana.org/assignments/stun-parameters/stun-parameters.xhtml

Bug: webrtc:0
Change-Id: Ie910e112afe33f3dbf7f2a221edc96af5ac7b139
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298560
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39617}
2023-03-21 13:28:43 +00:00
Tommi
3da04a93cd Allow SequenceChecker to be initialized detached.
The motivation for this is to not have to implement this pattern:

foo.h:

class Foo {
 public:
  Foo();
 private:
  SequenceChecker checker_;
};

foo.cc:

Foo::Foo() {
  checker_.Detach();
}

And instead be able to do this inline in the .h file:

class Foo {
 public:
  Foo();
 private:
  SequenceChecker checker_{SequenceChecker::kDetached};
};

Bug: none
Change-Id: Idd7ca82d15c2f77f3aaccf26f1943a49f4b40661
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298445
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39616}
2023-03-21 12:34:15 +00:00
Sergey Silkin
aa17f2f0a9 Add Initialize() to Encoder/Decoder API in video codec tester
Initialization of Android HW codecs takes hundreds milliseconds. Exclude this time from frame processing time of first frame by initializing codecs before starting encoding/decoding.

Bug: b/261160916, webrtc:14852
Change-Id: I9ec84c6b12c1d9821b59965cf521170224066563
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298304
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39613}
2023-03-21 08:04:48 +00:00
Danil Chapovalov
a2d85e4565 Use absl::string_view type as parameter for RTCError message
Bug: webrtc:13579
Change-Id: Ia9f90e6c3b008fc614d378cae4c407becfc597c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298447
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39610}
2023-03-20 18:16:10 +00:00
Jakob Ivarsson
63643357b4 Remove CNG state tracking from NetEq decision logic.
It seems like this is legacy and not useful. A comment mentions
transitioning between CNG and DTMF modes, but there is no way this can
happen currently.

Bug: webrtc:13322
Change-Id: I9e4706cb6ee145ee37a9e11e7cab6ea4ff697dc4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297980
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39590}
2023-03-17 15:00:17 +00:00
Henrik Boström
f6eae959bf Delete EncoderSimulcastProxy in favor of SimulcastEncoderAdapter.
Because the adapter has a passthrough mode, it can already handle both
singlecast and simulcast cases, meaning the proxy is no longer providing
value. Let's delete.

Bug: webrtc:15001
Change-Id: I480eaba599448e9b82b8cf7f829dc35ad6ce0434
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297740
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39579}
2023-03-16 13:25:44 +00:00
Philipp Hancke
b3e5969658 stats: use uint64_t for RTCSentRtpStreamStats.packetsSent
spec update from https://github.com/w3c/webrtc-stats/pull/744

BUG=webrtc:14989

Change-Id: I9d0adcf951501bc281054c77bb6bc03e47192523
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295505
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39575}
2023-03-16 06:46:19 +00:00
Artem Titov
ebce84a502 [DVQA] Add support for DVQA to pause/resume receiving of stream by peer
Bug: b/271542055, webrtc:14995
Change-Id: Ic02451347160f512588b6fef5d6ac4ad904b5e18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297440
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39568}
2023-03-15 18:16:49 +00:00
Henrik Boström
9a5de95af9 Add a flag to control legacy vs spec-compliant scalability mode.
The goal of the VP9 simulcast project is that when `scalability_mode`
is set, multiple encodings are always interpreted as simulcast, even
if VP9 or AV1 is used. This CL makes this so, but only if the flag
"WebRTC-AllowDisablingLegacyScalability" is "/Enabled/". This allows us
to make "SendingThreeEncodings_VP9_Simulcast" EXPECT VP9 simulcast.

When we are ready to ship we will remove the need to use the field
trial, but before we ship this we'll want to revisit if
SvcRateAllocator can be updated to support simulcast. (Today if we use
SvcRateAllocator when VP9 simulcast is used, all encodings except the
first one get bitrate=0, causing the test to fail because media is not
flowing on all layers.) For now, a TODO is added.

