Reason for revert:
Breaks upstream code.
Original issue's description:
> Refactor NACK bitrate allocation
>
> Nack bitrate allocation should not be done on a per-rtp-module basis,
> but rather shared bitrate pool per call. This CL moves allocation to the
> pacer and cleans up a bunch if bitrate stats handling.
>
> BUG=
> R=danilchap@webrtc.org, stefan@webrtc.org, tommi@webrtc.org
>
> Committed: 5fc59e810bTBR=tommi@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review-Url: https://codereview.webrtc.org/2131913003
Cr-Commit-Position: refs/heads/master@{#13417}
ASSERT(!crit_.CurrentThreadIsOwner()) was racy due to use of a rtc::IsThreadRefEqual which cannot compare the thread handlers without a lock unless one is already sure it is the thread owning the crit.
Review-Url: https://codereview.webrtc.org/2131503002
Cr-Commit-Position: refs/heads/master@{#13411}
The function uses a deprecated function and is unused, so delete it.
BUG=6027
NOTRY=true
Review-Url: https://codereview.webrtc.org/2114013002
Cr-Commit-Position: refs/heads/master@{#13400}
Everything calls GetOSVersionName() instead, so no need to expose that.
Since it uses deprecated functions, this will make it easier to rewrite
it in a future change.
BUG=webrtc:6027
NOTRY=true
Review-Url: https://codereview.webrtc.org/2117823002
Cr-Commit-Position: refs/heads/master@{#13399}
There was a fast path in PreprocessToAddData that would just use the
input timestamps if the input format was equal to the required format of
the encoder. This works well as long as the codec never changes. If we
are first doing resampling (specifically upsampling) and then change to
a codec that does not require resampling, we'll need to stick to
whatever input timestamp we left off at, rather than silently accepting
whatever we're sent.
BUG=622435
Review-Url: https://codereview.webrtc.org/2119393002
Cr-Commit-Position: refs/heads/master@{#13398}
we will periodically check if any network does not have any connection on it and if yes, attempt to re-gather on those networks.
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2025573002 .
Cr-Commit-Position: refs/heads/master@{#13367}
Permits CHECKing/DCHECKing that methods are being accessed in a
thread-safe manner, even if they are not used by one single thread
(thread pools such as VideoToolbox OK).
BUG=
R=danilchap@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/2097403002 .
Cr-Commit-Position: refs/heads/master@{#13358}
The parameters for Logging.enableTracing() were creating the impression
that they control level and severity of one tracing system and they are
meant to be used together. In fact the "levels" parameter controlled one
tracing system (WEBRTC_TRACE), and the "severity" parameter was
responsible for a completely different one: setting the severity level
above which log messages from LOG() will be directed to the
platform-specific debug output (logcat on Android).
The method signature suggested that the "path" parameter applied to both
systems - while it was only meaningful for the WEBRTC_TRACE; LOG
messages were directed to ADB logcat no matter what the Path value was.
It is possible to redirect LOG messages to a file, but that is done
using a completely different set of APIs
- PeerConnectionFactory.startInternalTracingCapture().
I've separated these two methods to make it more clear which of the
parameters controls which system.
NOTRY=true
Review-Url: https://codereview.webrtc.org/2110853003
Cr-Commit-Position: refs/heads/master@{#13334}
If the AutoDetectProxy timed out resolving one address, it was
attempting to use the same resolver to resolve the next address,
which would always result in an assertion. This happened recently
a couple times on the Windows DrMemory bot because of its slowness.
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2104203002 .
Cr-Commit-Position: refs/heads/master@{#13326}
The fake clock has a few advantages:
1. It lets use verify that operations take the expected number of
round trips.
2. It makes the tests faster by letting us remove the equivalent
of "Sleep(500)" all over the tests.
3. It makes the tests less flaky, because sometimes sleeping for
500ms or waiting for 1s is not enough.
R=honghaiz@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2097793003 .
Cr-Commit-Position: refs/heads/master@{#13304}
Reason for revert:
Didn't intend to land yet. Chromium CL still needed.
Original issue's description:
> Add virtual Initialize methods to PortAllocator and NetworkManager.
>
> This will allow PeerConnection to handle hopping to the right thread
> and doing thread-specific initialization for the PortAllocator.
> This eliminates a required thread-hop for whatever is passing the
> PortAllocator into CreatePeerConnection.
