Compile fixes for GN on iOS that finally gets our bots green.
Changes to system_wrappers:
* Updated to only use inclusive sources for maintainability
* Add a few missing GN headers.
* Cleanup GYP hack for atomic32_mac.cc
* Renamed changes sources to avoid problems with GYP/GN file
suffix rules:
- atomic32_mac.cc -> atomic32_darwin.cc
- atomic32_posix.cc -> atomic32_non_darwin_unix.cc
See https://code.google.com/p/chromium/codesearch#chromium/src/build/config/BUILDCONFIG.gn&l=325
for details on which extensions can/cannot be used.
BUG=webrtc:5586
NOTRY=True
Review-Url: https://codereview.webrtc.org/1999723002
Cr-Commit-Position: refs/heads/master@{#12897}
Splits VideoCapturer::OnFrameCaptured into helper methods,
which enables use of the VideoAdaptation logic without
using a frame factory.
Refactors AndroidVideoCapturer to make adaptation decision
earlier, so we can crop and rotate using
NV12ToI420Rotate.
BUG=webrtc:5682
Review-Url: https://codereview.webrtc.org/1973873003
Cr-Commit-Position: refs/heads/master@{#12895}
The NetEqNetworkStatistics::expand_rate was not incremented during muted
state, which caused under-reporting of that metric. This change fixes
that.
BUG=chromium:613321, webrtc:5608
Review-Url: https://codereview.webrtc.org/2003203004
Cr-Commit-Position: refs/heads/master@{#12894}
Channel's API remains unchanged, but the creation of a BuiltinAudioDecoderFactory is now in Channel. The next step would be to amend Channel's API (through CreateChannel, I believe) to allow an AudioDecoderFactory to be sent along.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/1992763002
Cr-Commit-Position: refs/heads/master@{#12893}
Previously, failure codes were ignored, which meant simulcasted codecs
couldn't e.g. trigger software fallback. This stops the simulcasted
Encode call at the first faiulre and returns that code.
Another option is to continue sending the frame to the other encoders
but still return the first failure code. It's not clear that that is any
better than just quitting as soon as a failure happens.
BUG=
Review-Url: https://codereview.webrtc.org/2008723002
Cr-Commit-Position: refs/heads/master@{#12892}
A new API is added which enables detection of support of pro-audio on
Android. This is part of a larger change and the new API is not used yet.
Most likely it will only be used for logging purposes.
BUG=webrtc:5925
Review-Url: https://codereview.webrtc.org/2015483002
Cr-Commit-Position: refs/heads/master@{#12890}
Earlier, strings like "aaaa" and "1111" would be interpreted as IP
addresses, which is not optimal. This CL improves the IP_PATTERN and
adds a test for it.
Review-Url: https://codereview.webrtc.org/2009493002
Cr-Commit-Position: refs/heads/master@{#12887}
The return type of PacketSource::NextPacket() is changed from a naked
pointer to an std::uniqe_ptr. The interface contract was and still is
that the ownership is passed from the callee to the caller, but a
unique_ptr makes this explicit.
BUG=webrtc:2692
Review-Url: https://codereview.webrtc.org/2005873002
Cr-Commit-Position: refs/heads/master@{#12884}
tool to support all the functionality needed for simulating
and analyzing the audio processing module behavior during
calls.
BUG=
Review-Url: https://codereview.webrtc.org/1907223003
Cr-Commit-Position: refs/heads/master@{#12882}
indicating the usage of this helper is local.
With local usage critical section become obvisously useless and removed.
BUG=webrtc:5565
R=åsapersson
Review-Url: https://codereview.webrtc.org/1959013003
Cr-Commit-Position: refs/heads/master@{#12881}
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}
This is the minumum allowed size, and will allow STUN pings to be smaller.
The unit tests on the the Gturn are also modified. A username with length of 16 bytes will be generated for Gturn only.
Review-Url: https://codereview.webrtc.org/1848083002
Cr-Commit-Position: refs/heads/master@{#12876}
Reason for revert:
Reverting temporarily. Need to fix tests downstream that pass invalid arguments.
Original issue's description:
> Adding a some checks and switching out a few assert for RTC_[D]CHECK.
> These changes are around use of AudioFrame.data_ to help us catch issues earlier since assert() is left out in release builds, including builds with DCHECK enabled. I've also added a few full-on CHECKs to avoid reading past buffer boundaries or continuing on in a failed state.
>
> BUG=chromium:613482
> NOTRY=true
> (using notry due to offline android_arm64_rel bot)
>
> Committed: https://crrev.com/d36df89d40bde3c62ee5cbff841933e50b3c007b
> Cr-Commit-Position: refs/heads/master@{#12870}
TBR=henrik.lundin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:613482
Review-Url: https://codereview.webrtc.org/2006243002
Cr-Commit-Position: refs/heads/master@{#12874}
Consider follow up and use actual bucket count when storing samples.
BUG=
Review-Url: https://codereview.webrtc.org/2008483002
Cr-Commit-Position: refs/heads/master@{#12872}
There were a series of changes in the calculation of echo metrics. There changes made the existing unittests lose, e.g., EXPECT_EQ become EXPECT_NEAR. It is good time to protect the echo calculation more strictly.
