And changed the minimum increase rate in |aimd_rate_control| to prevent the system from overusing on short twcc report send interval.
BUG=webrtc:6514
Review-Url: https://codereview.webrtc.org/2407143002
Cr-Commit-Position: refs/heads/master@{#17794}
in favor of GetPacketStatusCount/GetReceivedPackets
BUG=webrtc:5565
Review-Url: https://codereview.webrtc.org/2822153002
Cr-Commit-Position: refs/heads/master@{#17792}
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}
There's some code that resets the ICE role on an ICE restart (behavior
that's specified in ICE, but removed from ICEbis). And it wasn't taking
into account that the remote endpoint may be an ICE lite endpoint, in
which case the WebRTC endpoint's role should always be "controlling".
BUG=chromium:710760
Review-Url: https://codereview.webrtc.org/2812173003
Cr-Commit-Position: refs/heads/master@{#17779}
Instead of using the time on the first callback to Call::OnSentPacket, use the time when the first packet is sent from the pacer (to make sure this packet corresponds to an audio/video RTP packet).
BUG=webrtc:6244
Review-Url: https://codereview.webrtc.org/2825333002
Cr-Commit-Position: refs/heads/master@{#17777}
RTCVideoEncoder does not propagate RTP timestamps properly for encoded video frames, and as such whenever switching between simulcast layers there's a large timestamp gap that causes the incoming stream to freeze (timestamps look like they're either too far ahead or too far behind the previous frame).
Ideally RTCVideoEncoder would propagate these timestamps, but even so, when there's a large timestamp gap it would seem reasonable that the receiver resets quickly and consider this to be a new stream.
This CL detects the large jump for timestamps, if that happens, we reset the time extrapolator, which is the class for convertion from RTP timestamp to clock time.
BUG=chromium:705679
Review-Url: https://codereview.webrtc.org/2776813002
Cr-Commit-Position: refs/heads/master@{#17770}
Remove the ProbingIntervalEstimator and MockAimdRateControl.
BUG=webrtc:7441
Review-Url: https://codereview.webrtc.org/2789233005
Cr-Commit-Position: refs/heads/master@{#17769}
This CL improves the echo cancellation performance on setups where
headsets are used (systems with such low echo path gain
that no correlation between the render and capture signals
can be found) in 4 ways:
1) The echo path gain for systems with headsets is assumed to be
nonzero.
2) The stationary component of the render power is not included
in nonlinear echo power estimate.
3) The behavior after echo path gain changes is made less cautious.
4) The detection of systems with headsets is made more rapid.
BUG=chromium:712651, webrtc:6018
Review-Url: https://codereview.webrtc.org/2823903003
Cr-Commit-Position: refs/heads/master@{#17768}
It's not possible to enable it for the rtc_base_approved
target but since a larger refactoring is ongoing for webrtc/base
this CL doesn't attempt to fix that.
Changes made:
* Move webrtc/system_wrappers/include/stringize_macros.h into
webrtc/base:rtc_base_approved_unittests (and corresponding
unit test to rtc_base_approved_unittests).
* Move md5digest.* from rtc_base_approved to rtc_base_test_utils target.
* Move webrtc/system_wrappers/include/stringize_macros.h (+test) into
webrtc/base.
* Remove unused use include of webrtc/base/fileutils.h in
webrtc/base/pathutils.cc
BUG=webrtc:6828, webrtc:3806, webrtc:7480
NOTRY=True
Review-Url: https://codereview.webrtc.org/2717083002
Cr-Commit-Position: refs/heads/master@{#17766}
When degradation preference is kDegradationDisabled, do not update WebRTC.Video.CpuLimitedResolutionInPercent.
BUG=webrtc:6634
Review-Url: https://codereview.webrtc.org/2807133002
Cr-Commit-Position: refs/heads/master@{#17757}
The test isn't complete, since "track_id" ends up unset. But it's
better than having no test at all.
BUG=None
Review-Url: https://codereview.webrtc.org/2827643003
Cr-Commit-Position: refs/heads/master@{#17753}
These tests are already built into rtc_unittests, so they end up being
run three times. Fixed by creating a "p2p_test_utils" target that
contains the test utils that ortc_unittests and rtc_media_unittests
depend on, but not the tests themselves.
BUG=None
TBR=kjellander@webrtc.org
Review-Url: https://codereview.webrtc.org/2820263004
Cr-Commit-Position: refs/heads/master@{#17752}
Reason for revert:
Breaks checkdeps rules. Need to make a "p2p_test_utils" build target to include things like fakeicetransport.h.
Original issue's description:
> Remove rtc_p2p_unittests from ortc_unittests executable.
>
> These tests are already built into rtc_unittests; they shouldn't be
> built into two test executables.
>
> BUG=None
> TBR=kjellander@webrtc.org
> NOTRY=True
>
> Review-Url: https://codereview.webrtc.org/2820263004
> Cr-Commit-Position: refs/heads/master@{#17748}
> Committed: fe9d38f515TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=None
Review-Url: https://codereview.webrtc.org/2826703002
Cr-Commit-Position: refs/heads/master@{#17749}
Very similar to the current interface, but matches the new C++ structure, and
exposes the stats values as Objects which can be downcast to more specific
types (where the previous API only exposed the values as strings).
BUG=webrtc:6871
Review-Url: https://codereview.webrtc.org/2807933003
Cr-Commit-Position: refs/heads/master@{#17746}