>
> BUG=617648
> R=pthatcher@webrtc.org, skvlad@webrtc.org
>
> Committed: https://crrev.com/a6bdb0990a659ff9e7c4374f5033a6bcc4fbfb21
> Cr-Commit-Position: refs/heads/master@{#13283}
TBR=pthatcher@webrtc.org,skvlad@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=617648
Review-Url: https://codereview.webrtc.org/2092023004
Cr-Commit-Position: refs/heads/master@{#13284}
This will allow PeerConnection to handle hopping to the right thread
and doing thread-specific initialization for the PortAllocator.
This eliminates a required thread-hop for whatever is passing the
PortAllocator into CreatePeerConnection.
BUG=617648
R=pthatcher@webrtc.org, skvlad@webrtc.org
Review URL: https://codereview.webrtc.org/2097653002 .
Cr-Commit-Position: refs/heads/master@{#13283}
Reason for revert:
It turns out this revert was not necessary because the connection-state mapping for turn-turn connections was not done in connection.
Original issue's description:
> Revert of Revert "Revert of Update the BWE when the network route changes. (patchset #5 id:180001 of https://… (patchset #5 id:120001 of https://codereview.webrtc.org/2041593002/ )
>
> Reason for revert:
> ReadyToSendMedia did not consider the new presumed_writable state.
>
> Original issue's description:
> > Revert "Revert of Update the BWE when the network route changes. (patchset #5 id:180001 of https://codereview.webrtc.org/2000063003/ )"
> >
> > This reverts commit 72d41aa6da94dacb8a8464d1abd4ca7d1afffc65.
> >
> > New change made:
> > Do not reset the BWE when the new network route is not ready to send media.
> >
> > BUG=
> > R=pthatcher@webrtc.org, stefan@webrtc.org
> >
TBR=pthatcher@webrtc.org,stefan@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review-Url: https://codereview.webrtc.org/2094863003
Cr-Commit-Position: refs/heads/master@{#13282}
Reason for revert:
ReadyToSendMedia did not consider the new presumed_writable state.
Original issue's description:
> Revert "Revert of Update the BWE when the network route changes. (patchset #5 id:180001 of https://codereview.webrtc.org/2000063003/ )"
>
> This reverts commit 72d41aa6da94dacb8a8464d1abd4ca7d1afffc65.
>
> New change made:
> Do not reset the BWE when the new network route is not ready to send media.
>
> BUG=
> R=pthatcher@webrtc.org, stefan@webrtc.org
>
> Committed: https://crrev.com/5b5d2cdad7018993272525a723ef34f7da5c45f2
> Cr-Commit-Position: refs/heads/master@{#13280}
TBR=pthatcher@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review-Url: https://codereview.webrtc.org/2098703004
Cr-Commit-Position: refs/heads/master@{#13281}
If the specification for the speech encoder hasn't changed, we should
reuse it instead of recreating it. Otherwise, we lose its state. (This
problem was originally discovered because AudioEncoderOpus instances
would forget that they were supposed to be using DTX.)
BUG=webrtc:6020, chromium:622647
Review-Url: https://codereview.webrtc.org/2089183002
Cr-Commit-Position: refs/heads/master@{#13273}
Reason for revert:
The Webrtc waterfall indicates that this revert is not necessary.
Original issue's description:
> Revert of Do not delete a connection in the turn port with permission error or refresh error. (patchset #6 id:260001 of https://codereview.webrtc.org/2068263003/ )
>
> Reason for revert:
> It broke webrtc builds.
>
> Original issue's description:
> > Do not delete a connection in the turn port with permission error, refresh error, or binding error.
> >
> > Even if those error happened, the connection may still be able to receive packets for a while.
> > If we delete the connections, all packets arriving will be dropped.
> >
> > BUG=webrtc:6007
> > R=deadbeef@webrtc.org, pthatcher@webrtc.org
> >
> > Committed: https://crrev.com/3d77deb29c15bfb8f794ef3413837a0ec0f0c131
> > Cr-Commit-Position: refs/heads/master@{#13262}
>
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> TBR=pthatcher@webrtc.org,deadbeef@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:6007
>
> Committed: https://crrev.com/3159ffae6b1d5cba2ad972bd3d8074ac85f2c7f9
> Cr-Commit-Position: refs/heads/master@{#13265}
TBR=pthatcher@webrtc.org,deadbeef@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6007
Review-Url: https://codereview.webrtc.org/2090073003
Cr-Commit-Position: refs/heads/master@{#13266}
Reason for revert:
It broke webrtc builds.
Original issue's description:
> Do not delete a connection in the turn port with permission error, refresh error, or binding error.
>
> Even if those error happened, the connection may still be able to receive packets for a while.
> If we delete the connections, all packets arriving will be dropped.
