This breaks the link from sdp_offer_answer.cc to channel.h.
Bug: webrtc:13931
Change-Id: I75608f75713bf4e69013ac5f5b17c19e53d07519
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261060
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36757}
This is an implementation API, user classes should in principle
only use the channel_interface.h
Bug: webrtc:13931
Change-Id: I85c285217858dc087c90a50792e980f731f4439f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260185
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36674}
This cl/
1) move WebRtcKeyValueConfig from api/transport to api/ directory.
2) add a test/ScopedKeyValueConfig (compare ScopedFieldTrials).
3) removes usage of webrtc::field_trial:: from the pc/ directory.
4) removes a few unused includes of system_wrappers/field_trial.h.
Bug: webrtc:10335
Change-Id: If29c07900dbe791050b0a5ad05332bedfad035f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253903
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36160}
Also apply IWYU to all .cc files in pc/, and correct BUILD file to match.
Note: Some files came out wrong when iwyu was applied. These are not included.
Bug: none
Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36064}
This is a safe cleanup change since top-level const applied to
parameters in function declarations (that are not also
definitions) are ignored by the compiler. Hence, such changes do
not change the type of the declared functions and are simply
no-ops.
Bug: webrtc:13610
Change-Id: Ibafb92c45119a6d8bdb6f9109aa8dad6385163a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249086
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35802}
This is a reland of commit 704a834f685eb96c9fcf891ca345557bef4d138a,
after it was reverted in order to merge a CL to M93.
Original change's description:
> Fix bug where we assume new m= sections will always be bundled.
>
> A recent change [1] assumes that all new m= sections will share the
> first BUNDLE group (if one already exists), which avoids generating
> ICE candidates that are ultimately unnecessary. This is fine for JSEP
> endpoints, but it breaks the following scenarios for non-JSEP endpoints:
>
> * Remote offer adding a new m= section that's not part of any BUNDLE
> group.
> * Remote offer adding an m= section to the second BUNDLE group.
>
> The latter is specifically problematic for any application that wants
> to bundle all audio streams in one group and all video streams in
> another group when using Unified Plan SDP, to replicate the behavior of
> using Plan B without bundling. It may try to add a video stream only
> for WebRTC to bundle it with audio.
>
> This is fixed by doing some minor re-factoring, having BundleManager
> update the bundle groups at offer time.
>
> Also:
> * Added some additional validation for multiple bundle groups in a
> subsequent offer, since that now becomes relevant.
> * Improved rollback support, because now rolling back an offer may need
> to not only remove mid->transport mappings but alter them.
>
> [1]: https://webrtc-review.googlesource.com/c/src/+/221601
>
> Bug: webrtc:12906, webrtc:12999
> Change-Id: I4c6e7020c0be33a782d3608dee88e4e2fceb1be1
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225642
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34544}
Bug: webrtc:12906, webrtc:12999
Change-Id: Id6acab2e2d7430c65f4b6a1d7372388a70cc18ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228465
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34728}
This reverts commit 704a834f685eb96c9fcf891ca345557bef4d138a.
Reason for revert: Reverting this in order to revert
https://webrtc-review.googlesource.com/c/src/+/221601, so we can
merge that revert to M93.
Original change's description:
> Reland "Fix bug where we assume new m= sections will always be bundled."
>
> This is a reland of d2b885fd91909f1b17fb11292a8c989d5d883b22, after
> making sure transports that are just being kept alive in case of
> rollback don't contribute to connection state, which broke a WPT.
>
> Original change's description:
> > Fix bug where we assume new m= sections will always be bundled.
> >
> > A recent change [1] assumes that all new m= sections will share the
> > first BUNDLE group (if one already exists), which avoids generating
> > ICE candidates that are ultimately unnecessary. This is fine for JSEP
> > endpoints, but it breaks the following scenarios for non-JSEP endpoints:
> >
> > * Remote offer adding a new m= section that's not part of any BUNDLE
> > group.
> > * Remote offer adding an m= section to the second BUNDLE group.
> >
> > The latter is specifically problematic for any application that wants
> > to bundle all audio streams in one group and all video streams in
> > another group when using Unified Plan SDP, to replicate the behavior of
> > using Plan B without bundling. It may try to add a video stream only
> > for WebRTC to bundle it with audio.
