Commit Graph

217 Commits

Author SHA1 Message Date
5eb43b4777 Prefix HAVE_SCTP macro with WEBRTC_.
Generated automatically with:

  git grep -l "\bHAVE_SCTP\b" | xargs \
    sed -i '' 's/HAVE_SCTP/WEBRTC_HAVE_SCTP/g'

Bug: webrtc:11142
Change-Id: I30e16a40ca7a7e388940191df22b705265b42cb4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202251
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33042}
2021-01-20 10:51:07 +00:00
5932fe1392 RtpTransceiverInterface: introduce HeaderExtensionsNegotiated.
This changes adds exposure of a new transceiver method for
accessing header extensions that have been negotiated, following
spec details in https://w3c.github.io/webrtc-extensions/#rtcrtptransceiver-interface.

The change contains unit tests testing the functionality.

Note: support for signalling directionality of header extensions
in the SDP isn't implemented yet.

https://chromestatus.com/feature/5680189201711104.
Intent to prototype: https://groups.google.com/a/chromium.org/g/blink-dev/c/65YdUi02yZk

Bug: chromium:1051821
Change-Id: If963beed37e96eed2dff3a2822db4e30caaea4a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198126
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32860}
2020-12-17 23:43:42 +00:00
165c618bb9 Reland: Use CRYPTO_BUFFER APIs instead of X509 when building with BoringSSL.
Using CRYPTO_BUFFERs instead of legacy X509 objects offers memory and
security gains, and will provide binary size improvements as well once
the default list of built-in certificates can be removed; the code
dealing with them still depends on the X509 API.

Implemented by splitting openssl_identity and openssl_certificate
into BoringSSL and vanilla OpenSSL implementations.

No-Try: True
Bug: webrtc:11410
Change-Id: I86ddb361b94ad85b15ebb8743490de83632ca53f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196941
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32818}
2020-12-11 09:58:40 +00:00
c1ad1ff178 Do all BaseChannel operations within a single Thread::Invoke.
Instead of doing a separate Invoke for each channel, this CL first
gathers a list of operations to be performed on the signaling thread,
then does a single Invoke on the worker thread (and nested Invoke
on the network thread) to update all channels at once.

This includes the methods:
* Enable
* SetLocalContent/SetRemoteContent
* RegisterRtpDemuxerSink
* UpdateRtpHeaderExtensionMap

Also, removed the need for a network thread Invoke in
IsReadyToSendMedia_w by moving ownership of was_ever_writable_ to the
worker thread.

Bug: webrtc:12266
Change-Id: I31e61fe0758aeb053b09db84f234deb58dfb3d05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194181
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32817}
2020-12-11 03:25:43 +00:00
7e6290d1d2 Revert "Use CRYPTO_BUFFER APIs instead of X509 when building with BoringSSL."
This reverts commit 72f638a9a279e7abb5534fa66a0ade2cf18ec1a7.

Reason for revert: downstream build failures

Original change's description:
> Use CRYPTO_BUFFER APIs instead of X509 when building with BoringSSL.
>
> Using CRYPTO_BUFFERs instead of legacy X509 objects offers memory and
> security gains, and will provide binary size improvements as well once
> the default list of built-in certificates can be removed; the code
> dealing with them still depends on the X509 API.
>
> Implemented by splitting openssl_identity and openssl_certificate
> into BoringSSL and vanilla OpenSSL implementations.
>
> Bug: webrtc:11410
> Change-Id: Idc043462faac5e4ab1b75bedab2057197f80aba6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174120
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: David Benjamin <davidben@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Taylor <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32811}

TBR=deadbeef@webrtc.org,mbonadei@webrtc.org,davidben@webrtc.org,hta@webrtc.org

Change-Id: Ib5e55cb5798a2f3d25a4460f5311d2e650d3fa82
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11410
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196742
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32812}
2020-12-10 07:55:40 +00:00
72f638a9a2 Use CRYPTO_BUFFER APIs instead of X509 when building with BoringSSL.
Using CRYPTO_BUFFERs instead of legacy X509 objects offers memory and
security gains, and will provide binary size improvements as well once
the default list of built-in certificates can be removed; the code
dealing with them still depends on the X509 API.