Bug: webrtc:14884
Change-Id: Ie20ae748b0c0405162f3a1b015ab94956ef83dae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297340
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39552}
2023-03-14 12:05:24 +00:00
Philipp Hancke
22005ab39b Remove obsolete header extension API names
and update spec link.

BUG=chromium:1051821

Change-Id: I42dbe36b2299f01cb4eb8010c893623fde7472fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296702
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39548}
2023-03-13 14:49:05 +00:00
Philipp Hancke
d3289d2ec0 Reland "stats: remove RTCRtpInboundRTPStream and RTCRtpoutboundRTPStream aliases"
This is a reland of commit 9671d60925b81baefd4a0d6b05ad539fa4a782d7
after fixing more downstream dependencies

Original change's description:
> stats: remove RTCRtpInboundRTPStream and RTCRtpoutboundRTPStream aliases
>
> after upgrading downstream projects
>
> BUG=webrtc:14973
>
> Change-Id: I5df8e95a1c70b1d6078e255166c36ed01f868b6a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296820
> Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Cr-Commit-Position: refs/heads/main@{#39526}

No-Try: True
Bug: webrtc:14973
Change-Id: I33bd99ca211a82ed77e3e8676e00256915fde168
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296881
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39533}
2023-03-10 15:22:01 +00:00
Henrik Boström
4463ff0296 Revert "stats: remove RTCRtpInboundRTPStream and RTCRtpoutboundRTPStream aliases"
This reverts commit 9671d60925b81baefd4a0d6b05ad539fa4a782d7.

Reason for revert: Breaks dependencies, will re-land after fixes

Original change's description:
> stats: remove RTCRtpInboundRTPStream and RTCRtpoutboundRTPStream aliases
>
> after upgrading downstream projects
>
> BUG=webrtc:14973
>
> Change-Id: I5df8e95a1c70b1d6078e255166c36ed01f868b6a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296820
> Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Cr-Commit-Position: refs/heads/main@{#39526}

Bug: webrtc:14973
Change-Id: I50878526566660d9772f7c8664970eec8bd86341
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296940
Reviewed-by: Philipp Hancke <phancke@microsoft.com>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39530}
2023-03-10 13:24:32 +00:00
Philipp Hancke
9671d60925 stats: remove RTCRtpInboundRTPStream and RTCRtpoutboundRTPStream aliases
after upgrading downstream projects

BUG=webrtc:14973

Change-Id: I5df8e95a1c70b1d6078e255166c36ed01f868b6a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296820
Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39526}
2023-03-10 12:12:41 +00:00
Artem Titov
d877589e16 Make frame_generator_api not testonly
Bug: None
Change-Id: Id51f819df350f24b12ead698e4f75dcb760d18f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296821
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39519}
2023-03-09 21:54:34 +00:00
Danil Chapovalov
b40aedf911 Delete RTPHeader::payload_type_frequency as unused
payload type frequency is not communicated inside an RTP packet and
thus do not belong to the RTPHeader

Bug: None
Change-Id: Ic3e48f1b0507a96ddc697503145f7c8785962926
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296763
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39515}
2023-03-09 16:32:22 +00:00
Philipp Hancke
1f98b466b8 stats: rename RTCInboundRTPStreamStats and RTCOutboundRTPStreamStats
to RTCInboundRtpStreamStats and RTCOutboundRtpStreamStats respectively
which follows the camel-casing convention used elsewhere.

The old name is kept around as an alias for a limited amount of time
to allow upgrading dependencies.