> >
> > This is fixed by doing some minor re-factoring, having BundleManager
> > update the bundle groups at offer time.
> >
> > Also:
> > * Added some additional validation for multiple bundle groups in a
> > subsequent offer, since that now becomes relevant.
> > * Improved rollback support, because now rolling back an offer may need
> > to not only remove mid->transport mappings but alter them.
> >
> > [1]: https://webrtc-review.googlesource.com/c/src/+/221601
> >
> > Bug: webrtc:12906, webrtc:12999
> > Change-Id: I4c6e7020c0be33a782d3608dee88e4e2fceb1be1
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225642
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#34544}
>
> Bug: webrtc:12906, webrtc:12999
> Change-Id: I68bf988b1918dd2d51de76e53e4fd696fea5a09b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227120
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34596}
TBR=hta@webrtc.org
Bug: webrtc:12906, webrtc:12999
Change-Id: I129d9eb3b9831317fa24b0263db191027246cb99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227821
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34666}
This is a reland of d2b885fd91909f1b17fb11292a8c989d5d883b22, after
making sure transports that are just being kept alive in case of
rollback don't contribute to connection state, which broke a WPT.
Original change's description:
> Fix bug where we assume new m= sections will always be bundled.
>
> A recent change [1] assumes that all new m= sections will share the
> first BUNDLE group (if one already exists), which avoids generating
> ICE candidates that are ultimately unnecessary. This is fine for JSEP
> endpoints, but it breaks the following scenarios for non-JSEP endpoints:
>
> * Remote offer adding a new m= section that's not part of any BUNDLE
> group.
> * Remote offer adding an m= section to the second BUNDLE group.
>
> The latter is specifically problematic for any application that wants
> to bundle all audio streams in one group and all video streams in
> another group when using Unified Plan SDP, to replicate the behavior of
> using Plan B without bundling. It may try to add a video stream only
> for WebRTC to bundle it with audio.
>
> This is fixed by doing some minor re-factoring, having BundleManager
> update the bundle groups at offer time.
>
> Also:
> * Added some additional validation for multiple bundle groups in a
> subsequent offer, since that now becomes relevant.
> * Improved rollback support, because now rolling back an offer may need
> to not only remove mid->transport mappings but alter them.
>
> [1]: https://webrtc-review.googlesource.com/c/src/+/221601
>
> Bug: webrtc:12906, webrtc:12999
> Change-Id: I4c6e7020c0be33a782d3608dee88e4e2fceb1be1
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225642
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34544}
Bug: webrtc:12906, webrtc:12999
Change-Id: I68bf988b1918dd2d51de76e53e4fd696fea5a09b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227120
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34596}
This reverts commit d2b885fd91909f1b17fb11292a8c989d5d883b22.
Reason for revert: Speculative revert for Chromium importer
Original change's description:
> Fix bug where we assume new m= sections will always be bundled.
>
> A recent change [1] assumes that all new m= sections will share the
> first BUNDLE group (if one already exists), which avoids generating
> ICE candidates that are ultimately unnecessary. This is fine for JSEP
> endpoints, but it breaks the following scenarios for non-JSEP endpoints:
>
> * Remote offer adding a new m= section that's not part of any BUNDLE
> group.
> * Remote offer adding an m= section to the second BUNDLE group.
>
> The latter is specifically problematic for any application that wants
> to bundle all audio streams in one group and all video streams in
> another group when using Unified Plan SDP, to replicate the behavior of
> using Plan B without bundling. It may try to add a video stream only
> for WebRTC to bundle it with audio.
>
> This is fixed by doing some minor re-factoring, having BundleManager
> update the bundle groups at offer time.
>
> Also:
> * Added some additional validation for multiple bundle groups in a
> subsequent offer, since that now becomes relevant.
> * Improved rollback support, because now rolling back an offer may need
> to not only remove mid->transport mappings but alter them.
>
> [1]: https://webrtc-review.googlesource.com/c/src/+/221601
>
> Bug: webrtc:12906, webrtc:12999
> Change-Id: I4c6e7020c0be33a782d3608dee88e4e2fceb1be1
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225642
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34544}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:12906, webrtc:12999
Change-Id: I00179d7573f322ad539ff16cad1f85320cfb2270
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227081
Reviewed-by: Björn Terelius <terelius@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34578}
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}
A recent change [1] assumes that all new m= sections will share the
first BUNDLE group (if one already exists), which avoids generating
ICE candidates that are ultimately unnecessary. This is fine for JSEP
endpoints, but it breaks the following scenarios for non-JSEP endpoints:
* Remote offer adding a new m= section that's not part of any BUNDLE
group.
