In SrtpTransport::SetRtcpParams, send_rtcp_session_ should really call
SetSend rather than SetRecv.
Modified the LOG message in SrtpTransport::SetRtpParams.
Bug: webrtc:8436
Change-Id: Iccbfbc5ef2d4f4ebd5f876c3f6dcc81671fdc631
Reviewed-on: https://webrtc-review.googlesource.com/14562
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20403}
This is a reland of b49b66109ea8a0a33a3192ebccf91366af2e49ae.
Original change's description:
> Rewrite WebRtcSession BUNDLE tests as PeerConnection tests
>
> Bug: webrtc:8222
> Change-Id: Id47e4544dc073564ad7e63d02865ca80dd5a85ff
> Reviewed-on: https://webrtc-review.googlesource.com/8280
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20365}
Bug: webrtc:8222
Change-Id: If3dcd8090875c641881e2b9e92fc1db387ba1de5
Reviewed-on: https://webrtc-review.googlesource.com/14400
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20397}
This is a reland of 3df5dcac9b339ba4d3f4969602f094c2c8035b51
Original change's description:
> Rewrite WebRtcSession media tests as PeerConnection tests
>
> Bug: webrtc:8222
> Change-Id: I782a3227e30de70eb8f6c26a48723cb3510a84ad
> Reviewed-on: https://webrtc-review.googlesource.com/6640
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20364}
Bug: webrtc:8222
Change-Id: I0a5398170d469eb9223bc781bfb417a85a72a2d2
Reviewed-on: https://webrtc-review.googlesource.com/14380
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20377}
AddRef() now returns void, and Release() returns an enum
RefCountReleaseStatus, to indicate whether or not this Release
call implied deletion.
Bug: webrtc:8270
Change-Id: If2fb77f26118b61751b51c856af187c72112c630
Reviewed-on: https://webrtc-review.googlesource.com/3320
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20366}
This reverts commit 54d1da13a584680ae80a1f229291e5bb7e76e6e1.
Reason for revert: Breaking tests
Original change's description:
> BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic.
>
> This CL implements the main logic and IOS appRTC integration.
>
> Unit tests and Android appRTC will be in separate CL.
>
> Bug: webrtc:8243
> Change-Id: If8e5195294046a47316e9fade1b0dfec211155e1
> Reviewed-on: https://webrtc-review.googlesource.com/4860
> Commit-Queue: Alex Narest <alexnarest@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20329}
TBR=deadbeef@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,alexnarest@webrtc.org
Change-Id: I5be1da78f360f72be66f9d56dd6b88c1cc13e963
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8243
Reviewed-on: https://webrtc-review.googlesource.com/12560
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20330}
This CL implements the main logic and IOS appRTC integration.
Unit tests and Android appRTC will be in separate CL.
Bug: webrtc:8243
Change-Id: If8e5195294046a47316e9fade1b0dfec211155e1
Reviewed-on: https://webrtc-review.googlesource.com/4860
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20329}
They're about to be removed.
BUG=webrtc:8396
Change-Id: Ie9a45f4c0dccb4414d2a2f939aa5f142edc6e4b6
Reviewed-on: https://webrtc-review.googlesource.com/12280
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20328}
This test was disabled for being flaky on the bots.
Try reenabling it to see if something has changed.
Bug: webrtc:7247
Change-Id: I65ce2cf6ce7a3761247369255d9ba106aa3e53f9
Reviewed-on: https://webrtc-review.googlesource.com/3262
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20320}
By having a unique_ptr own the callback data instead of a raw pointer,
the compiler helps us ensure that it's destroyed exactly once,
and never used after being destroyed.
(This made the callback object move-only, so I had to add support
for move-only callbacks to rtc::Thread::Invoke().)
BUG=webrtc:8111
Change-Id: Ia0804e4662e63e91e5cee18ecc3f38d2cfe8a26b
Reviewed-on: https://webrtc-review.googlesource.com/10812
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20317}
Unit test now checks that ADM:Init() works before any test runs.
