cl was originally reviewed here:
https://codereview.webrtc.org/2060403002/
- Add task queue to Call with the intent of replacing the use of one of the process threads.
- Split VideoSendStream in two. VideoSendStreamInternal is created and used on the new task queue.
- BitrateAllocator is now created on libjingle's worker thread but always used on the new task queue instead of both encoder threads and the process thread.
- VideoEncoderConfig and VideoSendStream::Config support move semantics.
- The encoder thread is moved from VideoSendStream to ViEEncoder. Frames are forwarded directly to ViEEncoder which is responsible for timestamping ? and encoding the frames.
TBR=mflodman@webrtc.org
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/2250123002
Cr-Commit-Position: refs/heads/master@{#14014}
Move the webrtc/test/test_support/metrics sources into
test_support[_unittests] targets.
This is essentially reverting https://webrtc-codereview.appspot.com/5789004
and moving these sources back to the right target.
Add missing foreman_cif.yuv resource needed for these tests.
For MIPS, a compile error was surfacing for logcat_trace_context.h when
flipping bot to GN, which was fixed.
BUG=webrtc:5949
NOTRY=True
Review-Url: https://codereview.webrtc.org/2267113002
Cr-Commit-Position: refs/heads/master@{#13860}
Reason for revert:
Failed on Win 10 Chrome FYI.
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Win10%20Tester/builds/3847/steps/content_browsertests/logs/stdio
#
# Fatal error in e:\b\c\b\win_builder\src\third_party\webrtc\base\task_queue_win.cc, line 138
# last system error: 87
# Check failed: ((DWORD)0xFFFFFFFF) != result (4294967295 vs. 4294967295)
#
WebRtcBrowserTest
#
Original issue's description:
> - Add task queue to Call with the intent of replacing the use of one of the process threads.
>
> - Split VideoSendStream in two. VideoSendStreamInternal is created and used on the new task queue.
>
> - BitrateAllocator is now created on libjingle's worker thread but always used on the new task queue instead of both encoder threads and the process thread.
>
> - VideoEncoderConfig and VideoSendStream::Config support move semantics.
>
> - The encoder thread is moved from VideoSendStream to ViEEncoder. Frames are forwarded directly to ViEEncoder which is responsible for timestamping ? and encoding the frames.
>
> BUG=webrtc:5687
>
> Committed: https://crrev.com/cc168360f41322332860cb075edeb1cde21aa473
> Cr-Commit-Position: refs/heads/master@{#13767}
TBR=tommi@webrtc.org,mflodman@webrtc.org,stefan@webrtc.org,sprang@webrtc.org,pbos@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/2248713003
Cr-Commit-Position: refs/heads/master@{#13774}
- Split VideoSendStream in two. VideoSendStreamInternal is created and used on the new task queue.
- BitrateAllocator is now created on libjingle's worker thread but always used on the new task queue instead of both encoder threads and the process thread.
- VideoEncoderConfig and VideoSendStream::Config support move semantics.
- The encoder thread is moved from VideoSendStream to ViEEncoder. Frames are forwarded directly to ViEEncoder which is responsible for timestamping ? and encoding the frames.
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/2060403002
Cr-Commit-Position: refs/heads/master@{#13767}
It was possible that "A < B" and "B < A" both evaluated to true.
This manifested as an assert on Windows, and a memory leak on Linux.
Note that the concept of "less than" is meaningless for this object.
The operator is only needed so the object can be used as a key in an
std::map.
BUG=webrtc:6068
R=honghaiz@webrtc.org, kjellander@webrtc.org, skvlad@webrtc.org
Review URL: https://codereview.webrtc.org/2187913002 .
Cr-Commit-Position: refs/heads/master@{#13598}
SequencedTaskChecker is a helper class used to help verify that some methods
of a class are called on the same task queue or thread. A
SequencedTaskChecker is bound to a a task queue if the object is
created on a task queue, or a thread otherwise.
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/2125113003
Cr-Commit-Position: refs/heads/master@{#13452}
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}
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}
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}
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}
Reason for revert:
Fixed gyp bug.
Original issue's description:
> Revert of Android: Change camera fps range selection (patchset #4 id:100001 of https://codereview.webrtc.org/2013413002/ )
>
> Reason for revert:
> Breaks chromium fyi:
> https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/13565
> on step 'generate_build_files':
> gyp: /b/build/slave/Mac_Builder/build/src/third_party/build/android/test_runner.gypi not found
>
> Original issue's description:
> > Android: Change camera fps range selection
> >
> > This CL changes the logic in
> > CameraEnumerationAndroid.getClosestSupportedFramerateRange() to prefer
> > fps ranges with a low lower bound so the camera can adjust for
> > brightness conditions.
> >
> > To test the functionality of the fps range selection, JUnit tests are
> > added. This required a new target in api_tests.gyp. JUnit tests are
> > preferable over instrumentation tests
> > (libjingle_peerconnection_android_unittest) because they are faster and
> > simpler.
