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}
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@{#13043}
Using an "EXPECT_WAIT" pattern with a long timeout rather than calling
"SleepMs" with a margin of 250ms.
BUG=webrtc:5953
Review-Url: https://codereview.webrtc.org/2029853002
Cr-Commit-Position: refs/heads/master@{#13040}
Everything except the dependency on
sdk/sdk_tests.gyp:rtc_sdk_peerconnection_objc_tests is included.
That one actually doesn't make those tests run as part of rtc_unittests
(by design). Fixing that + add them will done in aseparate CL.
BUG=webrtc:5949
TESTED=Built and ran rtc_unittests on Mac.
Verified the number of tests matched a run on the bot (1213 tests executed).
R=perkj@webrtc.org
Review URL: https://codereview.webrtc.org/2034923003 .
Cr-Commit-Position: refs/heads/master@{#13036}
The VUI part an SPS may specify max_num_reorder_frames and
max_dec_frame_buffering. These may cause a decoder to buffer a number
of frame prior allowing decode, leading to delays, even if no frames
using such references (ie B-frames) are sent.
Because of this we update any SPS block emitted by the encoder.
Also, a bunch of refactoring of H264-related code to reduce code
duplication.
BUG=
Review-Url: https://codereview.webrtc.org/1979443004
Cr-Commit-Position: refs/heads/master@{#13010}
Move the sources of rtc_unittests and xmllite_xmpp_unittests
into the actual targets instead of depending on none-targets.
This will make it easier to create GN targets matching them.
BUG=webrtc:5949
NOTRY=True
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2029583002
Cr-Commit-Position: refs/heads/master@{#13008}
BUG=webrtc:5949
TESTED=Built and ran the tests on Mac.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2025343002
Cr-Commit-Position: refs/heads/master@{#13007}
This will make it much less likely for application developers to not
realize the object is reference counted.
It also fixes a bug in the Java PeerConnection binding, by allowing a
reference to be transferred in the OnRemoveStream call via std::move.
BUG=webrtc:5128
R=pthatcher@webrtc.org, tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/1972793003 .
Cr-Commit-Position: refs/heads/master@{#12976}
This will be useful for any tests that test objects with time-dependent
behavior. It will allow such tests to be written in such a way that their
outcome is more repeatable (less flaky), and will also allow such tests
to finish quicker. For example, a test for STUN timeout doesn't need to
wait the full timeout interval in real time; it can simply advance the
simulated clock.
BUG=webrtc:4925
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1895933003 .
Cr-Commit-Position: refs/heads/master@{#12950}
Lowers risk of the event-tracing thread affecting measurements (and
performance of device). Entirely speculative, but shouldn't hurt.
Timings are still done on the thread that calls the trace macros.
BUG=
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/2013363002 .
Cr-Commit-Position: refs/heads/master@{#12943}
This makes it clearer that the C++ SurfaceTextureHelper owns its associated java object it.
In addition, arrange so that the SurfaceTextureHelper.stopListening
method (in java) can be called from any thread.
BUG=
Review-Url: https://codereview.webrtc.org/1988043002
Cr-Commit-Position: refs/heads/master@{#12941}
Due to the experiment in chromium relying on KT_DEFAULT = KT_RSA (bug:
crbug.com/611698) a conditional was introduced. Now that the experiment is
ending and the experiment flag has been removed we can make KT_DEFAULT=KT_ECDSA
unconditionally.
BUG=chromium:611698
Review-Url: https://codereview.webrtc.org/2009533003
Cr-Commit-Position: refs/heads/master@{#12935}
This CL adds these classes but does not change any functonality or interface
yet. This is in preparation for future CLs. To be used for this:
https://codereview.webrtc.org/2000163002/
RTCCertificateGenerator is meant to replace DtlsIdentityStoreInterface and
implementations. In order to continue to support mocking and to help with the
transition, RTCCertificateGenerator gets an interface that it implements (just
like the store has both interface and impl).
PeerConnectionFactoryInterface::CreatePeerConnection will take an
RTCCertificateGeneratorInterface instead of DtlsIdentityStoreInterface. As to
not break Chromium, both versions of CreatePeerConnection need to exist for a
transition period. This will be done by wrapping a store into a generator
wrapper - RTCCertificateGeneratorStoreWrapper.