BUG=webrtc:14973

Change-Id: Ibf4e65933fd6cc2e7e89955042f6f8fb0f6c7853
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296261
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39497}
2023-03-07 14:27:47 +00:00
Philipp Hancke
9f6ae375e3 Rename header extension API methods
following spec updates from
  https://github.com/w3c/webrtc-extensions/pull/142

BUG=chromium:1051821

Change-Id: I1fd991a5024d38ac59ebe510ea1a48fd6f42d23b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296321
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39491}
2023-03-07 10:55:58 +00:00
Sergey Silkin
1c1382be0f Dump codec output to ivf/y4m
Bug: b/261160916, webrtc:14852
Change-Id: I19de2210aa03b56752db5ce8b6fd94498123d6f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296260
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39490}
2023-03-07 08:33:39 +00:00
Sergey Silkin
9259b5f72c Add rate adaptation tests
Bug: b/261160916, webrtc:14852
Change-Id: I58b3647218c961dcf0305c3902f79adb448b73e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295866
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39489}
2023-03-06 18:33:16 +00:00
Markus Handell
eb277527f0 Stop Posting tasks when we don't need to.
Under the combined network/worker thread project, tasks
are unnecessarily posted to the same thread.

This CL reaps 90% overhead savings in sent packet feedback
as measured with Perfetto under a 49p Meet call.

The identity of the posted calls was uncovered with WebRTC/Chrome's
new location-aware tracing.

TESTED=presubmit + local Meet calls.

Bug: chromium:1373439
Change-Id: I0c43aa4de884831838747d52b21c45fd360106e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295780
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39484}
2023-03-06 15:13:39 +00:00
Philipp Hancke
1f80451932 Fix stats inheritance and rename RTP to Rtp
making RTCOutboundRtpStreamStats inherit from RTCSentRtpStreamStats
as defined in
  https://www.w3.org/TR/webrtc-stats/#outboundrtpstats-dict*

This removes the duplicated definitions of packetsSent and bytesSent.

BUG=webrtc:14948

Change-Id: I184998b65d59dbd0d1288733d55d8a884e6de970
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295660
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39481}
2023-03-06 13:43:27 +00:00
Tove Petersson
1e2d951762 Add a clone method to the audio frame transformer API.
This will clone an encoded audio frame into a sender frame.

Bug: webrtc:14949
Change-Id: Ie62d9f5ec457541b335bde8f2f6e9b6d24704cf6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294560
Commit-Queue: Tove Petersson <tovep@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39480}
2023-03-06 08:22:25 +00:00
Danil Chapovalov
a76487ffd2 Relax string parameters in pclf api to absl::string_view
Bug: webrtc:13579
Change-Id: I53c133bcbba6a074f3be6b996a3991a71190b1fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295865
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39459}
2023-03-02 16:17:41 +00:00
Danil Chapovalov
298975aa89 Cleanup legacy name for VideoPlayoutDelay
Bug: webrtc:7660
Change-Id: Icdeaca06224def0effb304c8492ecdd64cc82e52
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295861
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39458}
2023-03-02 15:27:02 +00:00
Tony Herre
2311f93909 Remove uses of TransformableVideoFrame::GetMetadata and deprecate it
Chromium uses have been migrated to Metadata(), so we should be clear.
Other projects can easily migrate similarly.

Bug: chromium:1420245
Change-Id: I150654812676dabd5c957cff00d40d4c95eaf5d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295481
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39455}
2023-03-02 13:38:48 +00:00
Philipp Hancke
7f4270d160 Remove JsepSessionDescription::kDefaultVideoCodecName
which is only used in tests.

BUG=None

Change-Id: If215ad84e6756af2ee90777a27376400f8f4d8e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294721
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39450}
2023-03-02 12:28:29 +00:00
Alan Zhao
6cf8b486eb Fix missing libc++ includes in webrtc
Several files refer to symbols declared in headers not explicitly
included. This compiles now because libc++ tranitively includes these
headers via other libc++ headers; however, these transitive includes are
not guaranteed to exist and in Chrome, will no longer exist once libc++
is compiled with modules.