> >
> > R=kjellander@webrtc.org, sakal@webrtc.org
> >
> > Committed: https://crrev.com/b4ddb5c3d3706b1c02437f6a538576f3552ab908
> > Cr-Commit-Position: refs/heads/master@{#12964}
>
> TBR=sakal@webrtc.org,kjellander@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/b3f208d0ba45f140272e3e705b5cdadc3c76514b
> Cr-Commit-Position: refs/heads/master@{#12966}
TBR=sakal@webrtc.org,kjellander@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/2028583002
Cr-Commit-Position: refs/heads/master@{#12980}
We plan to add junit tests running with Robolectric
so naming these files "apk" is slightly confusing.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2020213002
Cr-Commit-Position: refs/heads/master@{#12971}
Reason for revert:
Breaks chromium fyi:
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/13565
on step 'generate_build_files':
gyp: /b/build/slave/Mac_Builder/build/src/third_party/build/android/test_runner.gypi not found
Original issue's description:
> Android: Change camera fps range selection
>
> This CL changes the logic in
> CameraEnumerationAndroid.getClosestSupportedFramerateRange() to prefer
> fps ranges with a low lower bound so the camera can adjust for
> brightness conditions.
>
> To test the functionality of the fps range selection, JUnit tests are
> added. This required a new target in api_tests.gyp. JUnit tests are
> preferable over instrumentation tests
> (libjingle_peerconnection_android_unittest) because they are faster and
> simpler.
>
> R=kjellander@webrtc.org, sakal@webrtc.org
>
> Committed: https://crrev.com/b4ddb5c3d3706b1c02437f6a538576f3552ab908
> Cr-Commit-Position: refs/heads/master@{#12964}
TBR=sakal@webrtc.org,kjellander@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/2021233002
Cr-Commit-Position: refs/heads/master@{#12966}
This CL changes the logic in
CameraEnumerationAndroid.getClosestSupportedFramerateRange() to prefer
fps ranges with a low lower bound so the camera can adjust for
brightness conditions.
To test the functionality of the fps range selection, JUnit tests are
added. This required a new target in api_tests.gyp. JUnit tests are
preferable over instrumentation tests
(libjingle_peerconnection_android_unittest) because they are faster and
simpler.
R=kjellander@webrtc.org, sakal@webrtc.org
Review URL: https://codereview.webrtc.org/2013413002 .
Cr-Commit-Position: refs/heads/master@{#12964}
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}
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}
Changed rtpdump converter and neteq tool to use new parser, but still aborting if the file is corrupt.
Review-Url: https://codereview.webrtc.org/1768773002
Cr-Commit-Position: refs/heads/master@{#12714}
- "WebRTC.Video.SendDelayInMs"
Change so that PacketOption packet id is always set in RtpSender (if having a TransportSequenceNumberAllocator).
Add SendDelayStats class for computing delays.
Add SendPacketObserver to RtpRtcp config and register SendDelayStats as observer.
Wire up OnSentPacket to SendDelayStats.
BUG=webrtc:5215
Review-Url: https://codereview.webrtc.org/1478253002
Cr-Commit-Position: refs/heads/master@{#12600}
- Places most ObjC code into webrtc/sdk/objc instead.
- New gyp targets to build, strip and export symbols for dylib.
- Removes old script used to generate dylib.
BUG=
Review URL: https://codereview.webrtc.org/1903663002
Cr-Commit-Position: refs/heads/master@{#12524}
The logging thread is always active. The main thread uses SwapQueues to pass events to the logging thread. The logging thread moves the events to either a RingBuffer history in memory, or to a string which is written to disc.
RtcEventLogImpl constructor takes a clock for easier testing.
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1687703002
Cr-Commit-Position: refs/heads/master@{#12476}
Reason for revert:
Breaks downstream for SRTP include paths. Will rework this and reland without that one.
Original issue's description:
> Remove the rtc_relative_path GYP variable and similar defines
>
> The defines that can be used to alter the include paths for Expat, SRTP
> and gtest are no longer needed in WebRTC or Chromium. Let's remove them
> to simplify the GYP a little.
>
> Removed defines:
> EXPAT_RELATIVE_PATH
> GTEST_RELATIVE_PATH
> SRTP_RELATIVE_PATH
>
> They're all set in the Chromium build so this shouldn't affect Chromium:
> https://code.google.com/p/chromium/codesearch#chromium/src/third_party/libjingle/libjingle.gyp
>
> BUG=webrtc:4256
> NOTRY=True
> NOPRESUBMIT=True
>
> Committed: https://crrev.com/e19cf59eb6ee44fd4d7e7fbcfdd1a6ea75063605
> Cr-Commit-Position: refs/heads/master@{#12467}
TBR=pthatcher@webrtc.org,perkj@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4256
Review URL: https://codereview.webrtc.org/1913043003
Cr-Commit-Position: refs/heads/master@{#12468}
