1158 Commits

Author SHA1 Message Date
Gustaf Ullberg
9a2e906b0c Added RTCMediaStreamTrackStats.concealmentEvents
The number of concealment events. This counter increases every time a concealed sample is
synthesized after a non-concealed sample. That is, multiple consecutive concealed samples
will increase the concealedSamples count multiple times but is a single concealment event.

Bug: webrtc:8246
Change-Id: I7ef404edab765218b1f11e3128072c5391e83049
Reviewed-on: https://webrtc-review.googlesource.com/1221
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19881}
2017-09-18 08:58:06 +00:00
deadbeef
d45aea8f42 Serialize "a=x-google-flag:conference".
There was a test for deserialization but not serialization. This was
probably always broken and no one noticed. I only noticed while
debugging something else.

BUG=None
TBR=pthatcher@webrtc.org

Review-Url: https://codereview.webrtc.org/3012383002
Cr-Commit-Position: refs/heads/master@{#19875}
2017-09-16 08:24:29 +00:00
deadbeef
5ada7acf60 If SRTP sessions exist, don't create new ones when applying answer.
Instead, call the "Update" methods of SrtpSession, which will just call
srtp_update, instead of wiping out the session state completely.

This was causing decryption to stop working when subsequent
offers/answers are applied. We don't know enough about SRTP to
understand the root cause, and I wasn't able to write an integration
test that reproduces the issue... But at least this fixes the bug that
can be reproduced reliably using Hangouts.

BUG=webrtc:8251

Review-Url: https://codereview.webrtc.org/3019443002
Cr-Commit-Position: refs/heads/master@{#19874}
2017-09-16 00:52:36 +00:00
Magnus Jedvert
58b0316f3d Expose new video codec factories in the PeerConnectionFactory API
This CL exposes the new type of video codec factories that represent all
video codecs in the PeerConnectionFactory API, i.e. no extra internal SW
video codecs will be added. Clients of the new functions will be
responsible for adding all SW video codecs themselves, and also handling
SW fallback and simulcast.

BUG=webrtc:7925
R=deadbeef@webrtc.org

Review-Url: https://codereview.webrtc.org/3004353002 .
Cr-Commit-Position: refs/heads/master@{#19866}
2017-09-15 17:02:50 +00:00
Mirko Bonadei
7120742701 Adding NOLINT for typedefs.h and common_types.h
Now that we have moved WebRTC from src/webrtc to src/, common_types.h
and typedefs.h are triggering a cpplint error.

The cpplint complaint is:
Include the directory when naming .h files  [build/include] [4]

This CL disables the error but we have to remove these two headers
from the root directory.

NOPRESUBMIT=true

Bug: webrtc:5876
Change-Id: I08e1b69aadcc4b28ab83bf25e3819d135d41d333
Reviewed-on: https://webrtc-review.googlesource.com/1577
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19859}
2017-09-15 13:03:51 +00:00
Patrik Höglund
563934e726 Clean up dependencies of peerconnection_unittest.
There were a number of unused includes and undeclared
dependencies. I removed the includes that were causing
problems and added dependencies for the includes that
turned out to be needed.

Bug: webrtc:7239,webrtc:6828
Change-Id: I5b57f9b8411d969e96eaa46fb49101b7b7c32284
Reviewed-on: https://webrtc-review.googlesource.com/1185
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19858}
2017-09-15 12:51:00 +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