It means that all tests will be skipped on bots that lack Pulse
support which is as how it worked before this CL as well. But then,
we detected the lack of support by checking that the audio layer had
changed from Pulse to Alsa.
As a consequence, I also decided to inject fake/mock ADMs in more
unit tests. One was actually already injected for other reasons
(see https://codereview.webrtc.org/2997383002/) but it had accidentally
been "reverted" later in combination with other changes.
To summarize: before this change we had a set of unit tests where real
audio was tested but it was not the intention of the test or required.
In addition, some Linux bots (VM:s) did not support PulseAudio and on
them the tests relied on a fallback mechanism to ALSA to work, i.e.,
we had a rather complex dependency on hardware. This dependency has now
been removed and it should result in more stable tests.
Bug: webrtc:7306, webrtc:7806
Change-Id: Ia0485658c04a4ef3b3f2dc0d557d73738067304b
Reviewed-on: https://webrtc-review.googlesource.com/8640
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20307}
This reverts commit a8264dbdd97f5e125d45fd0e84356f2e1f747df1.
Reason for revert: Reverting to unblock rolls into Chromium.
See failure here:
https://build.chromium.org/p/tryserver.chromium.linux/builders/linux_chromium_rel_ng/builds/565449
Fails: external/wpt/webrtc/RTCPeerConnection-setRemoteDescription-offer.html
I'm guessing these lines from the output are relevant:
12:15:32.525 11839 [1:19:1015/121532.495175:16438293900:ERROR:webrtcsession.cc(350)] Failed to set remote offer sdp: The order of m-lines in subsequent offer doesn't match order from previous offer/answer.
12:15:32.525 11839 [1:20:1015/121532.497199:16438296127:WARNING:delay_based_bwe.cc(326)] BWE Setting start bitrate to: 300000
12:15:32.525 11839 [1:1:1015/121532.498272:16438296963:ERROR:webrtcsdp.cc(359)] Failed to parse: "Invalid SDP". Reason: Expect line: v=
12:15:32.525 11839 [1:1:1015/121532.498364:16438297040:ERROR:rtc_peer_connection_handler.cc(2183)] Failed to create native session description. Type: offer SDP: Invalid SDP
12:15:32.525 11839 [1:1:1015/121532.498432:16438297104:ERROR:rtc_peer_connection_handler.cc(1458)] Failed to parse SessionDescription. Invalid SDP Expect line: v=
Original change's description:
> Reject the subsequent offer with fewer m= sections.
>
> If the subsequent offer contains fewer m= sections than the existing
> description, it would be rejected.
>
> The helper method MediaSectionsInSameOrder is modified and it will
> compare the number of m= sections before matching the media type.
>
> Bug: chromium:773620
> Change-Id: Ic8999445f4bc023da1d85a65659583db1687ec37
> Reviewed-on: https://webrtc-review.googlesource.com/9621
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20298}
TBR=deadbeef@webrtc.org,zhihuang@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:773620
Change-Id: I4a3ff7a42abb95144615b1dd37fb21585ee07b5d
Reviewed-on: https://webrtc-review.googlesource.com/10920
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20300}
If the subsequent offer contains fewer m= sections than the existing
description, it would be rejected.
The helper method MediaSectionsInSameOrder is modified and it will
compare the number of m= sections before matching the media type.
Bug: chromium:773620
Change-Id: Ic8999445f4bc023da1d85a65659583db1687ec37
Reviewed-on: https://webrtc-review.googlesource.com/9621
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20298}
This will allow Chrome to provide a RtcEventLogOutput object that reports the log back to Chrome, allowing Chrome to manage the log by itself - write it to a file, upload it to a server, etc.