* Remote offer adding an m= section to the second BUNDLE group.
The latter is specifically problematic for any application that wants
to bundle all audio streams in one group and all video streams in
another group when using Unified Plan SDP, to replicate the behavior of
using Plan B without bundling. It may try to add a video stream only
for WebRTC to bundle it with audio.
This is fixed by doing some minor re-factoring, having BundleManager
update the bundle groups at offer time.
Also:
* Added some additional validation for multiple bundle groups in a
subsequent offer, since that now becomes relevant.
* Improved rollback support, because now rolling back an offer may need
to not only remove mid->transport mappings but alter them.
[1]: https://webrtc-review.googlesource.com/c/src/+/221601
Bug: webrtc:12906, webrtc:12999
Change-Id: I4c6e7020c0be33a782d3608dee88e4e2fceb1be1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225642
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34544}
This entails instantly deleting the transport when it is no longer
referenced by any MID.
Also adds consistency checks to JsepTransportCollection.
Bug: webrtc:12837
Change-Id: I85775aeb676aac3a9aee74280cc72ac87a0f49b5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221982
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34273}
This is part of the work to make Bundle handling understandable,
so that we can get it to work right.
Bug: webrtc:12837
Change-Id: I77f046b4bac2d9709460b3b956a2edc3df0cdaac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221745
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34261}
This is step 1: Encapsulating the data.
Bug: webrtc:12837
Change-Id: I15df30dc294c90136a90b072608ed4c2e8925dcb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221602
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34245}
In this CL, JsepTransportController and MediaSessionDescriptionFactory
are updated not to assume that there only exists at most a single BUNDLE
group but a list of N groups. This makes it possible to create multiple
BUNDLE groups by having multiple "a=group:BUNDLE" lines in the SDP.
This makes it possible to have some m= sections in one group and some
other m= sections in another group. For example, you could group all
audio m= sections in one group and all video m= sections in another
group. This enables "send all audio tracks on one transport and all
video tracks on another transport" in Unified Plan. This is something
that was possible in Plan B because all ssrcs in the same m= section
were implicitly bundled together forming a group of audio m= section and
video m= section (even without use of the BUNDLE tag).
PeerConnection will never create multiple BUNDLE groups by default, but
upon setting SDP with multiple BUNDLE groups the PeerConnection will
accept them if configured to accept BUNDLE. This makes it possible to
accept an SFU's BUNDLE offer without having to SDP munge the answer.
C++ unit tests are added. This fix has also been verified manually on:
https://jsfiddle.net/henbos/to89L6ce/43/
Without fix: 0+2 get bundled, 1+3 don't get bundled.
With fix: 0+2 get bundled in first group, 1+3 get bundled in second
group.
Bug: webrtc:10208
Change-Id: Iaf451fa5459c484730c8018274166ef154b19af8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214487
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33838}
This reverts commit 5a40b3710545edfd8a634341df3de26f57d79281.
Reason for revert: Fixed the bug and ran layout tests.
Original change's description:
> Revert "Use the new DNS resolver API in PeerConnection"
>
> This reverts commit acf8ccb3c9f001b0ed749aca52b2d436d66f9586.
>
> Reason for revert: Speculative revert for https://ci.chromium.org/ui/p/chromium/builders/try/win10_chromium_x64_rel_ng/b8851745102358680592/overview.
>
> Original change's description:
> > Use the new DNS resolver API in PeerConnection
> >
> > Bug: webrtc:12598
> > Change-Id: I5a14058e7f28c993ed927749df7357c715ba83fb
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212961
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33561}
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> TBR=hta@webrtc.org
>
> Bug: webrtc:12598
> Change-Id: Idc9853cb569849c49052f9cbd865614710fff979
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213188
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33591}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:12598
Change-Id: Ief7867f2f23de66504877cdab1b23a11df2d5de4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214120
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33647}
This reverts commit 6e4fcac31312f2dda5b60d33874ff0cd62f94321.
Reason for revert: Parent CL issue has been resolved.
