43 Commits

Author SHA1 Message Date
Mirko Bonadei
54c90f2330 [-Wshadow] - Fix some warnings.
First CL to try to understand the extent of the cleanup needed in
order to remove -Wno-shadow and follow Chromium on enabling this
diagnostic.

Bug: webrtc:13219
Change-Id: Ie699762da50fe3dbc08b1fd92220962d4b7da86b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233641
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35134}
2021-10-03 11:53:16 +00:00
Artem Titov
880fa8169b Reland "Use backticks not vertical bars to denote variables in comments for /pc"
Original change's description:
> Revert "Use backticks not vertical bars to denote variables in comments for /pc"
>
> This reverts commit 37ee0f5e594dd772ec6d620b5e5ea8a751b684f0.
>
> Reason for revert: Revert in order to be able to revert https://webrtc-review.googlesource.com/c/src/+/225642
>
> Original change's description:
> > Use backticks not vertical bars to denote variables in comments for /pc
> >
> > Bug: webrtc:12338
> > Change-Id: I88cf10afa5fc810b95d2a585ab2e895dcc163b63
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226953
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Artem Titov <titovartem@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#34575}
>
> TBR=hta@webrtc.org,titovartem@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
>
> Change-Id: I5eddd3a14e1f664bf831e5c294fbc4de5f6a88af
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:12338
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227082
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34577}

Bug: webrtc:12338
Change-Id: I96bd229b73613c162d11d75fa4f5934e1b4295c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227087
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34611}
2021-07-30 22:13:59 +00:00
Björn Terelius
fd05d6f504 Revert "Use backticks not vertical bars to denote variables in comments for /pc"
This reverts commit 37ee0f5e594dd772ec6d620b5e5ea8a751b684f0.

Reason for revert: Revert in order to be able to revert https://webrtc-review.googlesource.com/c/src/+/225642

Original change's description:
> Use backticks not vertical bars to denote variables in comments for /pc
>
> Bug: webrtc:12338
> Change-Id: I88cf10afa5fc810b95d2a585ab2e895dcc163b63
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226953
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34575}

TBR=hta@webrtc.org,titovartem@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com

Change-Id: I5eddd3a14e1f664bf831e5c294fbc4de5f6a88af
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12338
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227082
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34577}
2021-07-27 22:10:24 +00:00
Artem Titov
37ee0f5e59 Use backticks not vertical bars to denote variables in comments for /pc
Bug: webrtc:12338
Change-Id: I88cf10afa5fc810b95d2a585ab2e895dcc163b63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226953
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34575}
2021-07-27 20:52:02 +00:00
Mirko Bonadei
7750d802a5 Rename rtc_base/ssl_stream_adapter.h constants.
Uppercase constants are more likely to conflict with macros (for
example rtc::SRTP_AES128_CM_SHA1_80 and OpenSSL SRTP_AES128_CM_SHA1_80).

This CL renames some constants and follows the C++ style guide.

Bug: webrtc:12997
Change-Id: I2398232568b352f88afed571a9b698040bb81c30
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226564
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34553}
2021-07-26 16:33:54 +00:00
Jakob Ivarsson
e54914a79e Implement nack_count metric for inbound audio rtp streams.
Bug: webrtc:12925
Change-Id: I4542ca0f14a7dd7485ad5a2b6f2bd7051076f71f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224085
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34401}
2021-07-01 10:38:44 +00:00
Tommi
2e1546887b Avoid generating a random id for candidate stats.
CandidateStats didn't use an initializer list which caused the
`candidate` member variable to be constructed with a random id
(calling an expensive rng method), only to be overwritten directly
thereafter.