>
> BUG=webrtc:6007
> R=deadbeef@webrtc.org, pthatcher@webrtc.org
>
> Committed: https://crrev.com/3d77deb29c15bfb8f794ef3413837a0ec0f0c131
> Cr-Commit-Position: refs/heads/master@{#13262}
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=pthatcher@webrtc.org,deadbeef@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6007
Review-Url: https://codereview.webrtc.org/2090833002
Cr-Commit-Position: refs/heads/master@{#13265}
Even if those error happened, the connection may still be able to receive packets for a while.
If we delete the connections, all packets arriving will be dropped.
BUG=webrtc:6007
R=deadbeef@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2068263003 .
Cr-Commit-Position: refs/heads/master@{#13262}
The Camera1 and Camera2 API use different size types. Camera1 uses
android.hardware.Camera.Size while Camera2 uses android.util.Size.
android.util.Size is only available from Lollipop forward so this CL
adds a similar Size class in CaptureFormat.
The purpose of this CL is to have a common size type that can be reused
from both Camera1 and Camera2 helper functions such as
CameraEnumerationAndroid.getClosestSupportedSize().
BUG=webrtc:5519
Review-Url: https://codereview.webrtc.org/2066773002
Cr-Commit-Position: refs/heads/master@{#13181}
An rtc::Buffer is used to write output of RBSP parsing, usually one byte
at a time. It turns out that container will then expand its capacity one
byte at a time, for each byte reallocating the whole buffer and copying
the contents over, turning this into an O(n^2) operation.
Fix is for now only to preallocate the container storage. Longer term, I
think we should mull over if we really need custom containers...
R=pbos@webrtc.orgTBR=mflodman@webrtc.org
BUG=
Review URL: https://codereview.webrtc.org/2073763002 .
Cr-Commit-Position: refs/heads/master@{#13167}
The function GetExecutablePath is a hack with poor portability. Delete
it and its caller GetTestFilePath. The latter was used in
videoframe_unittest.h, where it is replaced by
webrtc::test::ResourcePath.
Delete unused functions declared in base/testutils.h: ReadFile,
GetSiblingDirectory, GetGoogle3Directory, GetTalkDirectory,
CmpHelperFileEq, EXPECT_FILEEQ, ASSERT_FILEEQ.
Delete unused functions declared in media/base/testutils.h:
GetTestFilePath (see above), LoadPlanarYuvTestImage,
DumpPlanarYuvTestImage, ComputePSNR, ComputeSumSquareError.
The functions LoadPlanarYuvTestImage, DumpPlanarYuvTestImage were used
in yuvscaler_unittests.cc and planarfunctions_unittests.cc, under
webrtc/pc. However, these tests are never compiled or run, and appear
not to have been for the last few years, and are therefore deleted
rather than updated. It might make sense to check if libyuv have
comparable tests, and if not, resurrect them as part of libyuv
unittests.
BUG=
R=perkj@webrtc.org
Review URL: https://codereview.webrtc.org/2058043002 .
Cr-Commit-Position: refs/heads/master@{#13163}
Reason for revert:
Reverting after experiment. What seems to happen is that the thread can't be stopped when PostThreadMessage fails with this error code (or lack thereof).
Original issue's description:
> Attempt to figure out what the issue is on the Win10 FYI build bot in content_browsertests.
> Check for the case when PostThreadMessage returns false but GetLastError returns ERROR_SUCCESS.
>
> TBR=olka
> NOTRY=true
>
> Committed: https://crrev.com/e11041159d6e66fb25c2e045ad1558fc99d2d6cd
> Cr-Commit-Position: refs/heads/master@{#13143}
TBR=olka@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2067013003
Cr-Commit-Position: refs/heads/master@{#13147}
Check for the case when PostThreadMessage returns false but GetLastError returns ERROR_SUCCESS.
TBR=olka
NOTRY=true
Review-Url: https://codereview.webrtc.org/2063313003
Cr-Commit-Position: refs/heads/master@{#13143}
The error hasn't been noticed since we don't really do
(or support) Mac 32-bit builds.
BUG=webrtc:5949
NOTRY=True
Review-Url: https://codereview.webrtc.org/2065583002
Cr-Commit-Position: refs/heads/master@{#13111}
Changes:
* Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope
to match GYP.
* Enable sctpdataengine_unittest.cc for iOS, which should have
been done in https://codereview.webrtc.org/1587193006
* Renamed GN target rtc_base_test_utils -> rtc_base_tests_utils
to match GYP.
* Added dependencies on call, modules/video_coding and video for
rtc_media.
* Added dependency on audio for rtc_media_unitttests (couldn't be
added to rtc_media due to circular dependency problem).