Implemented by splitting openssl_identity and openssl_certificate
into BoringSSL and vanilla OpenSSL implementations.

Bug: webrtc:11410
Change-Id: Idc043462faac5e4ab1b75bedab2057197f80aba6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174120
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: David Benjamin <davidben@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32811}
2020-12-10 02:19:47 +00:00
d3ef499418 Enable payload type based demuxing with multiple tracks when applicable.
This fixes regressions caused by:
https://webrtc-review.googlesource.com/c/src/+/183120

... which disabled payload type demuxing when multiple video tracks are
present, to avoid one channel creating a default track intended for
another channel.

However, this isn't an issue when not bundling, as each track will be
delivered on separate transport.

And it's also not an issue when each track uses a distinct set of
payload types (e.g., VP8 is mapped to PT 96 in one m= section, and PT 97
in another).

This CL addresses both of those cases; PT demuxing is only disabled
when two bundled m= sections have overlapping payload types.

Bug: chromium:1139052, webrtc:12029
Change-Id: Ied844bffac2a5fac29147c11b56a5f83a95ecb36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187560
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32419}
2020-10-16 03:09:22 +00:00
1ee3325051 When stopping a transceiver, end the receiver's track.
Bug: webrtc:11840
Change-Id: Ib8171c58fcb13c33ab03398eb3021c07e55ff008
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185181
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32188}
2020-09-24 21:37:32 +00:00
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
abdb470d00 Make MessageHandler cleanup optional.
As documented in webrtc:11908 this cleanup is fairly invasive and
when a part of a frequently executed code path, can be quite costly
in terms of performance overhead. This is currently the case with
synchronous calls between threads (Thread) as well with our proxy
api classes.

With this CL, all code in WebRTC should now either be using MessageHandlerAutoCleanup
or calling MessageHandler(false) explicitly.

Next steps will be to update external code to either depend on the
AutoCleanup variant, or call MessageHandler(false).

Changing the proxy classes to use TaskQueue set of concepts instead of
MessageHandler. This avoids the perf overhead related to the cleanup
above as well as incompatibility with the thread policy checks in
Thread that some current external users of the proxies would otherwise
run into (if we were to use Thread::Send() for synchronous call).

Following this we'll move the cleanup step into the AutoCleanup class
and an RTC_DCHECK that all calls to the MessageHandler are setting
the flag to false, before eventually removing the flag and make
MessageHandler pure virtual.

Bug: webrtc:11908
Change-Id: Idf4ff9bcc8438cb8c583777e282005e0bc511c8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183442
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32049}
2020-09-07 12:57:15 +00:00
c03a187391 Default streams: don't block media even if on different transceiver.
This fixes some edge cases where early media could cause default
stream that block the actual signaled media from beind delivered.

Bug: webrtc:11477
Change-Id: I8b26df63a690861bd19f083102d1395e882f8733
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183120
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32030}
2020-09-02 22:28:55 +00:00
2bca008914 Reland "Make cricket::SctpTransportInternalFactory injectable through PeerConnectionFactory Deps"
This is to allow testing without using the singleton sctp library.
cricket::SctpTransportInternalFactory is renamed to webrtc::SctpTransportFactoryInterface and moved to the API folder to follow the API structure.
Tests can use test/pc/sctp/fake_sctp_transport.h to inject a faked data channel implementation.

patch 1 contain the original cl.
patch 2 modifications

Bug: none
Change-Id: Ic088da3eb7d9aada79e6d601dbf2d1aa2be777f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182840
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32024}
2020-09-01 12:17:00 +00:00
1f580a97e5 Revert "Make cricket::SctpTransportInternalFactory injectable through PeerConnectionFactory Deps"
This reverts commit 4c0a381137c04fd80830af8a041e25e3428dd33f.