Bug: webrtc:12840
Change-Id: I0366f674281d236896cb9539812dc2d88c1b37ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221600
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34244}
2021-06-08 08:00:01 +00:00
Markus Handell
518669d6d4 Add more trace events to interesting places.
Bug: webrtc:12840
Change-Id: I57e5373ae33060bd3743cea8ada21c845cbbd944
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221365
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34237}
2021-06-07 13:43:07 +00:00
Tommi
4ccdf932e1 VideoRtpReceiver & AudioRtpReceiver threading fixes.
For implementations where the signaling and worker threads are not
the same thread, this significantly cuts down on Thread::Invoke()s that
would block the signaling thread while waiting for the worker thread.

For Audio and Video Rtp receivers, the following methods now do not
block the signaling thread:
* GetParameters
* SetJitterBufferMinimumDelay
* GetSources
* SetFrameDecryptor / GetFrameDecryptor
* SetDepacketizerToDecoderFrameTransformer

Importantly this change also makes the track() accessor accessible
directly from the application thread (bypassing the proxy) since
for receiver objects, the track object is const.

Other changes:

* Remove RefCountedObject inheritance, use make_ref_counted instead.
* Every member variable in the rtp receiver classes is now RTC_GUARDED
* Stop() now fully clears up worker thread state, and Stop() is
  consistently called before destruction. This means that there's one
  thread hop instead of at least 4 before (sometimes more), per receiver.
* OnChanged triggered volume for audio tracks is done asynchronously.
* Deleted most of the JitterBufferDelay implementation. Turns out that
  it was largely unnecessary overhead and complexity.

It seems that these two classes are copy/pasted to a large extent
so further refactoring would be good in the future, as to not have to
fix each issue twice.

Bug: chromium:1184611
Change-Id: I1ba5c3abbd1b0571f7d12850d64004fd2d83e5e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218605
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34022}
2021-05-17 14:37:55 +00:00
Tomas Gunnarsson
788d805c38 Reland "Remove Invoke from VideoChannel::FillBitrateInfo."
This reverts commit 48a4d33719390b7bcaf8445a1581a00825f67bfb.

Reason for reland:

Relanding the original change but without the modification for
VideoSendStream::GetStats. Essentially there's a TODO there to fix
the downstream issue, which seems to be benign.

Original change's description:
> Revert "Remove Invoke from VideoChannel::FillBitrateInfo."
>
> This reverts commit 1a1795768e1bdb65054ebe15aa238c6edc78dd14.
>
> Reason for revert: Speculative revert (breaks downstream project).
>
> Original change's description:
> > Remove Invoke from VideoChannel::FillBitrateInfo.
> >
> > The method is relied upon by StatsCollector where it was called from the
> > signaling thread in a loop. Now there's at most one invoke (not N).
> >
> > Uncommenting thread checks and removing TODOs in SendStatisticsProxy,
> > VideoSendStream. Updating all related tests that fetched stats from
> > the wrong context.
> >
> > Bug: webrtc:12726
> > Change-Id: Ia7db1afd7e103ec4f9816f5647203c4e2495586e
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216688
> > Commit-Queue: Tommi <tommi@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33894}
>
> TBR=ilnik@webrtc.org,nisse@webrtc.org,tommi@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
>
> Change-Id: I2520957cdb33492d187f04320c7416788fd0f820
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:12726
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217240
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33898}

# Not skipping CQ checks because this is a reland.

Bug: webrtc:12726
Change-Id: I41cce3b11a29905cde982c22e82b9b1f5a98e654
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217222
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33902}
2021-05-03 15:16:34 +00:00
Mirko Bonadei
48a4d33719 Revert "Remove Invoke from VideoChannel::FillBitrateInfo."
This reverts commit 1a1795768e1bdb65054ebe15aa238c6edc78dd14.

Reason for revert: Speculative revert (breaks downstream project).