BUG=webrtc:5707, webrtc:5708
R=hta@webrtc.org, tommi@chromium.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/2001103002 .
Cr-Commit-Position: refs/heads/master@{#12879}
This is needed as there are targets such as newlib_pnacl/remoting_client_plugin_newlib.pexe that depend on rtc_base_approved but don't need TaskQueue. We could implement support for TaskQueue in nacl using ppapi types, but it looks like there isn't a need for it. Libevent isn't supported for nacl either, so I'm introducing a layer on top of rtc_base_approved for TaskQueue. It's conceivable that this target will morph into a target that holds other threading primitives such as platform_thread and possibly socket related operations, which is also an area that we currently #ifdef out for nacl in a few places.
Functional change: Removes the "is_nacl" check.
R=phoglund@webrtc.org
Review-Url: https://codereview.webrtc.org/2001913002
Cr-Commit-Position: refs/heads/master@{#12878}
Reason for revert:
Seems like this CL cause
DtlsTransportChannelTest.TestReceiveClientHelloBeforeRemoteFingerprint
DtlsTransportChannelTest.TestReceiveClientHelloBeforeWritable
to consistently fail on Win DrMemory Full and for
DtlsTransportChannelTest.TestReceiveClientHelloBeforeRemoteFingerprint
DtlsTransportChannelTest.TestReceiveClientHelloBeforeWritable
to consistently fail on Linux Memcheck
Original issue's description:
> Change initial DTLS retransmission timer from 1 second to 50ms.
>
> This will help ensure a timely DTLS handshake when there's packet
> loss. It will likely result in spurious retransmissions (since the
> RTT is usually > 50ms), but since exponential backoff is still used,
> there will at most be ~4 extra retransmissions. For a time-sensitive
> application like WebRTC this seems like a reasonable tradeoff.
>
> R=juberti@chromium.org, juberti@webrtc.org, pthatcher@webrtc.org
>
> Committed: https://crrev.com/1e435628366fb9fed71632369f05928ed857d8ef
> Cr-Commit-Position: refs/heads/master@{#12853}
TBR=pthatcher@webrtc.org,juberti@webrtc.org,juberti@chromium.org,deadbeef@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/2002403002
Cr-Commit-Position: refs/heads/master@{#12864}
This will help ensure a timely DTLS handshake when there's packet
loss. It will likely result in spurious retransmissions (since the
RTT is usually > 50ms), but since exponential backoff is still used,
there will at most be ~4 extra retransmissions. For a time-sensitive
application like WebRTC this seems like a reasonable tradeoff.
R=juberti@chromium.org, juberti@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1981463002 .
Cr-Commit-Position: refs/heads/master@{#12853}
This helps a lot on Android devices where the user threads can be scheduled with low priority when the app is in the background, causing spurious significantly delayed before a packet can be read from the socket. With this patch the timestamp is taken by the kernel when the packet actually arrives.
R=juberti@chromium.orgTBR=juberti@webrtc.org
BUG=webrtc:5773
Review URL: https://codereview.webrtc.org/1944683002 .
Cr-Commit-Position: refs/heads/master@{#12850}
if the network monitor detects it after the native code does.
Also set the network cost for ethernet, wifi, unknown, cellular network type to be 0, 10, 50, 900,
so that unknown networks will have lower precedence than known networks with low cost (like Wifi) but higher precedence than known networks with high cost.
And third, infer network type based on limited name matching in Android if there is no network monitor or network monitor did not find the type.
BUG=webrtc:5890
R=pthatcher@chromium.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1976683003 .
Cr-Commit-Position: refs/heads/master@{#12833}
ACCESS_ON is an alias of GUARDED_BY but should take thread-like object instead of mutex,
RUN_ON is an alias of EXCLUSIVE_LOCKS_REQUIRED to annotate function always run on same
thread/task_queue
RTC_DCHECK_RUN_ON - creates an object that allows use of annotated variables and functions, and adds a run-time DCHECK given thread/queue is current.
R=tommi@webrtc.org
Review-Url: https://codereview.webrtc.org/1981893002
Cr-Commit-Position: refs/heads/master@{#12812}
