12 Commits

Author SHA1 Message Date
Niels Möller
ff40b142c0 Delete obsolete enable argument to SetVideoSend.
This argument was previously used to implement track muting
(black frames) in the video engine, but that now happens in
the VideoTrack/VideoBroadcaster upstream.

Bug: webrtc:6983
Change-Id: Ib721b297d9fbe55b641c56690dbbd37a52edbb2f
Reviewed-on: https://webrtc-review.googlesource.com/67341
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22785}
2018-04-09 08:45:29 +00:00
Niels Möller
f120cba82d Delete AudioMonitor and related code.
Bug: webrtc:8760
Change-Id: I0b11ec66b0f2576f52866864ba046191034a4d2d
Reviewed-on: https://webrtc-review.googlesource.com/39003
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Noah Richards <noahric@chromium.org>
Cr-Commit-Position: refs/heads/master@{#21801}
2018-01-30 09:48:29 +00:00
Zach Stein
ba37b4b075 Change return type of RtpSenderInterface::SetParameters from bool to RTCError
Note that RTCErrorTypes are mapped to the following DOMException names:
INTERNAL_ERROR -> OperationError
UNSUPPORTED_PARAMETER -> OperationError
INVALID_STATE -> InvalidStateError
INVALID_MODIFICATION -> InvalidModificationError
INVALID_RANGE -> RangeError

Bug: webrtc:8772
Change-Id: I44e3fe2456b007b8fb227d37d74b07ba226a19e4
Reviewed-on: https://webrtc-review.googlesource.com/37141
Commit-Queue: Zach Stein <zstein@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21766}
2018-01-25 19:06:04 +00:00
Steve Anton
4ab68eec96 Move sessiondescription.h/cc from p2p/base to pc/
SDP is a detail of PeerConnection and is not used by anything in p2p, so
it belongs in the pc/ directory. This also allows
MediaContentDescription to be inlined in the future.

Bug: webrtc:8620
Change-Id: I38b65ede9942e29eb15035ab29f2be988da1e5ce
Reviewed-on: https://webrtc-review.googlesource.com/33781
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21376}
2017-12-20 00:21:52 +00:00
Patrik Höglund
aba85d1f53 Resolve circular dependency in rtc_media_base.
This one was pretty straightforward fortunately.

Bug: webrtc:6828
Change-Id: Ie7b5e71f1298c409dbca2c74eaa09c0986e41d8f
Reviewed-on: https://webrtc-review.googlesource.com/25821
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20914}
2017-11-28 15:28:58 +00:00
Zhi Huang
5f5918f4ef Merge MediaChannel's OnTransportOverheadChanged and OnNetworkRouteChanged callbacks.
OnTransportOverChanged is merged into OnNetworkRouteChanged in MediaChannel
because the transport overhead will be added to rtc::NetworkRoute structure.

This CL depends on https://webrtc-review.googlesource.com/c/src/+/13520

Bug: None
Change-Id: I6ed6583f6c91db4ce61a89406de39774239f3a04
Reviewed-on: https://webrtc-review.googlesource.com/15200
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20678}
2017-11-14 20:42:36 +00:00
Steve Anton
e78bcb97c3 Enable cpplint in media/
Bug: webrtc:5584
Change-Id: I2fd1395d35596d9002e19cc90fcda3a5d4cde9e7
Reviewed-on: https://webrtc-review.googlesource.com/16564
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20504}
2017-10-31 17:46:42 +00:00
Steve Anton
8d3444df2d Reland "Rewrite WebRtcSession media tests as PeerConnection tests"
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}
2017-10-21 01:38:14 +00:00
Olga Sharonova
f2662f08e5 Revert "Rewrite WebRtcSession media tests as PeerConnection tests"
This reverts commit 3df5dcac9b339ba4d3f4969602f094c2c8035b51.

Reason for revert: suspected of breaking chromium.webrtc.fyi:
WebRtcBrowserTest.NegotiateUnsupportedVideoCodec
WebRtcBrowserTest.NegotiateNonCryptoCall

android https://uberchromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28L%20Nexus5%29/builds/25506
linux https://uberchromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Linux%20Tester/builds/38809
mac
https://uberchromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Mac%20Tester/builds/44120
windows
https://uberchromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Win10%20Tester/builds/9236

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}

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

Change-Id: Iaacc950d050ba2835d262908658dc045f234ef5b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8222
Reviewed-on: https://webrtc-review.googlesource.com/14160
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20375}
2017-10-20 13:00:45 +00:00
Steve Anton
3df5dcac9b 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}
2017-10-20 01:52:23 +00:00
Mirko Bonadei
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00
Mirko Bonadei
bb547203bf Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00