Original change's description:
> Remove Invoke from VideoChannel::FillBitrateInfo.
>
> The method is relied upon by StatsCollector where it was called from the
> signaling thread in a loop. Now there's at most one invoke (not N).
>
> Uncommenting thread checks and removing TODOs in SendStatisticsProxy,
> VideoSendStream. Updating all related tests that fetched stats from
> the wrong context.
>
> Bug: webrtc:12726
> Change-Id: Ia7db1afd7e103ec4f9816f5647203c4e2495586e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216688
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33894}

TBR=ilnik@webrtc.org,nisse@webrtc.org,tommi@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com

Change-Id: I2520957cdb33492d187f04320c7416788fd0f820
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12726
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217240
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33898}
2021-05-03 12:41:25 +00:00
Tommi
1a1795768e Remove Invoke from VideoChannel::FillBitrateInfo.
The method is relied upon by StatsCollector where it was called from the
signaling thread in a loop. Now there's at most one invoke (not N).

Uncommenting thread checks and removing TODOs in SendStatisticsProxy,
VideoSendStream. Updating all related tests that fetched stats from
the wrong context.

Bug: webrtc:12726
Change-Id: Ia7db1afd7e103ec4f9816f5647203c4e2495586e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216688
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33894}
2021-05-03 12:12:30 +00:00
Tomas Gunnarsson
bfd9ba8802 Fix unsafe variable access in RTCStatsCollector
With this change, all production callers of BaseChannel::transport_name()
will be making the call from the right thread and we can safely delegate
the call to the transport itself. Some tests still need to be updated.
This facilitates the main goal of not needing synchronization inside
of the channel classes, being able to apply thread checks and eventually
remove thread hops from the channel classes.

A downside of this particular change is that a blocking call to the
network thread from the signaling thread inside of RTCStatsCollector
needs to be done. This is done once though and fixes a race.

Bug: webrtc:12601, webrtc:11687, webrtc:12644
Change-Id: I85f34f3341a06da9a9efd936b1d36722b10ec487
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213080
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33775}
2021-04-19 16:22:23 +00:00
Niels Möller
572f50fc04 Delete left-over references to AsyncInvoker
Bug: webrtc:12339
Change-Id: I16c7e83a043939e76ee7cd0cb9402bc08584eb6c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213142
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33742}
2021-04-15 10:43:00 +00:00
Tomas Gunnarsson
e1c8a43b2a Reduce thread hops in StatsCollector and fix incorrect variable access.
StatsCollector::ExtractSessionInfo was run fully on the signaling thread
and several calls were being made to methods that need to run on the
network thread.

Additionally, BaseChannel::transport_name() was being read directly
on the signaling thread (needs to be read on the network thread).
So with shifting the work that needs to happen on the network thread
over to that thread, we now also grab the transport name there and
use the name with the work that still needs to happen on the signaling
thread.

These changes allow us to remove Invoke<>() calls to the network thread from
callback functions implemented in PeerConnection:
* GetPooledCandidateStats
* GetTransportNamesByMid
* GetTransportStatsByNames
* Also adding a correctness thread check to:
  * GetLocalCertificate
  * GetRemoteSSLCertChain

Because PeerConnection now has a way of knowing when things are
or have been uninitialized on the network thread, all of these
functions can exit early without doing throw away work.

Additionally removing thread hops that aren't needed anymore from
JsepTransportController.

Using the RTC_LOG_THREAD_BLOCK_COUNT() macro in GetStats, the number
of Invokes (when >1), goes down by 3. Typically from 8->5, 7->4, 6->3.

Bug: webrtc:11687
Change-Id: I06ab25eab301e192e99076d7891444bcb61b491f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214135
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33656}
2021-04-08 14:06:20 +00:00
Taylor Brandstetter
80d661cb7a Use monotonic clock for legacy stats caching.
Fixes a bug where, if the wall clock goes backwards (from changing time
zone, daylight savings, manual change by user, etc.), the stats
collector stops updating stats.