Bug: chromium:543704
Change-Id: I638bb02df3d050a48345248e80aebd2dd60956c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295762
Auto-Submit: Alan Zhao <ayzhao@google.com>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39448}
2023-03-02 10:14:51 +00:00
Emil Lundmark
4e86aa0870 Remove mentions of already deleted field trials
- WebRTC-Audio-Agc2ForceExtraSaturationMargin
- WebRTC-Audio-Agc2ForceInitialSaturationMargin
- WebRTC-Audio-BitrateAdaptation
- WebRTC-Audio-TransientSuppressorVadMode
- WebRTC-FrameBuffer3
- WebRTC-IntelVP8
- WebRTC-UseActiveIceController

Bug: None
Change-Id: I3545727c09f761867f2f4c2bb5c400012ce146d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295723
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39444}
2023-03-01 15:53:37 +00:00
Markus Handell
ae61aca9b1 Implement support for Chrome task origin tracing. #3.7/4
This CL completes migration to the new TaskQueueBase interface
permitting location tracing in Chrome.

Bug: chromium:1416199
Change-Id: Iff7ff5796752a1520384a3db0135a1d4b9438988
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294540
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39439}
2023-03-01 14:20:03 +00:00
Markus Handell
a1ceae206b Implement support for Chrome task origin tracing. #3.5/4
This CL migrates unit tests to the new TaskQueueBase interface.

Bug: chromium:1416199
Change-Id: Ic15c694b28eb67450ac99fdd56754de1246a4d95
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295621
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39434}
2023-03-01 11:11:37 +00:00
Sergey Silkin
fddc9131a5 Aggregate and log video codec metrics
Bug: b/261160916, webrtc:14852
Change-Id: Idcb7e96b12ca38af49b9b1f10d1e23cc7faac92b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293945
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39427}
2023-03-01 08:27:54 +00:00
Tony Herre
a6135bcd43 Remove deprecated TransformableVideoFrame::GetAdditionalData
It was marked deprecated on Feb 9th, ~3 weeks ago.

Bug: chromium:1414370
Change-Id: I251b91984ca9a958e221f6eaf01c63b05c5a7a48
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295506
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#39422}
2023-02-28 16:23:52 +00:00
Tove Petersson
1fccaa4485 Reland "operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test"
This reverts commit 8bf321062973939ef35f529640f5e69852e89a7e.

Reason for revert: Initialized an uninitialized member in GofInfoVP9 (+ removed some redundant initialization of members already initialized by SetGofInfoVP9())

Original change's description:
> Revert "operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test"
>
> This reverts commit 437bf78ed9518b21fc39b94f6ee42d5b157e6084.
>
> Reason for revert: Breaks upstream project
>
> Original change's description:
> > operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test
> >
> > Added equality and inequality operators for VideoFrameMetadata and used the equality operator to check that the cloned metadata property is equal to the original metadata in RtpSenderVideoFrameTransformerDelegateTest.CloneSenderVideoFrame.
> >
> > Also default-initialized VideoFrameMetadata::ssrc_ to 0.
> >
> > Bug: webrtc:14708
> > Change-Id: If1f5153069bc986061ff9f0a6abaa2a4a5a98dd1
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293560
> > Commit-Queue: Tove Petersson <tovep@google.com>
> > Reviewed-by: Tony Herre <herre@google.com>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#39411}
>
> Bug: webrtc:14708
> Change-Id: Icbec1b65ed22b89766606cb9514dde6f4e9124be
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295500
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Auto-Submit: Andrey Logvin <landrey@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39413}

Bug: webrtc:14708
Change-Id: I843d29f7dd0da2c7f16968a7fc08dc02cd359fc1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295520
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tove Petersson <tovep@google.com>
Cr-Commit-Position: refs/heads/main@{#39418}
2023-02-28 15:44:21 +00:00
Emil Lundmark
9109e856d5 Add option to log a warning for unregistered field trials
Until now you only had the option to RTC_DCHECK for unregistered field
trials. This makes it possible to log a warning instead.

Bug: webrtc:14154
Change-Id: I8628054e3c9b5d690f241a93e61299126b732ed0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295300
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39417}
2023-02-28 15:43:18 +00:00