This CL also implements support for getting the native context on
EGL 1.4. It's a bit tricker to get the native handle for EGL 1.0 so it
will be done in a separate CL.
Bug: webrtc:8257
Change-Id: I269e75c357f19507098180077fa9d1b1ac4dce23
Reviewed-on: https://webrtc-review.googlesource.com/1880
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19890}
After moving WebRTC from src/webrtc to src/ we can remove -webrtc from
the checkdeps configuration.
In this CL I also remove "+voice_engine_configurations.h" because this
header does not exist anymore.
NOTRY= True
Bug: chromium:611808
Change-Id: I4de427c51d78707f8107dd2dd1f834362d1c4da2
Reviewed-on: https://webrtc-review.googlesource.com/1845
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19888}
This check has been skipped during the migration from src/webrtc to
src. It was also reporting false positives. Now it should be fixed.
NOTRY=True
Bug: chromium:611808
Change-Id: Id8567dd92099e75ac35351f053829deebf28a9d1
Reviewed-on: https://webrtc-review.googlesource.com/1580
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19887}
Reason for revert:
Speculative revert since all Android bots on WebRTC FYI started to fail when this CL landed.
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28L%20Nexus6%29
Original issue's description:
> 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}
> Committed: https://webrtc.googlesource.com/src/+/5ada7acf603e90e71990e9d4ff8f49389f24958cTBR=zhihuang@webrtc.org,deadbeef@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:8251
NOTRY=TRUE
Review-Url: https://codereview.webrtc.org/3017543002
Cr-Commit-Position: refs/heads/master@{#19882}
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}
By making RtcEventLogImpl's ctor public, we remove the necessity to make the function a friend. Visibility of RtcEventLogImpl is still limited to the .cc file.
BUG=webrtc:8111
Change-Id: I774d2e93620a8d9f24299ef2a94f7593b490839d
Reviewed-on: https://webrtc-review.googlesource.com/1237
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19876}
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}
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}
I'm not sure why we ever had this in the first place, and it confuses
people on a nearly weekly basis, so let's get rid of it. The protocols
are enabled right after the corresponding gathering is done, so the only
real effect it has is to produce confusing log messages (first
"candidate not signaled because protocol not enabled", then "protocol
enabled, signaling candidate" right afterwards).
BUG=None
Review-Url: https://codereview.webrtc.org/3018483002
Cr-Commit-Position: refs/heads/master@{#19873}
Many of the tests follow a pattern of "wait for N candidates to be
gathered, then (without waiting) assert that gathering is complete". But
this only works if the "gathering complete" signal happens in the same
task as the last candidate being gathered, which isn't an API guarantee.
So the tests will be less fragile if they do the reverse: "wait for
gathering to be complete, then (without waiting) assert that N candidates
were gathered".
Also fixing some somewhat unrelated issues elsewhere. Like a test that
was supposed to be waiting for some period of time and ensuring no
additional candidates were gathered, but wasn't actually waiting at all.
BUG=None
Review-Url: https://codereview.webrtc.org/3018493002
Cr-Commit-Position: refs/heads/master@{#19872}
This CL adds an offset to the delay estimation used in AEC3 for
determining the alignment between the render and capture signals.
This ensures that there is no possibility for the capture loss to
cause the delay estimation to miss aligning the signals.
BUG=webrtc:8247, chromium:765242
Change-Id: I526dc7971b13425a28e99d69168fd3722a4cfdae
Reviewed-on: https://webrtc-review.googlesource.com/1232
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19871}
The limit on logs is not specific to the implementation of the logs, but is rather shared between all possible logs.
Also, by making it local to the .cc, not a member, we reduce the necessity of making RtcEventLog::Create a friend of the implementation. This necessity is removed completely by a following CL.
BUG=webrtc:8111
NOPRESUBMIT=True
Change-Id: I03044ed55ceeaf0064d5207b7407926571590699
Reviewed-on: https://webrtc-review.googlesource.com/1236
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19870}
rtc_event_log2text doesn't currently handle all possible RtcEvent-s.
1. Previous CL - to make sure events are not forgotten in the future, change the succession of if-statements to a switch, so that the compiler would complain if events are ever added, but are not handled here.
2. This CL - add handling of currently-unhandled events.
BUG=webrtc:8111
Change-Id: I5c726c077483b5d85cf8060674c8191a90cb84cc
Reviewed-on: https://webrtc-review.googlesource.com/1244
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19869}
Wrapper pattern is widely used in DesktopCapturer implementations. So this
change adds DesktopCapturerWrapper and CaptureResultDesktopCapturerWrapper as
the base classes of other wrappers. Implementing a new wrapper should become
easy, the implementation does not need to care about the uninteresting
overrides.
Bug: chromium:764258
Change-Id: If91c1b5e778805906f7f77854ea5600aa61bf64a
Reviewed-on: https://webrtc-review.googlesource.com/1420
Commit-Queue: Zijie He <zijiehe@google.com>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19868}
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}
Stats added for number of forced SW fallback changes per minute and percentage of time fallback is enabled for sent video streams:
- "WebRTC.Video.Encoder.ForcedSwFallbackChangesPerMinute.Vp8"
- "WebRTC.Video.Encoder.ForcedSwFallbackTimeInPercent.Vp8"
BUG=webrtc:6634
Review-Url: https://codereview.webrtc.org/3012863002
Cr-Commit-Position: refs/heads/master@{#19862}
rtc_event_log2text doesn't currently handle all possible RtcEvent-s.
1. This CL - to make sure events are not forgotten in the future, change the succession of if-statements to a switch, so that the compiler would complain if events are ever added, but are not handled here.
2. Next CL - add handling of currently-unhandled events.
BUG=webrtc:8111
NOPRESUBMIT=True
Change-Id: Ia4459b4e760eb0208823fdab69996de0e8420703
Reviewed-on: https://webrtc-review.googlesource.com/1242
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19861}
Set limits of NetEq minimum and maximum delay to 0-10000 ms closed interval.
Fixed error message in Audio Coding Module.
Bug: webrtc:6861
Change-Id: Id1b9928f808bdb6e1088c6895f2ec4a53b00efb2
Reviewed-on: https://webrtc-review.googlesource.com/1343
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19860}
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}
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}
Since webrtc/* has been moved to the top level
we should ignore it so it can be easily cleaned.
Right now there are usually at least .pyc files.
BUG=chromium:611808
NOTRY=True
Change-Id: If04284353a4e467583f810b2e5423c32269ba3cf
Reviewed-on: https://webrtc-review.googlesource.com/1571
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19851}
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}