Reason for revert: Breaks downstream test

Original change's description:
> Make cricket::SctpTransportInternalFactory injectable through PeerConnectionFactory Deps
> 
> This is to allow testing without using the singleton sctp library. 
> cricket::SctpTransportInternalFactory is renamed to webrtc::SctpTransportFactoryInterface and moved to the API folder to follow the API structure.
> Tests can use test/pc/sctp/fake_sctp_transport.h to inject a faked data channel implementation.
> 
> Bug: none
> Change-Id: I482241269463595062548870750d33f31238c6b1
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182082
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Taylor <deadbeef@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32007}

TBR=deadbeef@webrtc.org,mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org

Change-Id: I46d5ba89fe723caccd065b0ac41d77ed45373838
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182802
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32008}
2020-08-27 13:59:57 +00:00
4c0a381137 Make cricket::SctpTransportInternalFactory injectable through PeerConnectionFactory Deps
This is to allow testing without using the singleton sctp library. 
cricket::SctpTransportInternalFactory is renamed to webrtc::SctpTransportFactoryInterface and moved to the API folder to follow the API structure.
Tests can use test/pc/sctp/fake_sctp_transport.h to inject a faked data channel implementation.

Bug: none
Change-Id: I482241269463595062548870750d33f31238c6b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182082
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32007}
2020-08-27 13:19:14 +00:00
e574a31c50 [Perfect Negotiation] Fire onnegotiationneeded when chain is empty.
This CL generates "negotiationneeded" events if negotiation is needed
when the Operations Chain becomes empty. This is only implemented in
Unified Plan to avoid Plan B regressions (the event is pretty useless
in Plan B as it fires repeatedly).

In order to implement the spec-compliant behavior of only firing the
event when the chain is empty, this CL introduces
PeerConnectionObserver::OnNegotiationNeededEvent() and
PeerConnectionInterface::ShouldFireNegotiationNeededEvent() to allow
validating the event before firing it. This is needed because the event
must not be fired until a task has been posted and subsequently chained
operations could invalidate it in the meantime.

Test coverage is added for both legacy and modern "negotiationneeded"
events.

Bug: chromium:1060083
Change-Id: I1dbaa8f6ddb1c6e7c8abd8da3b92efcb64060383
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180620
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31989}
2020-08-25 09:56:39 +00:00
6060df5948 Reland "Implement transceiver.stop()"
This is a reland of 11dc6571cb4ff3e71dee1557dfff8d9076e108d3

One fix that makes Web Platform Tests pass in debug mode is applied.

Original change's description:
> Implement transceiver.stop()
>
> This adds RtpTransceiver.StopStandard(), which behaves according to
> the specification at
> https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop
>
> It modifies RTCPeerConnection.getTransceivers() to return only
> transceivers that have not been stopped.
>
> Rebase of armax' https://webrtc-review.googlesource.com/c/src/+/172762
>
> Bug: chromium:980879
> Change-Id: I7d383ee874ccc0a006fdcf280496b5d4235425ce
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180580
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31893}

Bug: chromium:980879
Change-Id: Ide31d929ac5ea118d83fdf6a35a592af23f7dfa7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181263
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31907}
2020-08-11 10:46:23 +00:00
a88c9776de Revert "Implement transceiver.stop()"
This reverts commit 11dc6571cb4ff3e71dee1557dfff8d9076e108d3.

Reason for revert: Breaks Chromium WPT tests

Original change's description:
> Implement transceiver.stop()
> 
> This adds RtpTransceiver.StopStandard(), which behaves according to
> the specification at
> https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop
> 
> It modifies RTCPeerConnection.getTransceivers() to return only
> transceivers that have not been stopped.
> 
> Rebase of armax' https://webrtc-review.googlesource.com/c/src/+/172762
> 
> Bug: chromium:980879
> Change-Id: I7d383ee874ccc0a006fdcf280496b5d4235425ce
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180580
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31893}

TBR=sakal@webrtc.org,kthelgason@webrtc.org,hta@webrtc.org,guidou@webrtc.org,marinaciocea@webrtc.org

Change-Id: Ibdc24f7d41e481293ca74ba6d1572de64f7e4654
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:980879
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181262
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31897}
2020-08-10 18:06:30 +00:00
11dc6571cb Implement transceiver.stop()
This adds RtpTransceiver.StopStandard(), which behaves according to
the specification at
https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop

It modifies RTCPeerConnection.getTransceivers() to return only
transceivers that have not been stopped.