BUG=webrtc:5949
TESTED=Built and ran the tests on Mac.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2050313002
Cr-Commit-Position: refs/heads/master@{#13106}
Due to a bug, the NetworkManager was reconnecting to the NetworkMonitor's NetworkChanged signal every time the network manager is stopped and restarted. After each calls, one more listener was added to the signal and never removed - which caused OnNetworksChanged to be called multiple times on each actual network change.
Not sure if this had any negative effect other than the extraneous "Network changed" lines in WebRTC logs, but it wasn't working correctly either way.
The fix is to only subscribe to the signal once, when the NetworkMonitor is created.
TBR=pthatcher
BUG=
NOTRY=True
Review-Url: https://codereview.webrtc.org/2054583002
Cr-Commit-Position: refs/heads/master@{#13105}
Every message will now be traced with the location from which it was
posted, including function name, file and line number.
This CL also writes a normal LOG message when the dispatch took more
than a certain amount of time (currently 50ms).
This logging should help us identify messages that are taking
longer than expected to be dispatched.
R=pthatcher@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/2019423006 .
Cr-Commit-Position: refs/heads/master@{#13104}
To avoid the case where a single data point or too short window is used,
causing bad behavior due to bad stats, update RateStatistics to return
an Optional rather than a plain rate.
There was also a strange off by one bug where the rate was slightly
overestimated (N + 1 buckets, N ms time window).
These changes requires updates to a number of places, and may very well
cause seeming perf regressions (but the stats were probablty more wrong
previously).
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2029593002 .
Cr-Commit-Position: refs/heads/master@{#13103}
to be compatible with projects that has own base/thread_annotation.h
NOTRY=TRUE
Review-Url: https://codereview.webrtc.org/2055473002
Cr-Commit-Position: refs/heads/master@{#13090}
Previously RefCountedObject was passing all parameters by value.
This meant that it was hard to use it with movable types, such
as unique_ptr<>. Now there is a constructor that takes r-value,
which means that RefCountedObject<std::unique_ptr<foo>> can be
initialized by passing std::unique_ptr<foo> to the constructor.
Review-Url: https://codereview.webrtc.org/2036123002
Cr-Commit-Position: refs/heads/master@{#13079}
In https://codereview.webrtc.org/2034923003 it was discovered
that a test binary rtc_sdk_peerconnection_objc_tests was
a dependency to rtc_unittests. Unfortunately gtest doesn't
include dependent executables into the same test executable;
only libraries (so theses tests weren't run).
This CL incorporates those tests into rtc_unittests and
does the same changes to the GN build.
BUG=webrtc:5949
TESTED=Built and ran rtc_unittests locally on Mac.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2041743003
Cr-Commit-Position: refs/heads/master@{#13060}
When the fake clock's time is advanced, it now ensures all pending
queued messages have been dispatched. This allows us to write a
"SIMULATED_WAIT" macro that ticks the simulated clock by milliseconds up
until the target time.
Useful in this case, where we know the STUN timeout should take a total
of 9500ms, but it would be overly complex to write test code that waits
for each individual timeout, ensures a STUN packet has been
retransmited, etc.
(The test described above *should* be written, but it belongs in
p2ptransportchannel_unittest.cc, not webrtcsession_unittest.cc).
Review-Url: https://codereview.webrtc.org/2024813004
Cr-Commit-Position: refs/heads/master@{#13052}
Reason for revert:
There seems to be a TSan warning that wasn't caught by the trybot: https://build.chromium.org/p/client.webrtc/builders/Linux%20Tsan%20v2/builds/6732/steps/peerconnection_unittests/logs/stdio
Apparently a thread is still alive even after destroying WebRTCSession. Need to think of a way to fix this, without adding a critical section around g_clock (since that would hurt performance).
Original issue's description:
> Improving the fake clock and using it to fix a flaky STUN timeout test.
>
> When the fake clock's time is advanced, it now ensures all pending
> queued messages have been dispatched. This allows us to write a
> "SIMULATED_WAIT" macro that ticks the simulated clock by milliseconds up
> until the target time.
>
> Useful in this case, where we know the STUN timeout should take a total
> of 9500ms, but it would be overly complex to write test code that waits
> for each individual timeout, ensures a STUN packet has been
> retransmited, etc.
>
> (The test described above *should* be written, but it belongs in
> p2ptransportchannel_unittest.cc, not webrtcsession_unittest.cc).
>
> Committed: https://crrev.com/ffbe0e17e2c9b7fe101023acf40574dc0c95631a
> Cr-Commit-Position: refs/heads/master@{#13043}
TBR=pthatcher@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2038213002
Cr-Commit-Position: refs/heads/master@{#13045}