Original change's description:
> Revert "Remove thread hops from events provided by JsepTransportController."
>
> This reverts commit f554b3c577f69fa9ffad5c07155898c2d985ac76.
>
> Reason for revert: Parent CL breaks FYI bots.
> See https://webrtc-review.googlesource.com/c/src/+/206466
>
> Original change's description:
> > Remove thread hops from events provided by JsepTransportController.
> >
> > Events associated with Subscribe* methods in JTC had trampolines that
> > would use an async invoker to fire the events on the signaling thread.
> > This was being done for the purposes of PeerConnection but the concept
> > of a signaling thread is otherwise not applicable to JTC and use of
> > JTC from PC is inconsistent across threads (as has been flagged in
> > webrtc:9987).
> >
> > This change makes all CallbackList members only accessible from the
> > network thread and moves the signaling thread related work over to
> > PeerConnection, which makes hops there more visible as well as making
> > that class easier to refactor for thread efficiency.
> >
> > This CL removes the AsyncInvoker from JTC (webrtc:12339)
> >
> > The signaling_thread_ variable is also removed from JTC and more thread
> > checks added to catch errors.
> >
> > Bug: webrtc:12427, webrtc:11988, webrtc:12339
> > Change-Id: Id232aedd00dfd5403b2ba0ca147d3eca7c12c7c5
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206062
> > Commit-Queue: Tommi <tommi@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33195}
>
> TBR=nisse@webrtc.org,tommi@webrtc.org
>
> Change-Id: I6134b71b74a9408854b79d44506d513519e9cf4d
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:12427
> Bug: webrtc:11988
> Bug: webrtc:12339
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206467
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33203}
TBR=nisse@webrtc.org,tommi@webrtc.org,guidou@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:12427
Bug: webrtc:11988
Bug: webrtc:12339
Change-Id: I4e2e1490e1f9a87ed6ac4d722fd3c442e3059ae0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206809
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33225}
This reverts commit 6b143c1c0686519bc9d73223c1350cee286c8d78.
Reason for revert:
Relanding with updated expectations for SctpTransport::Information
based on TransceiverStateSurfacer in Chromium.
Original change's description:
> Revert "Fix unsynchronized access to mid_to_transport_ in JsepTransportController"
>
> This reverts commit 6cd405850467683cf10d05028ea0f644a68a91a4.
>
> Reason for revert: Breaks WebRTC Chromium FYI Bots
>
> First failure:
> https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Android%20Tests%20%28dbg%29%20%28L%20Nexus5%29/1925
>
> Failed tests:
> WebRtcDataBrowserTest.CallWithSctpDataAndMedia
> WebRtcDataBrowserTest.CallWithSctpDataOnly
>
> Original change's description:
> > Fix unsynchronized access to mid_to_transport_ in JsepTransportController
> >
> > * Added several thread checks to JTC to help with programmer errors.
> > * Avoid a few Invokes() to the network thread here and there such
> > as for fetching sctp transport name for getStats(). The transport
> > name is now cached when it changes on the network thread.
> > * JsepTransportController instances now get deleted on the network
> > thread rather than on the signaling thread + issuing an Invoke()
> > in the dtor.
> > * Moved some thread hops from JTC over to PC which is where the problem
> > exists and also (imho) makes it easier to see where hops happen in
> > the PC code.
> > * The sctp transport is now started asynchronously when we push down the
> > media description.
> > * PeerConnection proxy calls GetSctpTransport directly on the network
> > thread instead of to the signaling thread + blocking on the network
> > thread.
> > * The above changes simplified things for webrtc::SctpTransport which
> > allowed for removing locking from that class and delete some code.
> >
> > Bug: webrtc:9987, webrtc:12445
> > Change-Id: Ic89a9426e314e1b93c81751d4f732f05fa448fbc
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205620
> > Commit-Queue: Tommi <tommi@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33191}
>
> TBR=tommi@webrtc.org,hta@webrtc.org
>
> Change-Id: I7b2913d5133807589461105cf07eff3e9bb7157e
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9987
> Bug: webrtc:12445
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206466
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33204}
TBR=tommi@webrtc.org,hta@webrtc.org,guidou@webrtc.org
# Not skipping CQ checks because this is a reland.
Bug: webrtc:9987
Bug: webrtc:12445
Change-Id: Icb205cbac493ed3b881d71ea3af4fb9018701bf4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206560
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33219}
This reverts commit 6cd405850467683cf10d05028ea0f644a68a91a4.