Bug: webrtc:12439
Change-Id: Ie45f484110d43444aea6d6b1a218b05d0f8b6a0c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206622
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33240}
2021-02-11 21:45:30 +00:00
Artem Titov
d15a575ec3 Use SequenceChecker from public API
Bug: webrtc:12419
Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 15:04:55 +00:00
Harald Alvestrand
5761e7b3ff Running apply-iwyu on ~all files in pc/
Bug: none
Change-Id: Ieebdfb743e691f7ae35e1aa354f68ce9e771064d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204381
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33105}
2021-01-29 16:14:10 +00:00
Harald Alvestrand
445e6b034a Break out separate compile targets for various classes
This reduces the degree of interdependency among modules related
to the PeerConnection class, and makes it easier to isolate inappropriate
external dependencies.

Bug: webrtc:11967
Change-Id: Id9777a2ab690cc349dd5842a3a95e24478144c71
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185882
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32235}
2020-09-29 15:14:22 +00:00
Niels Möller
6b4d962947 Fix standard GetStats to not modify NetEq state.
Add a get_and_clear_legacy_stats flag to AudioReceiveStream::GetStats,
to distinguish calls from standard GetStats and legacy GetStats.

Add const method NetEq::CurrentNetworkStatistics to get current
values of stateless NetEq stats. Standard GetStats will then call this
method instead of NetEq::NetworkStatistics.

Bug: webrtc:11622
Change-Id: I3833a246a9e39b18c99657a738da22c6e2bd5f5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183600
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32092}
2020-09-14 09:51:21 +00:00
Taylor Brandstetter
3a034e15b4 Split DataChannel into two separate classes for RTP and SCTP.
Done in preparation for some threading changes that would be quite
messy if implemented with the class as-is.

This results in some code duplication, but is preferable to
one class having two completely different modes of operation.

RTP data channels are in the process of being removed anyway,
so the duplicated code won't last forever.

Bug: webrtc:9883
Change-Id: Idfd41a669b56a4bb4819572e4a264a4ffaaba9c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178940
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31691}
2020-07-10 00:03:21 +00:00
Tomas Gunnarsson
2e94de596e Add GetSctpStats to PeerConnectionInternal, remove sctp_data_channels()
This removes code from DataChannelController that exposes
an internal vector of data channels and puts the onus of
returning stats for a data channel, on the data channel
object itself. This will come in handy as we make threading
changes to the data channel object.

Change-Id: Ie164cc5823cd5f9782fc5c9a63aa4c76b8229639
Bug: webrtc:11547, webrtc:11687
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177244
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31533}
2020-06-16 16:36:42 +00:00
Henrik Boström
a0ff50c031 Reland "Improve outbound-rtp statistics for simulcast"
This reverts commit 9a925c9ce33a6ccdd11b545b11ba68e985c2a65d.

Reason for revert: The original CL is updated in PS #2 to
fix the googRtt issue which was that when the legacy sender
stats were put in "aggregated_senders" we forgot to update
rtt_ms the same way that we do it for "senders".

Original change's description:
> Revert "Improve outbound-rtp statistics for simulcast"
>
> This reverts commit da6cda839dac7d9d18eba8d365188fa94831e0b1.
>
> Reason for revert: Breaks googRtt in legacy getStats API
>
> Original change's description:
> > Improve outbound-rtp statistics for simulcast
> >
> > Bug: webrtc:9547
> > Change-Id: Iec4eb976aa11ee743805425bedb77dcea7c2c9be
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168120
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Eldar Rello <elrello@microsoft.com>
> > Cr-Commit-Position: refs/heads/master@{#31097}
>
> TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:9547
> Change-Id: I06673328c2a5293a7eef03b3aaf2ded9d13df1b3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174443
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31165}

TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com

# Not skipping CQ checks because this is a reland.

Bug: webrtc:9547
Change-Id: I723744c496c3c65f95ab6a8940862c8b9f544338
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174480
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31169}
2020-05-05 20:22:19 +00:00
Henrik Boström
9a925c9ce3 Revert "Improve outbound-rtp statistics for simulcast"
This reverts commit da6cda839dac7d9d18eba8d365188fa94831e0b1.