Rebase of armax' https://webrtc-review.googlesource.com/c/src/+/172762

Bug: chromium:980879
Change-Id: I7d383ee874ccc0a006fdcf280496b5d4235425ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180580
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31893}
2020-08-10 13:29:15 +00:00
831ae4ef65 Reland "[Perfect Negotiation] Implement non-racy version of SetLocalDescription."
This is a reland of d4089cae47334a4228b69d6bb23f2e49ebb7496e
with the following fix:

Invoke MaybeStartGathering as the last step of DoSetLocalDescription.
This ensures that candidates and onicegatheringstatechange does not
happen before SLD is resolved. This is important for passing
external/wpt/webrtc/RTCPeerConnection-iceGatheringState.html.

Original change's description:
> [Perfect Negotiation] Implement non-racy version of SetLocalDescription.
>
> BACKGROUND
>
> When SLD is invoked with SetSessionDescriptionObserver, the observer is
> called by posting a message back to the execution thread, delaying the
> call. This delay is "artificial" - it's not necessary; the operation is
> already complete. It's a post from the signaling thread to the signaling
> thread. The rationale for the post was to avoid the observer making
> recursive calls back into the PeerConnection. The problem with this is
> that by the time the observer is called, the PeerConnection could
> already have executed other operations and modified its states.
>
> This causes the referenced bug: one can have a race where SLD is
> resolved "too late" (after a pending SRD is executed) and the signaling
> state observed when SLD resolves doesn't make sense.
>
> When implementing Unified Plan, we fixed similar issues for SRD by
> adding a version that takes SetRemoteDescriptionObserverInterface as
> argument instead of SetSessionDescriptionObserver. The new version did
> not have the delay. The old version had to be kept around not to break
> downstream projects that had dependencies both on he delay and on
> allowing the PC to be destroyed midst-operation without informing its
> observers.
>
> THIS CL
>
> This does the old SRD fix for SLD as well: A new observer interface is
> added, SetLocalDescriptionObserverInterface, and
> PeerConnection::SetLocalDescription() is overloaded. If you call it with
> the old observer, you get the delay, but if you call it with the new
> observer, you don't get a delay.
>
> - SetLocalDescriptionObserverInterface is added.
> - SetLocalDescription is overloaded.
> - The adapter for SetSessionDescriptionObserver that causes the delay
>   previously only used for SRD is updated to handle both SLD and SRD.
> - FakeSetLocalDescriptionObserver is added and
>   MockSetRemoteDescriptionObserver is renamed "Fake...".
>
> Bug: chromium:1071733
> Change-Id: I920368e648bede481058ac22f5b8794752a220b3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179100
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31798}

TBR=hta@webrtc.org

Bug: chromium:1071733
Change-Id: Ic6e8d96afa1c19604762f373716c08dbfa9d178c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180481
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31804}
2020-07-29 11:27:43 +00:00
4c9c75a2a6 Revert "[Perfect Negotiation] Implement non-racy version of SetLocalDescription."
This reverts commit d4089cae47334a4228b69d6bb23f2e49ebb7496e.

Reason for revert: Breaks chromium WPT that is timing sensitive to onicegatheringstatechanges.
This CL accidentally moved the MaybeStartGatheringIceCandidates to after completing the SLD call. The fix is to move it back. I'll do that in a re-land.

