This CL splits the iOS sdk into separate static libraries for video,
audio, ui, common, and peerconnection-related code. This will in the
future make it easier to compile WebRTC without unneeded components.
BUG=webrtc:4867
Review-Url: https://codereview.webrtc.org/2862543002
Cr-Commit-Position: refs/heads/master@{#18166}
In the windows audio device implementation, play block size should be
number of samples per 10ms times the number of channels. Meanwhile
RequestPlayoutData is expecting number of samples per channel in a
block, and we should pass in the per channel number here to avoid debug
check.
BUG=7627
Review-Url: https://codereview.webrtc.org/2876593003
Cr-Commit-Position: refs/heads/master@{#18088}
This allows it to be reused with Android AudioRecords created outside
WebRtcAudioRecord. WebRtcAudioEffects provides useful methods for enabling
hardware effects (such as echo cancellation) only on appropriate devices. It
also allows some control of these effects through WebRtcAudioUtils.
BUG=webrtc:7448
Review-Url: https://codereview.webrtc.org/2786603004
Cr-Commit-Position: refs/heads/master@{#18053}
Reason for revert:
https://chromium.googlesource.com/external/webrtc/+/3c1e558449309be965815e1bf...
I suspect this breaks Chrome ios.
https://build.chromium.org/p/tryserver.chromium.mac/builders/ios-simulator/bu...
https://luci-logdog.appspot.com/v/?s=chromium%2Fbb%2Ftryserver.chromium.mac%2...
Undefined symbols for architecture x86_64:
"base::allocator::g_replaced_default_zone", referenced from:
base::allocator::IsAllocatorInitialized() in libbase.a(allocator_check.o)
ld: symbol(s) not found for architecture x86_64
clang: error: linker command failed with exit code 1 (use -v to see invocation)
Traceback (most recent call last):
File "../../build/toolchain/mac/linker_driver.py", line 229, in <module>
Main(sys.argv)
File "../../build/toolchain/mac/linker_driver.py", line 79, in Main
subprocess.check_call(compiler_driver_args)
File
"/System/Library/Frameworks/Python.framework/Versions/2.7/lib/python2.7/subprocess.py",
line 540, in check_call
raise CalledProcessError(retcode, cmd)
Original issue's description:
> Extract iOS SDK helpers to separate target.
>
> This will reduce the redundant objects built
> in Chromium when opening up test_support for the fuzzer
> tests.
>
> BUG=webrtc:6828
> NOTRY=True
>
> Review-Url: https://codereview.webrtc.org/2849823002
> Cr-Commit-Position: refs/heads/master@{#17972}
> Committed: 3c1e558449TBR=mbonadei@webrtc.org,kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6828
Review-Url: https://codereview.webrtc.org/2853273002
Cr-Commit-Position: refs/heads/master@{#17979}
This will reduce the redundant objects built
in Chromium when opening up test_support for the fuzzer
tests.
BUG=webrtc:6828
NOTRY=True
Review-Url: https://codereview.webrtc.org/2849823002
Cr-Commit-Position: refs/heads/master@{#17972}
This target keeps track of .h the files under webrtc/modules/include/
that are not part of any target.
If a .h file is not part of a target the 'gn check' utility is not
able to spot if a target is missing a dependency because even if
it parses '#include' directives it is not able to find a target that
contains these headers.
BUG=webrtc:7513
NOTRY=True
Review-Url: https://codereview.webrtc.org/2838873002
Cr-Commit-Position: refs/heads/master@{#17880}
Make all rtc_source_test target that contains tests that
are included in a test executable only be visible to the
rtc_test target. Doing this exposed a couple of errors and
dependency problems that were resolved. Having this could
have prevented duplicated execution of tests like the case that
was recently fixed by deadbeef@ in
https://codereview.webrtc.org/2820263004
New targets:
* //webrtc/modules/rtp_rtcp:fec_test_helper
* //webrtc/modules/rtp_rtcp:mock_rtp_rtcp
* //webrtc/modules/remote_bitrate_estimator:mock_remote_bitrate_observer
The mock files and targets should probably be moved into webrtc/test in
the future, but that's out of the scope of this CL.
BUG=webrtc:5716
NOTRY=True
Review-Url: https://codereview.webrtc.org/2828793003
Cr-Commit-Position: refs/heads/master@{#17863}
Otherwise, the activeCount will become negative.
BUG=webrtc:7471
Review-Url: https://codereview.webrtc.org/2822233002
Cr-Commit-Position: refs/heads/master@{#17816}
Follow-up CL on https://codereview.webrtc.org/2788883002/ where I add a new
test which has to be enabled manually (will not run by default on bots).
Measures loopback latency and reports the min, max and average values for
a full duplex audio session.
The latency is measured like so:
- Insert impulses periodically on the output side.
