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}
Now that webrtc/* is moved to the root, this needs to be updated.
BUG=chromium:611808
NOTRY=True
TBR=mbonadei@webrtc.org
Change-Id: I947628886fb949972501e81e010bbdb9e9099872
Reviewed-on: https://webrtc-review.googlesource.com/1575
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19854}
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}
This was missed during review of https://codereview.webrtc.org/3010153002
Having python2 in the shebang makes it fail presubmit locally on Mac.
Disable 'invalid-name' PyLint rule in 3 places to pass presubmit.
NOTRY=True
NOTREECHECKS=True
TBR=charujain@webrtc.org
Bug: none
Change-Id: I85cc5783ba11774792cd8c2f6c0b4ff47ad89270
Reviewed-on: https://webrtc-review.googlesource.com/1566
Commit-Queue: Henrik Kjellander <kjellander@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19850}
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}
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}
in current local description.
When setting the descriptions, the order of m= sections would be compared
against existing m= sections and an error would be returned if the order
doesn't match.
Previously reviewed on: https://codereview.webrtc.org/3012313002/
BUG=chromium:763842
TBR=deadbeef@webrtc.org
Change-Id: I577e3424830b0a4c5ecd5524923873e30ad23d43
Reviewed-on: https://webrtc-review.googlesource.com/1200
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19842}
These rules are missing and this triggers a presubmit error when we move src/webrtc into src/.
NOTRY=True
TBR=solenberg@webrtc.org,stefan@webrtc.org,perkj@webrtc.org
Bug: chromium:611808
Change-Id: If81e5e42911c5de8bdd1288bc7aa61b713c2c5fd
Reviewed-on: https://webrtc-review.googlesource.com/1342
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19841}
We hit this CHECK even though the format wasn't even L16, because we
did the checked_cast before testing the codec name.
BUG=chromium:760994
TBR=ossu@webrtc.org
Change-Id: I382a2f841e51944495500f87650258024030d355
Reviewed-on: https://webrtc-review.googlesource.com/1224
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19835}
Rietveld has been having problems since the migration.
NOTRY=True
Bug: chromium:738330
Change-Id: I54538eee9f5734fac731702fb592580afcae3fae
Reviewed-on: https://webrtc-review.googlesource.com/1231
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19834}
- Adds a new AudioStatsTest, with better coverage of the same features, based on call_test.
- Adds an AudioEndToEndTest utility, which AudioStatsTest and LowBandwidthAudioTest uses.
BUG=webrtc:4690
R=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/3008273002 .
Cr-Commit-Position: refs/heads/master@{#19833}
Changed function definition from private to public. This was needed to test the function and to maintain the consistency.
BUG=webrtc:8197
NOTRY=True
R=kjellander@webrtc.org
Review-Url: https://codereview.webrtc.org/3010153002 .
Cr-Commit-Position: refs/heads/master@{#19831}
The content of webrtc/config.h has been moved to webrtc/api/rtpparameters.h, webrtc/call/rtp_config.h and webrtc/call/video_config.h.
BUG=webrtc:5876
NOTRY=True
TBR=stefan@webrtc.org
Change-Id: Id8d5b3b82b2362d561376d744fd1807c36076cae
Reviewed-on: https://webrtc-review.googlesource.com/1220
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19829}
This CL adds interfaces for the new video codec factories and wires them
up in WebRtcVideoEngine. The default behavior is unmodified however, and
the new code is currently unused except for the tests.
A follow-up CL will be uploaded for exposing them in the
PeerConnectionFactory API: https://codereview.webrtc.org/3004353002/.
BUG=webrtc:7925
R=andersc@webrtc.org, stefan@webrtc.org
Review-Url: https://codereview.webrtc.org/3007073002 .
Cr-Commit-Position: refs/heads/master@{#19828}