Original change's description:
> [Perfect Negotiation] Implement non-racy version of SetLocalDescription.
> 
> BACKGROUND
> 
> When SLD is invoked with SetSessionDescriptionObserver, the observer is
> called by posting a message back to the execution thread, delaying the
> call. This delay is "artificial" - it's not necessary; the operation is
> already complete. It's a post from the signaling thread to the signaling
> thread. The rationale for the post was to avoid the observer making
> recursive calls back into the PeerConnection. The problem with this is
> that by the time the observer is called, the PeerConnection could
> already have executed other operations and modified its states.
> 
> This causes the referenced bug: one can have a race where SLD is
> resolved "too late" (after a pending SRD is executed) and the signaling
> state observed when SLD resolves doesn't make sense.
> 
> When implementing Unified Plan, we fixed similar issues for SRD by
> adding a version that takes SetRemoteDescriptionObserverInterface as
> argument instead of SetSessionDescriptionObserver. The new version did
> not have the delay. The old version had to be kept around not to break
> downstream projects that had dependencies both on he delay and on
> allowing the PC to be destroyed midst-operation without informing its
> observers.
> 
> THIS CL
> 
> This does the old SRD fix for SLD as well: A new observer interface is
> added, SetLocalDescriptionObserverInterface, and
> PeerConnection::SetLocalDescription() is overloaded. If you call it with
> the old observer, you get the delay, but if you call it with the new
> observer, you don't get a delay.
> 
> - SetLocalDescriptionObserverInterface is added.
> - SetLocalDescription is overloaded.
> - The adapter for SetSessionDescriptionObserver that causes the delay
>   previously only used for SRD is updated to handle both SLD and SRD.
> - FakeSetLocalDescriptionObserver is added and
>   MockSetRemoteDescriptionObserver is renamed "Fake...".
> 
> Bug: chromium:1071733
> Change-Id: I920368e648bede481058ac22f5b8794752a220b3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179100
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31798}

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

Change-Id: Ie1e1ecc49f3b1d7a7e230db6d36decbc4cbe8c86
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1071733
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180480
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31802}
2020-07-29 09:46:56 +00:00
d4089cae47 [Perfect Negotiation] Implement non-racy version of SetLocalDescription.
BACKGROUND

When SLD is invoked with SetSessionDescriptionObserver, the observer is
called by posting a message back to the execution thread, delaying the
call. This delay is "artificial" - it's not necessary; the operation is
already complete. It's a post from the signaling thread to the signaling
thread. The rationale for the post was to avoid the observer making
recursive calls back into the PeerConnection. The problem with this is
that by the time the observer is called, the PeerConnection could
already have executed other operations and modified its states.

This causes the referenced bug: one can have a race where SLD is
resolved "too late" (after a pending SRD is executed) and the signaling
state observed when SLD resolves doesn't make sense.

When implementing Unified Plan, we fixed similar issues for SRD by
adding a version that takes SetRemoteDescriptionObserverInterface as
argument instead of SetSessionDescriptionObserver. The new version did
not have the delay. The old version had to be kept around not to break
downstream projects that had dependencies both on he delay and on
allowing the PC to be destroyed midst-operation without informing its
observers.

THIS CL

This does the old SRD fix for SLD as well: A new observer interface is
added, SetLocalDescriptionObserverInterface, and
PeerConnection::SetLocalDescription() is overloaded. If you call it with
the old observer, you get the delay, but if you call it with the new
observer, you don't get a delay.

- SetLocalDescriptionObserverInterface is added.
- SetLocalDescription is overloaded.
- The adapter for SetSessionDescriptionObserver that causes the delay
  previously only used for SRD is updated to handle both SLD and SRD.
- FakeSetLocalDescriptionObserver is added and
  MockSetRemoteDescriptionObserver is renamed "Fake...".

Bug: chromium:1071733
Change-Id: I920368e648bede481058ac22f5b8794752a220b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179100
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31798}
2020-07-28 10:05:57 +00:00
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
6fcd0f8031 Migrate pc/ to webrtc::Mutex.
Bug: webrtc:11567
Change-Id: I1adc22d2998966958750138e66108cf39a8c3d57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178840
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31654}
2020-07-07 18:25:09 +00:00
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
7d3cfbf90d Inject signaling and network threads to DataChannel.
Add a few DCHECKs and comments about upcoming work.