Bug: webrtc:8111
Change-Id: I6a2a6945fc8586ef10e0fb9c56eaa8fda00dfc98
Reviewed-on: https://webrtc-review.googlesource.com/8081
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20295}
This will cause the application to be aborted before it encounters
something worse like a heap overflow, in case any bug in this code
exists or is introduced in the future.
TBR=zhihuang@webrtc.org
Bug: chromium:773620
Change-Id: Idd4e31aa63a3f673eefd3e8cb2ae3f4a5092ca4e
Reviewed-on: https://webrtc-review.googlesource.com/9040
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20293}
This is the same thing we're doing for usrsctp. Before this CL, the
first SrtpSession to call SetKey would initialize libsrtp, and
ChannelManager's destructor would deinitialize it. This works for an
application that only uses one instance of ChannelManager simultaneously
(or one instance of PeerConnectionFactory), but not one that uses
multiple.
Now, libsrtp is effectively reference-counted, with the first
SrtpSession to take a reference initializing it, and the last to remove
its reference deinitializing it.
This issue was discovered recently due to a change that resulted in
using srtp_update. Without using that method, the issue went unnoticed;
maybe srtp_protect/srtp_unprotect don't require initialization?
Bug: webrtc:8388
Change-Id: If1329360f0b469e454810e62e9b5acfbd4cba100
Reviewed-on: https://webrtc-review.googlesource.com/9000
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20262}
PeerConnectionInterfaceTest_StartAndStopLoggingAfterPeerConnectionClosed was using an invalid file, then checking that StartRtcEventLog returns false. Such a test might return a false positive, since StartRtcEventLog might fail because it was given an invalid file, rather than because the PC was already closed.
Bug: webrtc:8111
Change-Id: I844eb3b948b1406bb6f5cc63928eb26f0fb7b694
Reviewed-on: https://webrtc-review.googlesource.com/8541
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20253}
This patch adds an interface that allows modification of stun messages
sent by TurnPort. A user can inject a TurnCustomizer on the RTCConfig
and the TurnCustomizer will be invoked by TurnPort before sending
message. This allows user to e.g add custom attributes as described
in rtf5389.
BUG=webrtc:8313
Change-Id: I6f4333e9f8ff7fd20f32677be19285f15e1180b6
Reviewed-on: https://webrtc-review.googlesource.com/7618
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20233}
This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40.
Reason for revert:
Fixed the flake.
Original change's description:
> Revert "Added PeerConnectionObserver::OnRemoveTrack."
>
> This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc.
>
> Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway.
>
> Original change's description:
> > Added PeerConnectionObserver::OnRemoveTrack.
> >
> > This corresponds to processing the removal of a remote track step of
> > the spec, with processing the addition of a remote track already
> > covered by OnAddTrack.
> > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks
> >
> > Bug: webrtc:8260, webrtc:8315
> > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700
> > Reviewed-on: https://webrtc-review.googlesource.com/4722
> > Commit-Queue: Henrik Boström <hbos@webrtc.org>
> > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#20214}
>
> TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org
>
> Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8260, webrtc:8315
> Reviewed-on: https://webrtc-review.googlesource.com/7940
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Commit-Queue: Alex Loiko <aleloi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20218}
TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org
Change-Id: Iab7500bebf98535754b102874259de43831fff6b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8260, webrtc:8315
Reviewed-on: https://webrtc-review.googlesource.com/8180
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20227}
This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc.
Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway.
Original change's description:
> Added PeerConnectionObserver::OnRemoveTrack.
>
> This corresponds to processing the removal of a remote track step of
> the spec, with processing the addition of a remote track already
> covered by OnAddTrack.
> https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks
>
> Bug: webrtc:8260, webrtc:8315
> Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700
> Reviewed-on: https://webrtc-review.googlesource.com/4722
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20214}
TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org
Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8260, webrtc:8315
Reviewed-on: https://webrtc-review.googlesource.com/7940
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20218}
This corresponds to processing the removal of a remote track step of
the spec, with processing the addition of a remote track already
covered by OnAddTrack.
https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks
Bug: webrtc:8260, webrtc:8315
Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700
Reviewed-on: https://webrtc-review.googlesource.com/4722
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20214}
This reverts commit b23ed7f1af467a228cbdc63e839cac8856e9df8d.