Reason for revert: Breaks WebRTC Chromium FYI Bots
First failure:
https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Android%20Tests%20%28dbg%29%20%28L%20Nexus5%29/1925
Failed tests:
WebRtcDataBrowserTest.CallWithSctpDataAndMedia
WebRtcDataBrowserTest.CallWithSctpDataOnly
Original change's description:
> Fix unsynchronized access to mid_to_transport_ in JsepTransportController
>
> * Added several thread checks to JTC to help with programmer errors.
> * Avoid a few Invokes() to the network thread here and there such
> as for fetching sctp transport name for getStats(). The transport
> name is now cached when it changes on the network thread.
> * JsepTransportController instances now get deleted on the network
> thread rather than on the signaling thread + issuing an Invoke()
> in the dtor.
> * Moved some thread hops from JTC over to PC which is where the problem
> exists and also (imho) makes it easier to see where hops happen in
> the PC code.
> * The sctp transport is now started asynchronously when we push down the
> media description.
> * PeerConnection proxy calls GetSctpTransport directly on the network
> thread instead of to the signaling thread + blocking on the network
> thread.
> * The above changes simplified things for webrtc::SctpTransport which
> allowed for removing locking from that class and delete some code.
>
> Bug: webrtc:9987, webrtc:12445
> Change-Id: Ic89a9426e314e1b93c81751d4f732f05fa448fbc
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205620
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33191}
TBR=tommi@webrtc.org,hta@webrtc.org
Change-Id: I7b2913d5133807589461105cf07eff3e9bb7157e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9987
Bug: webrtc:12445
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206466
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33204}
This reverts commit f554b3c577f69fa9ffad5c07155898c2d985ac76.
Reason for revert: Parent CL breaks FYI bots.
See https://webrtc-review.googlesource.com/c/src/+/206466
Original change's description:
> Remove thread hops from events provided by JsepTransportController.
>
> Events associated with Subscribe* methods in JTC had trampolines that
> would use an async invoker to fire the events on the signaling thread.
> This was being done for the purposes of PeerConnection but the concept
> of a signaling thread is otherwise not applicable to JTC and use of
> JTC from PC is inconsistent across threads (as has been flagged in
> webrtc:9987).
>
> This change makes all CallbackList members only accessible from the
> network thread and moves the signaling thread related work over to
> PeerConnection, which makes hops there more visible as well as making
> that class easier to refactor for thread efficiency.
>
> This CL removes the AsyncInvoker from JTC (webrtc:12339)
>
> The signaling_thread_ variable is also removed from JTC and more thread
> checks added to catch errors.
>
> Bug: webrtc:12427, webrtc:11988, webrtc:12339
> Change-Id: Id232aedd00dfd5403b2ba0ca147d3eca7c12c7c5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206062
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33195}
TBR=nisse@webrtc.org,tommi@webrtc.org
Change-Id: I6134b71b74a9408854b79d44506d513519e9cf4d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12427
Bug: webrtc:11988
Bug: webrtc:12339
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206467
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33203}
Events associated with Subscribe* methods in JTC had trampolines that
would use an async invoker to fire the events on the signaling thread.
This was being done for the purposes of PeerConnection but the concept
of a signaling thread is otherwise not applicable to JTC and use of
JTC from PC is inconsistent across threads (as has been flagged in
webrtc:9987).
This change makes all CallbackList members only accessible from the
network thread and moves the signaling thread related work over to
PeerConnection, which makes hops there more visible as well as making
that class easier to refactor for thread efficiency.
This CL removes the AsyncInvoker from JTC (webrtc:12339)
The signaling_thread_ variable is also removed from JTC and more thread
checks added to catch errors.
Bug: webrtc:12427, webrtc:11988, webrtc:12339
Change-Id: Id232aedd00dfd5403b2ba0ca147d3eca7c12c7c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206062
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33195}
* Added several thread checks to JTC to help with programmer errors.
* Avoid a few Invokes() to the network thread here and there such
as for fetching sctp transport name for getStats(). The transport
name is now cached when it changes on the network thread.
* JsepTransportController instances now get deleted on the network
thread rather than on the signaling thread + issuing an Invoke()
in the dtor.