Reason for revert: Breaks googRtt in legacy getStats API

Original change's description:
> Improve outbound-rtp statistics for simulcast
> 
> Bug: webrtc:9547
> Change-Id: Iec4eb976aa11ee743805425bedb77dcea7c2c9be
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168120
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Eldar Rello <elrello@microsoft.com>
> Cr-Commit-Position: refs/heads/master@{#31097}

TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com

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

Bug: webrtc:9547
Change-Id: I06673328c2a5293a7eef03b3aaf2ded9d13df1b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174443
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31165}
2020-05-05 13:38:51 +00:00
Eldar Rello
da6cda839d Improve outbound-rtp statistics for simulcast
Bug: webrtc:9547
Change-Id: Iec4eb976aa11ee743805425bedb77dcea7c2c9be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168120
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Eldar Rello <elrello@microsoft.com>
Cr-Commit-Position: refs/heads/master@{#31097}
2020-04-17 11:28:00 +00:00
Jonas Oreland
08d1806e54 Extend rtc::AdapterType with 2g, 3G, 4G & 5G enum values.
This patch adds new enum values for different types of cellular
connections.

The new costs are currently blocked when sending to remote,
(so that arbitrary network switches does not starts occurring).

The end-game for this series to be able to distinguish between
different type of cellular connections in the ice-layer (e.g when
selecting/switching connections).

BUG: webrtc:11473
Change-Id: I587ac8fdff4f6cdd0f8905f327232f58818db4f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172582
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30970}
2020-04-02 07:48:36 +00:00
Niels Möller
ac0a4cbbd8 Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received"
This is a reland of fbde32e596f06893d6dda13eb7d29f4c251cf08b

The chromium problem should be fixed with
https://chromium-review.googlesource.com/c/chromium/src/+/1862437

Original change's description:
> Fix GetStats bytesSent/Received, wireup headerBytesSent/Received
>
> Changes the standard GetStats, legacy GetStats unchanged.
>
> Bug: webrtc:10525
> Change-Id: Ie10fe8079f1d8b4cc6bbe513f6a2fc91477b5441
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156084
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29462}

Tbr: kwiberg@webrtc.org
Bug: webrtc:10525
Change-Id: I3b61f9535aa3f1fca2ed84f068233803d4ec9fe2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157045
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29485}
2019-10-15 10:43:59 +00:00
Mirko Bonadei
ef0627fb50 Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received"
This reverts commit fbde32e596f06893d6dda13eb7d29f4c251cf08b.

Reason for revert: It seems to break WebRTC FYI tests in Chromium.

https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Linux%20Tester/4763

Original change's description:
> Fix GetStats bytesSent/Received, wireup headerBytesSent/Received
> 
> Changes the standard GetStats, legacy GetStats unchanged.
> 
> Bug: webrtc:10525
> Change-Id: Ie10fe8079f1d8b4cc6bbe513f6a2fc91477b5441
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156084
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29462}

TBR=kwiberg@webrtc.org,hbos@webrtc.org,nisse@webrtc.org,hta@webrtc.org

Change-Id: I6a983ea4d5ff38e49f096a8ff5cd9b426768f955
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10525
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157043
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29478}
2019-10-15 08:55:06 +00:00
Niels Möller
fbde32e596 Fix GetStats bytesSent/Received, wireup headerBytesSent/Received
Changes the standard GetStats, legacy GetStats unchanged.

Bug: webrtc:10525
Change-Id: Ie10fe8079f1d8b4cc6bbe513f6a2fc91477b5441
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156084
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29462}
2019-10-14 13:07:13 +00:00
Mirko Bonadei
317a1f09ed Use std::make_unique instead of absl::make_unique.
WebRTC is now using C++14 so there is no need to use the Abseil version
of std::make_unique.