Reason for revert: Breaks Chromium FYI build
Sample error log:
../../remoting/test/fake_port_allocator.cc:52:7: error: no matching constructor for initialization of 'cricket::BasicPortAllocator'
: BasicPortAllocator(network_manager, socket_factory),
^ ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
../../third_party/webrtc/p2p/client/basicportallocator.h:32:12: note: candidate constructor not viable: requires single argument 'network_manager', but 2 arguments were provided
explicit BasicPortAllocator(rtc::NetworkManager* network_manager);
^
../../third_party/webrtc/p2p/client/basicportallocator.h:27:7: note: candidate constructor (the implicit copy constructor) not viable: requires 1 argument, but 2 were provided
class BasicPortAllocator : public PortAllocator {
^
../../third_party/webrtc/p2p/client/basicportallocator.h:29:3: note: candidate constructor not viable: requires 3 arguments, but 2 were provided
BasicPortAllocator(rtc::NetworkManager* network_manager,
^
../../third_party/webrtc/p2p/client/basicportallocator.h:33:3: note: candidate constructor not viable: requires 3 arguments, but 2 were provided
BasicPortAllocator(rtc::NetworkManager* network_manager,
^
../../third_party/webrtc/p2p/client/basicportallocator.h:36:3: note: candidate constructor not viable: requires 5 arguments, but 2 were provided
BasicPortAllocator(rtc::NetworkManager* network_manager,
Original change's description:
> TurnCustomizer - an interface for modifying stun messages sent by TurnPort
>
> This patch adds an interface that allows modification of stun messages
> sent by TurnPort. A user can inject a TurnCustomizer on the RTCConfig
> and the TurnCustomizer will be invoked by TurnPort before sending
> message. This allows user to e.g add custom attributes as described
> in rtf5389.
>
> BUG=webrtc:8313
>
> Change-Id: Ibf5cc10af84c57288f1eb4c578ca064611a769f1
> Reviewed-on: https://webrtc-review.googlesource.com/4781
> Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20197}
TBR=deadbeef@webrtc.org,sakal@webrtc.org,jonaso@webrtc.org
Change-Id: I624efb22f6e3ceac1b2ff8af1ec47e4cfdde9140
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8313
Reviewed-on: https://webrtc-review.googlesource.com/7680
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20199}
This patch adds an interface that allows modification of stun messages
sent by TurnPort. A user can inject a TurnCustomizer on the RTCConfig
and the TurnCustomizer will be invoked by TurnPort before sending
message. This allows user to e.g add custom attributes as described
in rtf5389.
BUG=webrtc:8313
Change-Id: Ibf5cc10af84c57288f1eb4c578ca064611a769f1
Reviewed-on: https://webrtc-review.googlesource.com/4781
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20197}
These versions of StartRtcEventLog() are not used.
Bug: webrtc:8111
Change-Id: I1fb543a908decff203b13f8358598f75d875c111
Reviewed-on: https://webrtc-review.googlesource.com/6782
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20191}
This is in preparation for deleting the include in rtc_base/refcount.h,
but that change has to wait for some downstream applications to
not rely in the indirect include.
Partial reland of "Make rtc_base/refcount.h self contained, not including refcountedobject.h."
This is a reland of b7239a9dc825ddb25dbc800aed3a065163b9a10e
Original change's description:
> Make rtc_base/refcount.h self contained, not including refcountedobject.h.
>
> The refcount.h file doesn't depend on anything from
> refcountedobject.h. The motivation of this change to make it possible
> to add additional declarations to refcount.h, and include it from
> refcountedobject.h.
