Added thumbnail streams functionality to video quality test.
Changed simulcast full-stack tests to be 30fps instead of 50 to
better reflect real usecases (expect all kind of perf metrics to
improve).
BUG=webrtc:7095, webrtc:7301
Review-Url: https://codereview.webrtc.org/2733943003
Cr-Commit-Position: refs/heads/master@{#17092}
CurrentNtp return time by taking two output parameters by reference
(also breaks style guide)
CurrentNtpTime treat ntp time as single entity and returns it using NtpTime structure.
(making interface clearer)
BUG=None
Review-Url: https://codereview.webrtc.org/2733823002
Cr-Commit-Position: refs/heads/master@{#17088}
These changes are all no-op, only affecting the appearance of the code. The file names are changed to match the class name.
BUG=none
Review-Url: https://codereview.webrtc.org/2731993002
Cr-Commit-Position: refs/heads/master@{#17087}
This CL removes most of the global frame state in VideoProcessor and
replaces that with a vector of frame states. This is useful for pipelining
codecs, where the encoded/decoded frame may not be immediately outputted
after it has been sent to the codec.
The callers (VideoProcessorIntegrationTest and video_quality_measurement)
still call VideoProcessor in a sequential fashion. A follow-up CL will be
submitted that enables batch mode in VideoProcessorIntegrationTest.
Note that VideoProcessor is still not thread safe. Currently, we can run
fairly well on Android due to the synchronicity of our MediaCodec wrapper,
but we still cannot run on iOS due to async issues. This will be fixed in
the future.
BUG=webrtc:6634
Review-Url: https://codereview.webrtc.org/2711133002
Cr-Commit-Position: refs/heads/master@{#17084}
Add system version check functionality in UIDevice+RTCDevice category.
Check for iOS system version when handle capture session interruption.
BUG=webrtc:7201
Review-Url: https://codereview.webrtc.org/2733773003
Cr-Commit-Position: refs/heads/master@{#17079}
for consistency with the WebRTC 1.0 standard as suggested in a TODO.
BUG=None
Review-Url: https://codereview.webrtc.org/2732663004
Cr-Commit-Position: refs/heads/master@{#17077}
vector<unique_ptr<StunAttribute>> as suggested in a TODO.
BUG=NONE
Review-Url: https://codereview.webrtc.org/2735523002
Cr-Commit-Position: refs/heads/master@{#17076}
WebRtcVideoChannel2Test::SetRecvCodecsSetsFecForAllVideoCodecs was never
fully implemented and hasn't been touched in over a year.
BUG=NONE
Review-Url: https://codereview.webrtc.org/2736483002
Cr-Commit-Position: refs/heads/master@{#17075}
It was defined unconditionally and the code for non-HAVE_SRTP was unmaintained
and failed to compile.
BUG=webrtc:7294
Review-Url: https://codereview.webrtc.org/2729373002
Cr-Commit-Position: refs/heads/master@{#17074}
Reason for revert:
webrtc_perf_tests crashes on android and windows due to too large test.
Original issue's description:
> Added large room scenario to full-stack tests. Added thumbnail streams functionality to video quality test.
>
> Changed simulcast full-stack tests to be 30fps instead of 50 to better reflect real usecases (expect all kind of perf metrics to improve).
>
> BUG=webrtc:7095
>
> Review-Url: https://codereview.webrtc.org/2730073002
> Cr-Commit-Position: refs/heads/master@{#17068}
> Committed: d8bd1b1d82TBR=sprang@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:7095
Review-Url: https://codereview.webrtc.org/2734753004
Cr-Commit-Position: refs/heads/master@{#17071}
Changed simulcast full-stack tests to be 30fps instead of 50 to better reflect real usecases (expect all kind of perf metrics to improve).
BUG=webrtc:7095
Review-Url: https://codereview.webrtc.org/2730073002
Cr-Commit-Position: refs/heads/master@{#17068}
The check: 'RTC_CHECK_GE(slice_height, height);' has been observed to
fail after a reconfig. It looks like |slice_height| is still using the
previous resolution. |slice_height| isn't used for texture output and
hopefully this issue is texture specific. This CL only extracts and
checks |slice_height| when it's actually used.
BUG=b/35932686
Review-Url: https://codereview.webrtc.org/2736603003
Cr-Commit-Position: refs/heads/master@{#17065}
This will allow the trybots to be updated to start running this new test
executable, so that they can be used when landing this CL which will
replace the dummy test with real code:
https://codereview.webrtc.org/2694203002
Most likely, the trybots will just run the test binary while the perf bots
will run a Python wrapper script that takes care of the post-processing
to calculate audio quality using PESQ.
BUG=webrtc:7229
NOTRY=True
Review-Url: https://codereview.webrtc.org/2717683002
Cr-Commit-Position: refs/heads/master@{#17063}
Reason for revert:
The issue is now hopefully fixed.
Original issue's description:
> Revert of Add QP for FFmpeg H264 decoder. (patchset #4 id:200001 of https://codereview.webrtc.org/2649133007/ )
>
> Reason for revert:
> Let's revert this while we investigate a problem in H264 bitstream parser.
>
> Original issue's description:
> > Add QP for FFmpeg H264 decoder.
> >
> > BUG=webrtc:6541
> >
> > Review-Url: https://codereview.webrtc.org/2649133007
> > Cr-Commit-Position: refs/heads/master@{#16942}
> > Committed: 879f4f6c31
>
> TBR=sprang@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:6541, chromium:697795
>
> Review-Url: https://codereview.webrtc.org/2726973003
> Cr-Commit-Position: refs/heads/master@{#16974}
> Committed: 4c6df8893eTBR=sprang@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6541, chromium:697795
Review-Url: https://codereview.webrtc.org/2735733002
Cr-Commit-Position: refs/heads/master@{#17061}
We currently don't know how to parse these NALus and we don't need
any information from them anyway so we might as well skip parsing them
and not break.
BUG=chromium:697795
Review-Url: https://codereview.webrtc.org/2732623002
Cr-Commit-Position: refs/heads/master@{#17057}
This CL is a reland of https://codereview.webrtc.org/2722423003 which got
reverted due to compile errors when rolling into Chromium.
Original CL description:
Improve testing of SRTP external auth code paths.
Previously code behind ENABLE_EXTERNAL_AUTH was only compiled with Chromium
but developed in WebRTC, which made testing rather complicated. This caused
some trouble in the past (e.g. https://crbug.com/628400#c1)
This CL helps in that the external auth code is now compiled with WebRTC
and the srtpfilter integration gets tested.
BUG=chromium:628400
Review-Url: https://codereview.webrtc.org/2735613002
Cr-Commit-Position: refs/heads/master@{#17052}
After profiling, sakal@ found that this method was taking very long,
and causing the bitstream parsing to take up to 1ms per frame. The
culprit proved to be rtc::Buffer::AppendData, which was called for
every byte and subsequently calls memcpy.
BUG=webrtc:7293
Review-Url: https://codereview.webrtc.org/2728093002
Cr-Commit-Position: refs/heads/master@{#17051}