* Moved some thread hops from JTC over to PC which is where the problem
exists and also (imho) makes it easier to see where hops happen in
the PC code.
* The sctp transport is now started asynchronously when we push down the
media description.
* PeerConnection proxy calls GetSctpTransport directly on the network
thread instead of to the signaling thread + blocking on the network
thread.
* The above changes simplified things for webrtc::SctpTransport which
allowed for removing locking from that class and delete some code.
Bug: webrtc:9987, webrtc:12445
Change-Id: Ic89a9426e314e1b93c81751d4f732f05fa448fbc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205620
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33191}
Also removing need for lock for ice restart flag, fix call paths and
add information about how JsepTransportController's events could live
fully on the network thread and complexity around signaling thread
should be handled by PeerConnection (more details in webrtc:12427).
Bug: webrtc:12426, webrtc:12427
Change-Id: I9b1fae8acf16d90d9716054fc3c390700877a82a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205221
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33159}
This makes the config_ member thread-safe.
Required breaking out active_reset_srtp_params as a new member
variable, guarded by the network thread.
Bug: none
Change-Id: I81d542744116e5355c53695ea5531735587ba438
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204200
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33103}
- Remove the last sigslot from JsepTransportController.
- Tested the potential test failure on chromium blink test by importing
this CL to chromium source.
Bug: webrtc:11943
Change-Id: I107d05606350aff655ca73a5cb640dff1a7036ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202741
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33085}
- Replace few sigslot usages in jsep_transport_controller.
- There is still one sigslot usages in this file so keeping the inheritance
and that is the reason for not having a binary size gain in this CL.
- Remaining sigslot will be removed in a separate CL.
Bug: webrtc:11943
Change-Id: Idb8fa1090b037c48eeb62f54cffd3c485cebfcda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190146
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33034}
As discussed on a design review, the name RoboCaller is not clear
enough and switching to CallbackList will provide readability benefits.
Bug: webrtc:11943
Change-Id: I010cf0a91b5323e4e9c96b83703be7af1e67439c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190142
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32478}
This is a reland of 40261c3663fe316cfe40262c59cee993165ccf63
Note: Instead of changing the type of JsepTransportController->SignalSSLHandshakeError
added a new member with a different name and used it in webrtc code.
After this change do two more follow up CLs to completely remove the old code
from google3.
Original change's description:
> Replace sigslot usages with robocaller library.
>
> - Replace all the top level signals from jsep_transport_controller.
> - There are still sigslot usages in this file so keep the inheritance
> and that is the reason for not having a binary size gain in this CL.
>
> Bug: webrtc:11943
> Change-Id: I249d3b9710783aef70ba273e082ceeafe3056898
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185540
> Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32321}
Bug: webrtc:11943
Change-Id: Ia07394ee395f94836f6b576c3a97d119a7678e1a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186946
Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32359}
This reverts commit 40261c3663fe316cfe40262c59cee993165ccf63.
Reason for revert: Breaks downstream project
Original change's description:
> Replace sigslot usages with robocaller library.
>
> - Replace all the top level signals from jsep_transport_controller.
> - There are still sigslot usages in this file so keep the inheritance
> and that is the reason for not having a binary size gain in this CL.
>
> Bug: webrtc:11943
> Change-Id: I249d3b9710783aef70ba273e082ceeafe3056898
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185540
> Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32321}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,glahiru@webrtc.org
Change-Id: Icf438f87c3d95940d858db3cc5848b23abb82fc4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11943
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186844
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32324}
- Replace all the top level signals from jsep_transport_controller.
- There are still sigslot usages in this file so keep the inheritance
and that is the reason for not having a binary size gain in this CL.
Bug: webrtc:11943
Change-Id: I249d3b9710783aef70ba273e082ceeafe3056898
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185540
Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32321}
The eventual goal is to replace sigslot entirely, but we need to
start small, tread carefully, and evaluate how it works out.
Also add a few more RoboCaller unit tests to cover the types we
now use with RoboCaller.
Change-Id: I9a5814d1668a37546ea484ca88ec9c2be1913d25
Bug: webrtc:11943
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184660
Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32266}
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}
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}
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}
It was used only to provide parameters for MediaTransport.
Bug: webrtc:9719
Change-Id: I42e451ef84251ecf2b15010c7a3923b6fa2436be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177350
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31541}