>
> Bug: webrtc:8270
> Change-Id: I24f6131f471e675570968d00065ff9b1f55e3373
> Reviewed-on: https://webrtc-review.googlesource.com/5760
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20106}
Bug: webrtc:8270
Change-Id: I63a42712f6c1ec83823c629d1a954fd1a04d4a6c
Reviewed-on: https://webrtc-review.googlesource.com/7281
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20185}
This is a reland of b7239a9dc825ddb25dbc800aed3a065163b9a10e
Original change's description:
> Make rtc_base/refcount.h self contained, not including refcountedobject.h.
>
> The refcount.h file doesn't depend on anything from
> refcountedobject.h. The motivation of this change to make it possible
> to add additional declarations to refcount.h, and include it from
> refcountedobject.h.
>
> Bug: webrtc:8270
> Change-Id: I24f6131f471e675570968d00065ff9b1f55e3373
> Reviewed-on: https://webrtc-review.googlesource.com/5760
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20106}
Bug: webrtc:8270
Change-Id: I9738f6680ab52d0f43639a1a39175fdba5957681
Reviewed-on: https://webrtc-review.googlesource.com/5840
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20180}
This is a reland of 9185aca9ce1f66e983d9a5e797cab77a64cc46b0
> Original change's description:
> > > Clean up libjingle API dependencies.
> > >
> > > This CL moves candidate.h into the public API, since it has
> > > been implicitly included before.
> > >
> > > This is a straightforward way of solving the circular
> > > dependencies involving that file. For instance,
> > > libjingle_peerconnection_api includes candidate.h from
> > > jsepicecandidate.h, but _api can't depend on rtc_p2p, which
> > > depends on _api. In fact, _api can't depend on much at all
> > > since it's a very high level abstraction; instead, things
> > > should depend on it.
> > >
> > > Furthermore, we have the case where deprecated headers
> > > include headers in internal modules. I just have to turn
> > > off include checking for those, but that's not a big deal.
> > >
> > > This CL punts the problem of callfactoryinterface.h being
> > > implicitly included, and pulling in most of the call
> > > module with it. This should be addressed in a follow-up
> > > CL.
> Bug: webrtc:7504
> Change-Id: Icae0ba1a0488550e2871cc65e66d3661707aa5b6
> Reviewed-on: https://webrtc-review.googlesource.com/6460
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20156}
TBR=deadbeef@webrtc.org
Bug: webrtc:7504
Change-Id: Ic6598ac2af9355b60bbd289c86dc75e0ae9fed2e
Reviewed-on: https://webrtc-review.googlesource.com/6801
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20167}
This reverts commit 9185aca9ce1f66e983d9a5e797cab77a64cc46b0.
Reason for revert: Still breaks Chromium:
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/29052
You cannot trust the *chromium* trybots due to https://bugs.chromium.org/p/chromium/issues/detail?id=771159
Original change's description:
> Reland "Clean up libjingle API dependencies."
>
> This is a reland of 5117b047875970cf61f2403b590c44c37bfa8272
> Original change's description:
> > > Clean up libjingle API dependencies.
> > >
> > > This CL moves candidate.h into the public API, since it has
> > > been implicitly included before.
> > >
> > > This is a straightforward way of solving the circular
> > > dependencies involving that file. For instance,
> > > libjingle_peerconnection_api includes candidate.h from
> > > jsepicecandidate.h, but _api can't depend on rtc_p2p, which
> > > depends on _api. In fact, _api can't depend on much at all
> > > since it's a very high level abstraction; instead, things
> > > should depend on it.
> > >
> > > Furthermore, we have the case where deprecated headers
> > > include headers in internal modules. I just have to turn
> > > off include checking for those, but that's not a big deal.
> > >
> > > This CL punts the problem of callfactoryinterface.h being
> > > implicitly included, and pulling in most of the call
> > > module with it. This should be addressed in a follow-up
> > > CL.