This CL has been created with the following steps:

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

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

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

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

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

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

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

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

# Repead the gn_check_autofix step for other platforms

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

Bug: webrtc:10945
Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 15:47:29 +00:00
Jonas Oreland
149dc72dfa Add support for RTCTransportStats.selectedCandidatePairChanges
This patch adds accounting and reporting needed for
newly added RTCTransportStats.selectedCandidatePairChanges,
https://w3c.github.io/webrtc-stats/#dom-rtctransportstats-selectedcandidatepairchanges

a) P2PTransportChannel counts everytime selected_connection_
is modified and reports this counter in the GetStats()-call.
b) RTCStatsCollector puts the counter into the standardized
stats object.

Bug: webrtc:10900
Change-Id: Ibaeca18706b8edcbcb44b0c6f2754854bcb545ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149830
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28987}
2019-08-28 13:22:08 +00:00
Alex Narest
bbeb10925e Reporting audio device underrun counter
Bug: webrtc:10884
Change-Id: I35636fcbc1e2a19a89242379cdff6ec5c12fd21a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149200
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Alex Narest <alexnarest@google.com>
Cr-Commit-Position: refs/heads/master@{#28874}
2019-08-16 11:49:55 +00:00
Alex Narest
5b5d97c938 Reland of "Reporting of decoding_codec_plc events""
This is a reland of 0a88ea050cda58de81d624cf2764d46929447ed5.

The new stat will not be reported unless it is GT 0.

Reporting of decoding_codec_plc events

Bug: webrtc:10838
Change-Id: Ic8585b4eeae9a2643374f15bc2578d1141e59683
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148448
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Alex Narest <alexnarest@google.com>
Cr-Commit-Position: refs/heads/master@{#28797}
2019-08-07 18:41:46 +00:00
Steve Anton
032ea9cbc7 Remove "Missing track ID" StatsCollector log message
Bug: None
Change-Id: I94a70d5cbe4238220bf0939ba5807cf51195fe7f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147661
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28727}
2019-07-31 19:33:16 +00:00
Mirko Bonadei
bedb7a8aea Revert "Reporting of decoding_codec_plc events"
This reverts commit 0a88ea050cda58de81d624cf2764d46929447ed5.

Reason for revert: This CL breaks Chromium's FYI bots (example: https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Linux%20Tester/4033).

Original change's description:
> Reporting of decoding_codec_plc events
> 
> Change-Id: Id71b37244bc98bffaf25131a519127b3d2b86a8f
> 
> Bug: webrtc:10838
> Change-Id: Id71b37244bc98bffaf25131a519127b3d2b86a8f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147263
> Commit-Queue: Alex Narest <alexnarest@google.com>
> Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28700}

TBR=mflodman@webrtc.org,alexnarest@google.com

Change-Id: I5e5dd29ee375ba422f79932d4b8c3fd028a53db4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10838
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147269
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28707}
2019-07-30 14:39:09 +00:00
Alex Narest
0a88ea050c Reporting of decoding_codec_plc events
Change-Id: Id71b37244bc98bffaf25131a519127b3d2b86a8f

Bug: webrtc:10838
Change-Id: Id71b37244bc98bffaf25131a519127b3d2b86a8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147263
Commit-Queue: Alex Narest <alexnarest@google.com>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28700}
2019-07-29 16:40:23 +00:00
Henrik Andreassson
fc02a793c2 Revert "Piping audio interruption metrics to API layer"
This reverts commit 299c4e68461f1c4428b2a919913b161115025dff.