The change is not simply generating a new reference file and change EXPECT_NEAR to EXPECT_EQ. It strengthens the test as well. Main changes are
1. the old test only sample a metric at the end of processing, while the new test takes metrics during the call with a certain time interval. This gives a much stronger protection.
2. added protection of a newly added metric, called divergent_filter_fraction.
3. as said, use EXPECT_EQ (actually ASSERT_EQ) instead of EXPECT_NEAR as much as possible, even for float point values. This may be too restrictive. But it can be good to be restrictive at the beginning.
BUG=
Review-Url: https://codereview.webrtc.org/1969403003
Cr-Commit-Position: refs/heads/master@{#12871}
These changes are around use of AudioFrame.data_ to help us catch issues earlier since assert() is left out in release builds, including builds with DCHECK enabled. I've also added a few full-on CHECKs to avoid reading past buffer boundaries or continuing on in a failed state.
BUG=chromium:613482
NOTRY=true
(using notry due to offline android_arm64_rel bot)
Review-Url: https://codereview.webrtc.org/2007563002
Cr-Commit-Position: refs/heads/master@{#12870}
I looked around and couldn't find any use of these dependencies.
NOTRY=true
(setting NOTRY since merge_libs.gyp isn't actually referenced by any gyp file, it's only used downstream)
Review-Url: https://codereview.webrtc.org/2007883002
Cr-Commit-Position: refs/heads/master@{#12868}
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}
For simplicity, this CL removes the need for Design Support Library.
UI is slightly changed for this to be possible. Floating Action Button
for adding favorite is removed and instead there's a button next to the
TextView.
Review-Url: https://codereview.webrtc.org/2003983002
Cr-Commit-Position: refs/heads/master@{#12861}
This CL makes the loop stop when all frames have been delivered and
start again when a new frame is inserted.
BUG=webrtc:5680
Review-Url: https://codereview.webrtc.org/2000103002
Cr-Commit-Position: refs/heads/master@{#12860}
Check for dropped frames by instead checking the
frame_buffer pointer directly.
Also add RTC_DCHECK to verify that a webrtc::VideoFrame never
has video_frame_buffer_ set to nullptr (except by the default
constructor).
BUG=webrtc:5682
Review-Url: https://codereview.webrtc.org/1995343002
Cr-Commit-Position: refs/heads/master@{#12859}
Some build tools downstream breaks since find on Linux
doesn't support the -E flag.
Shell commands also shouldn't be executed as part of GYP unless there's
no other way around the problem.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2008693002
Cr-Commit-Position: refs/heads/master@{#12858}
According to JSEP, the candidate filter does not affect pooled
candidates because they can be filtered once they're ready to be
surfaced to the application.
So, pooled port allocator sessions will use a filter of CF_ALL, with a
new filter applied when the session is taken by a P2PTransportChannel.
When the filter is applied:
* Some candidates may no longer be returned by ReadyCandidates()
* Some candidates may no longer have a "related address" (for privacy)
* Some ports may no longer be returned by ReadyPorts()
To simplify this, the candidate filtering logic is now moved up from
the Ports to the BasicPortAllocator, with some helper methods to perform
the filtering and stripping out of data.
R=honghaiz@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1998813002 .
Cr-Commit-Position: refs/heads/master@{#12856}
The STUN timeout is 9500ms, and the tests are waiting for 10000ms.
The 500ms margin of error is not enough for some bots (such as UBSan).
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2001093003 .
Cr-Commit-Position: refs/heads/master@{#12854}
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}
When local candidates are removed, signal to RTCPeerConnection
and eventually send to the remote client.
When a candidate-removal message is received, notify the native PeerConnection.
BUG=
R=tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/1972483002 .
Cr-Commit-Position: refs/heads/master@{#12852}
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}
Drop any pending texture frame when SurfaceTextureHelper.startListening()
is called because the frame might be from the previous
startListening()/stopListening() capture session. This typically happens
when switching between the front/back camera, and an old frame will get
incorrect rotation and mirroring because of the front/back camera
mismatch.
Dropping the frame in SurfaceTextureHelper also removes the need for
the |dropNextFrame| logic in VideoCapturerAndroid.
R=perkj@webrtc.org
Review URL: https://codereview.webrtc.org/2002963002 .
Cr-Commit-Position: refs/heads/master@{#12849}
During a period (e.g. 2 seconds), different metrics can be computed, e.g.:
- average of samples
- percentage/permille of samples
- units per second
Each periodic metric can be either:
- reported to an observer each period
- aggregated during the call (e.g. min, max, average)
BUG=webrtc:5283
Review-Url: https://codereview.webrtc.org/1640053003
Cr-Commit-Position: refs/heads/master@{#12847}
Of course, functions called WebRtcSpl_AddSatW32 and WebRtcSpl_SubSatW32 are supposed to handle overflow gracefully, and they probably did. But since the overflow handling depended on undefined behavior, a sufficiently smart optimizing compiler would have realized that it could just ignore the possibility of overflow and omit all the overflow handling code, leaving only the unadorned addition or subtraction.
Also, the new implementations, unlike the old ones, result in branch-free code (tested with clang 3.9 with -O2).
BUG=chromium:601728
Review-Url: https://codereview.webrtc.org/2002523002
Cr-Commit-Position: refs/heads/master@{#12846}