> > >
> > > Bug: webrtc:7504
> > > Change-Id: I1b1729408158418333ccdf702bf529386090f0d7
> > > Reviewed-on: https://webrtc-review.googlesource.com/2020
> > > Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
> > > Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#20034}
>
> Bug: webrtc:7504
> Change-Id: Icae0ba1a0488550e2871cc65e66d3661707aa5b6
> Reviewed-on: https://webrtc-review.googlesource.com/6460
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20156}
TBR=phoglund@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org
Change-Id: I699c68bd330b537005c3f2b8fe31702025df4e39
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7504
Reviewed-on: https://webrtc-review.googlesource.com/6800
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20157}
This is a reland of 5117b047875970cf61f2403b590c44c37bfa8272
Original change's description:
> > Clean up libjingle API dependencies.
> >
> > This CL moves candidate.h into the public API, since it has
> > been implicitly included before.
> >
> > This is a straightforward way of solving the circular
> > dependencies involving that file. For instance,
> > libjingle_peerconnection_api includes candidate.h from
> > jsepicecandidate.h, but _api can't depend on rtc_p2p, which
> > depends on _api. In fact, _api can't depend on much at all
> > since it's a very high level abstraction; instead, things
> > should depend on it.
> >
> > Furthermore, we have the case where deprecated headers
> > include headers in internal modules. I just have to turn
> > off include checking for those, but that's not a big deal.
> >
> > This CL punts the problem of callfactoryinterface.h being
> > implicitly included, and pulling in most of the call
> > module with it. This should be addressed in a follow-up
> > CL.
> >
> > Bug: webrtc:7504
> > Change-Id: I1b1729408158418333ccdf702bf529386090f0d7
> > Reviewed-on: https://webrtc-review.googlesource.com/2020
> > Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
> > Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#20034}
Bug: webrtc:7504
Change-Id: Icae0ba1a0488550e2871cc65e66d3661707aa5b6
Reviewed-on: https://webrtc-review.googlesource.com/6460
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20156}
Always use the PF_SRTP_BYPASS when sending RTP/RTCP packets.
Change the condition in BaseChannel::GetSrtpOverhead.
Get the SRTP overhead when using either SDES or DTLS-SRTP.
Bug: None
Change-Id: I44aeff8b75e56b12acefd73299a95a3e38cd401b
Reviewed-on: https://webrtc-review.googlesource.com/6580
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20152}
This is a reland of 34cdd2d402b08aee4e17a6fd38c87e0e5cd7aa30
Original change's description:
> Remove AudioDeviceObserver and make ADM not inherit from the Module interface.
>
> (Re-upload of https://codereview.webrtc.org/3020493002/)
>
> Bug: webrtc:4690, webrtc:7306
> Change-Id: I67fb9ebca1296aabc08eae8a292a5c69832dc35e
> Reviewed-on: https://webrtc-review.googlesource.com/5360
> Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20083}
Bug: webrtc:4690, webrtc:7306
Change-Id: Ib019439fe6ab0e6b759819e1e9bd320ba1d983bd
Reviewed-on: https://webrtc-review.googlesource.com/6300
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20137}
We need to support two modes of writing to the output:
1. Current way - the application lets lets WebRTC know which file to write to, and WebRTC is then in charge of the writing.
2. New way - the application would receive indications from WebRTC about (encoded) RTC events, and would itself be in charge of processing them (be it writing it to a file, uploading it somewhere, etc.).
We achieve this by creating an interface for output - RtcEventLogOutput. By providing an instance of the subclass, RtcEventLogOutputFile, the old behavior is achieved. The subclass of the new behavior is to be added by a later CL.
TBR=stefan@webrtc.org
Bug: webrtc:8111
Change-Id: I9c50521a7f7144d86d8353a65995795862e19c44
Reviewed-on: https://webrtc-review.googlesource.com/2686
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20135}