Bug: webrtc:11547
Change-Id: I2d42f48cb93f31e70cf9fe4b3b62241c38bc9d8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177106
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31530}
2020-06-16 10:22:19 +00:00
4c1e7cc19b [Adaptation] Add ability to inject resources on the PeerConnection.
This unblocks injecting platform-specific resources, such as power
usage signals in Chrome.

This CL adds AddAdaptationResource to PeerConnectionInterface and
integration tests verifying that if an injected resource is overusing,
resolution will soon be reduced.

To aid testing, some testing-only classes have been updated.

Bug: webrtc:11525
Change-Id: I820099e79f18d910fd641ee1412ad064b99ebce9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177003
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31505}
2020-06-11 14:17:01 +00:00
0ca13d97d2 Introduce BYPASS_PROXY_CONSTMETHOD0.
This allows const getters that query const state to be called without
marshalling calls between threads. This must not be used to
return pointers/references etc.

I'm starting by using this macro with the data channel which has a
few of these getters, as well as changing things a bit to make more
parts of the implementation, const.

Change-Id: I6ec7a3774cd8f7be2ef122fb7c7fc5919afee600
Bug: webrtc:11547
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176846
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31489}
2020-06-10 13:52:36 +00:00
2af35ab984 FakeAudioCaptureModule: remove lock recursions.
This change removes lock recursions and adds thread annotations.

The module had incorrect locking WRT the callback critical section:

ProcessFrameP: locks crit_
ReceiveFrameP: locks crit_callback_
-------------
SendFrameP: locks crit_callback_
MicrophoneVolume: locks crit_

Lock crit_callback_ was rolled in under crit_ instead.

Bug: webrtc:11567
Change-Id: I974fe91d44de0ddf1a1287fe91db9dfe63a61af9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175662
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31313}
2020-05-18 18:01:58 +00:00
3a35312b64 In pc/ replace mock macros with unified MOCK_METHOD macro
Bug: webrtc:11564
Change-Id: I09b28654b7b71a77224e7cf72fdf6a1e4823e67a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175137
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31310}
2020-05-18 17:06:25 +00:00
61f74d91f8 Reland "Expose can_trickle_ice_candidates on PeerConnection"
This reverts commit cb8c40138ca170f841bc45fa6771cdfc4b966e5f.

Reason for revert: Added missing default.

Original change's description:
> Revert "Expose can_trickle_ice_candidates on PeerConnection"
>
> This reverts commit c6a65c8866487c6adc0a7bb472d3bad9389501f9.
>
> Reason for revert: Breaks downstream due to missing default
>
> Original change's description:
> > Expose can_trickle_ice_candidates on PeerConnection
> >
> > Bug: chromium:708484
> > Change-Id: I9a40e75066341f0d9f965bd3718bfcb3f0459533
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169450
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Taylor <deadbeef@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30653}
>
> TBR=deadbeef@webrtc.org,hta@webrtc.org
>
> Change-Id: Iaa5b977c4237715a8a5127cf167cf6512a3f7059
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:708484
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169540
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30655}

TBR=deadbeef@webrtc.org,hta@webrtc.org

Change-Id: I608da7781f158b4b02dd226d4dcd5615c4935fa8
Bug: chromium:708484
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169541
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30656}
2020-03-02 11:35:53 +00:00
cb8c40138c Revert "Expose can_trickle_ice_candidates on PeerConnection"
This reverts commit c6a65c8866487c6adc0a7bb472d3bad9389501f9.