- Detect the impulses on the input side.
- Measure the time difference between the transmit time and receive time.
- Store time differences in a vector and calculate min, max and average.
This test needs the '--gtest_also_run_disabled_tests' flag to run and also
some sort of audio feedback loop. E.g. a headset where the mic is placed
close to the speaker to ensure highest possible echo. It is also recommended
to run the test at highest possible output volume.
How to run:
./out/Debug/modules_unittests --gtest_filter=AudioDeviceMeasureLoopbackLatency --gtest_also_run_disabled_tests
Example output (on Linux machine):
[==========] Running 1 test from 1 test case.
[----------] Global test environment set-up.
[----------] 1 test from AudioDeviceTest
[ RUN ] AudioDeviceTest.DISABLED_MeasureLoopbackLatency
[..........]
[..........] [min, max, avg]=[59, 67, 64] ms
[ OK ] AudioDeviceTest.DISABLED_MeasureLoopbackLatency (10034 ms)
[----------] 1 test from AudioDeviceTest (10034 ms total)
[----------] Global test environment tear-down
[==========] 1 test from 1 test case ran. (10036 ms total)
[ PASSED ] 1 test.
BUG=webrtc:7273
Review-Url: https://codereview.webrtc.org/2826073002
Cr-Commit-Position: refs/heads/master@{#17791}
If we are on API level 21 or higher, it is possible to use a special AudioTrack
constructor that uses AudioAttributes and AudioFormat as input. It allows us to
supersede the notion of stream types for defining the behavior of audio playback,
and to allow certain platforms or routing policies to use this information for more
refined volume or routing decisions.
BUG=b/30018586
Review-Url: https://codereview.webrtc.org/2804013002
Cr-Commit-Position: refs/heads/master@{#17737}
Added new callback class that enables routing both the locally recorded
and playout audio data to external module.
A factory method that creates an AudioDeviceModule instance that also
registers the AudioTransportCapture with the device is also added.
BUG=webrtc:7337
R=solenberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2753453002 .
Cr-Commit-Position: refs/heads/master@{#17570}
The warnings in question are benign and should be supressed locally so
we can keep the analysis for the rest of the files.
BUG=webrtc:5748
Review-Url: https://codereview.webrtc.org/2790883002
Cr-Commit-Position: refs/heads/master@{#17485}
Finally we are able to remove this class entirely, along with the last
vestiges of it's use. I've also removed some legacy files that were only
used for windows XP support.
BUG=webrtc:7035
Review-Url: https://codereview.webrtc.org/2790533002
Cr-Commit-Position: refs/heads/master@{#17480}
Deletes left-over includes of trace.h and critical_section_wrapper.h.
BUG=webrtc:7035
Review-Url: https://codereview.webrtc.org/2784873002
Cr-Commit-Position: refs/heads/master@{#17460}
Always treat returning to foreground as interruption ended event, to help fix out of sync issues with interruption state.
BUG=0
Review-Url: https://codereview.webrtc.org/2780263002
Cr-Commit-Position: refs/heads/master@{#17457}
Makes sense, since it not recording/playing to begin with.
BUG=b/35415663
Review-Url: https://codereview.webrtc.org/2783673002
Cr-Commit-Position: refs/heads/master@{#17423}
This makes a few things a lot clearer when looking at perf trace data:
* What module instances (where they were created) are called
* On what thread
* How frequently
* For how long
ProcessThread will be replaced by TaskQueue moving forward and this is a step towards understanding the behavior of the affected code.
BUG=webrtc:7219
Review-Url: https://codereview.webrtc.org/2729053002
Cr-Commit-Position: refs/heads/master@{#16998}
Reason for revert:
I unfortunately have to revert this since trybots are still trying to run these tests and now they fail because the tests aren't there :-/ Before relanding, can you work with engprod to make sure we don't run them?
Original issue's description:
> Removes a test suite that is no longer used or maintained.
>
> BUG=NONE
>
> Review-Url: https://codereview.webrtc.org/2716843002
> Cr-Commit-Position: refs/heads/master@{#16825}
> Committed: bb5136f940TBR=solenberg@webrtc.org,nisse@webrtc.org,henrika@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=NONE
Review-Url: https://codereview.webrtc.org/2721433002
Cr-Commit-Position: refs/heads/master@{#16845}
This implementation uses various legacy classes such as EventTimeWrapper,
CriticalSectionWrapper, EventWrapper etc and hasn't been maintained
(or used?) for a long time.
Instead of spending time on testing and updating the class, I think
we should just remove it. For versions of Windows that we support,
following Win7, we use the CoreAudio implementation.
BUG=webrtc:7183
R=solenberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2700983002 .
Cr-Commit-Position: refs/heads/master@{#16678}