Reason for revert: https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Win10%20Tester/2753

../../chrome/browser/media/webrtc/webrtc_browsertest_base.cc(539): error: Expected equality of these values:
  "ok-got-stats"
  ExecuteJavascript("verifyLegacyStatsGenerated()", tab)
    Which is: "Test failed: Error: \"googInterruptionCount\" is not a whitelisted stat. Exposing new metrics in the legacy getStats() API is not allowed. Please follow the standardization process: https://docs.google.com/document/d/1q1CJVUqJ6YW9NNRc0tENkLNny8AHrKZfqjy3SL89zjc/edit?usp=sharing\n    at failTest (http://127.0.0.1:50650/webrtc/test_functions.js:46:15)\n    at http://127.0.0.1:50650/webrtc/peerconnection.js:481:19"
With diff:
@@ -1,1 +1,3 @@
-ok-got-stats
+Test failed: Error: \"googInterruptionCount\" is not a whitelisted stat. Exposing new metrics in the legacy getStats() API is not allowed. Please follow the standardization process: https://docs.google.com/document/d/1q1CJVUqJ6YW9NNRc0tENkLNny8AHrKZfqjy3SL89zjc/edit?usp=sharing
+    at failTest (http://127.0.0.1:50650/webrtc/test_functions.js:46:15)
+    at http://127.0.0.1:50650/webrtc/peerconnection.js:481:19

Original change's description:
> 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}

TBR=henrik.lundin@webrtc.org,kwiberg@webrtc.org,ivoc@webrtc.org

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

Bug: webrtc:10549
Change-Id: I345306ba9758c0a3b1597724fa860d3e3fdb8995
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134464
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27802}
2019-04-29 11:23:16 +00:00
Henrik Lundin
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
Steve Anton
efe4c92d54 Use RtpSender/RtpReceiver track ID for legacy GetStats
Previously, legacy GetStats would look up the track ID by querying the
local/remote SDP by SSRC. This doesn't work with Unified Plan since the
RtpSender/RtpReceiver track IDs may not correspond to the track ID
stored in the SDP.

This CL changes legacy GetStats to pull the track ID directly from the
RtpSenders and RtpReceivers as it generates the stats. This has a few
additional benefits:
1) Unsignaled receive SSRC stats should now get correctly matched to
   the unsigneled RtpReceiver track ID for both Plan B and Unified
   Plan.
2) Removes a couple methods on PeerConnection that were only used by
   the legacy StatsCollector.
3) Keeps the SSRC -> track ID mapping more localized which should make
   the code easier to understand.

Bug: chromium:943493
Change-Id: I43ecde8c3a3d1c5f9c749ba6c8dfb11e8c4950fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129782
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27324}
2019-03-27 18:14:00 +00:00
Philipp Hancke
fe0b499634 legacy stats: update timestamp on localcandidate/remotecandidate
updates the timestamp on the local and remote candidate stats for consistency
with other places. This also makes the graphs on chrome://webrtc-internals
work (even though most values don't update so showing graphs is not meaningful)

BUG=chromium:937833

Change-Id: I3267dd7a5f5a887dcd0756137077b8f02c201905
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128765
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27248}
2019-03-22 17:26:20 +00:00
Mirko Bonadei
739baf097b [clang-tidy] Apply performance-for-range-copy fixes.
This CL applies clang-tidy's performance-for-range-copy [1] on the
WebRTC codebase.

All changes in this CL are automatically generated by both clang-tidy
and 'git cl format'.

[1] - https://clang.llvm.org/extra/clang-tidy/checks/performance-for-range-copy.html

Bug: webrtc:10215
Change-Id: I7c83290b8866d76129bbec4e24e6701f5014102e
Reviewed-on: https://webrtc-review.googlesource.com/c/120043
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26420}
2019-01-28 09:53:50 +00:00
Steve Anton
10542f21c8 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
Mechanically generated by running this command:

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

Then manually updating:

tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 17:11:39 +00:00
Steve Anton
1c05765831 (3) Rename files to snake_case: move the files
Mechanically generated with this command:

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

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I8b05b6eab9b9d18b29c2199bbea239e9add1e690
Reviewed-on: https://webrtc-review.googlesource.com/c/115481
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26225}
2019-01-11 17:05:20 +00:00