Reason for revert: Breaks downstream due to missing default

Original change's description:
> Expose can_trickle_ice_candidates on PeerConnection
> 
> Bug: chromium:708484
> Change-Id: I9a40e75066341f0d9f965bd3718bfcb3f0459533
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169450
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Taylor <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30653}

TBR=deadbeef@webrtc.org,hta@webrtc.org

Change-Id: Iaa5b977c4237715a8a5127cf167cf6512a3f7059
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:708484
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169540
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30655}
2020-03-02 10:14:14 +00:00
c6a65c8866 Expose can_trickle_ice_candidates on PeerConnection
Bug: chromium:708484
Change-Id: I9a40e75066341f0d9f965bd3718bfcb3f0459533
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169450
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30653}
2020-03-02 05:19:16 +00:00
0c626afcf3 Use newer version of TimeDelta and TimeStamp factories in webrtc
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g"
git cl format

Bug: None
Change-Id: I87469d2e4a38369654da839ab7c838215a7911e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168402
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30491}
2020-02-10 12:21:17 +00:00
1cb929fb9e Cleanup: remove unused sctp_content_name
This accessor seems to be unused, and has a name that we don't
want to support ("content_name").

Bug: none
Change-Id: I2f332176429dd8e1895f821d30e4beaaa4650ec2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168195
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30460}
2020-02-05 19:49:28 +00:00
33f9d2b383 Migrate WebRTC on FrameGeneratorInterface and remove FrameGenerator class
Bug: webrtc:10138
Change-Id: If85290581a72f81cf60181de7a7134cc9db7716e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161327
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30033}
2019-12-07 00:54:26 +00:00
934afc6ba1 Deprecate RtpReceiver's SetParameters method
This removes the SetParameters method from AudioRtpReceiver and Video
RtpReceiver, which is currently not used and is not part of the
specifications.


Bug: webrtc:11111
Change-Id: I6f67773bfef2d4b51e9ab670bde17b5fbf5f94c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159307
Reviewed-by: Patrik Höglund <phoglund@google.com>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Daniela Jovanoska Petrenko <denicija@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Saurav Das <dinosaurav@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29995}
2019-12-03 19:50:42 +00:00
4e19670d3a [PeerConnection] Implement parameterless SetLocalDescription().
For background, motivation, requirements and implementation notes, see
https://docs.google.com/document/d/1XLwNN2kUIGGTwz9LQ0NwJNkcybi9oKnynUEZB1jGA14/edit?usp=sharing

The parameterless SetLocalDescription() will implicitly create an
offer or answer to be set by chaining create offer or answer with
setting the session description, as per spec:
https://w3c.github.io/webrtc-pc/#dom-peerconnection-setlocaldescription

Bug: chromium:980885
Change-Id: Ia430160869df18fd47b756b9adf9e7e23ba8e969
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157444
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29653}
2019-10-30 10:24:44 +00:00
7b04a91f4a Delete almost all default methods on PeerConnectionInterface
Keeping default implementations only for methods involved in
ongoing transitions.

Intended to catch inconsistencies between the interface and the
PeerConnectionProxy class, at compile time.

Bug: webrtc:10716
Change-Id: I4cb126c353855f7288ba09273fa6f87aaa0f32eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140860
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29224}
2019-09-18 16:27:44 +00:00
317a1f09ed Use std::make_unique instead of absl::make_unique.
WebRTC is now using C++14 so there is no need to use the Abseil version
of std::make_unique.

This CL has been created with the following steps:

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

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

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

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

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

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

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

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

# Repead the gn_check_autofix step for other platforms

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

Bug: webrtc:10945
Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 15:47:29 +00:00
7262fc29a0 Refactor Rtp Receivers to accept SSRC 0.
Changes Rtp Receivers to use a null value of ssrc to mean a default
receive stream.

Bug: webrtc:8694
Change-Id: I835199345f7add993b9078c8b0e7988d5cdd6646
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152425
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Saurav Das <dinosaurav@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29201}
2019-09-16 21:29:58 +00:00
e78fd80cc2 New class DummyPeerConnection
Intended as a utility base class for tests, to make it easier to
delete default implementations of PeerConnectionInterface methods.

Bug: webrtc:10716
Change-Id: Ie125747ad88d209c4797cc13253aef61275ed7b5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152820
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29184}
2019-09-13 13:23:34 +00:00
340e0c5f7a Delete old version of PeerConnection::SetConfiguration
Followup to https://webrtc-review.googlesource.com/c/src/+/149166

Bug: None
Change-Id: I7b33ee241e3259b8d43f924a38a1e79ec2cd697f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149812
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29057}
2019-09-04 08:23:18 +00:00
2579f0c584 RTCError as return type for PeerConnectionInterface::SetConfiguration
Bug: None
Change-Id: I6dd7378ceac617e29945d72906cb8e2e0bd49538
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149166
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28910}
2019-08-20 06:52:05 +00:00
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
79b6980020 [PeerConnection] Implement restartIce().
This is part of "Perfect Negotiation" (https://crbug.com/980872).
Spec PR here (merged): https://github.com/w3c/webrtc-pc/pull/2169
Spec: https://w3c.github.io/webrtc-pc/#dfn-localufragstoreplace

The restartIce() makes the next createOffer() generate new ICE
credentials, as if {iceRestart:true} was passed in as options. It also
causes negotiationneeded. This is better than manually restarting ICE
because it survives rollbacks (when that is implemented) and
restartIce() can be called regardless of current signalingState.

Bug: chromium:980881
Change-Id: I8e70bec31ce9d4d6a303bd35e91b2dcc28fcad60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144941
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28596}
2019-07-18 10:00:10 +00:00
0182a0300f Reland "Remove the injectable bitrate allocation strategy API."
This is a reland of 80cb3f6db622442b6360e67851e8903aa0d06d03

Original change's description:
> Remove the injectable bitrate allocation strategy API.
>
> This removes PeerConnectionInterface::SetBitrateAllocationStrategy()
> plus a ton of now-dead code.
>
> Bug: webrtc:10556
> Change-Id: Icfae3bdd011588552934d9db4df16000847db7c3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133169
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28523}

TBR=kwiberg@webrtc.org

Bug: webrtc:10556
Change-Id: Ic17a7a7cc447292306876ee9582ad62fd2499765
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145900
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28585}
2019-07-17 10:20:45 +00:00
e95b57cdfc Revert "Remove the injectable bitrate allocation strategy API."
This reverts commit 80cb3f6db622442b6360e67851e8903aa0d06d03.

Reason for revert: Performance regression on downstream project.

Original change's description:
> Remove the injectable bitrate allocation strategy API.
> 
> This removes PeerConnectionInterface::SetBitrateAllocationStrategy()
> plus a ton of now-dead code.
> 
> Bug: webrtc:10556
> Change-Id: Icfae3bdd011588552934d9db4df16000847db7c3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133169
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28523}

TBR=henrika@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,srte@webrtc.org,alexnarest@webrtc.org,jonasolsson@webrtc.org

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

Bug: webrtc:10556
Change-Id: Ife905d661e7b1a227662395c729a9336c62fd2d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145338
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28560}
2019-07-12 15:27:19 +00:00
80cb3f6db6 Remove the injectable bitrate allocation strategy API.
This removes PeerConnectionInterface::SetBitrateAllocationStrategy()
plus a ton of now-dead code.

Bug: webrtc:10556
Change-Id: Icfae3bdd011588552934d9db4df16000847db7c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133169
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28523}
2019-07-10 13:13:25 +00:00
a4d873786f Format almost everything.
This CL was generated by running

git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \
grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \
grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \
grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \
grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \
| xargs clang-format -i ; git cl format

Most of these changes are clang-format grouping and reordering includes
differently.

Bug: webrtc:9340
Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28505}
2019-07-08 13:45:15 +00:00
695cf6ac42 Delete deprecated StartRtcEventLog override with PlatformFile
Bug: webrtc:6463
Change-Id: I57c2372a232d72b054d8e3e4f423e11b3fb22430
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134460
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28131}
2019-06-03 09:00:56 +00:00