The codecs expected by HasCorrectCodecs now depends which codecs were
enabled by build flags.
SendSideBweWithOverheadTest.MinAndMaxBitrate now expects different
values for min bitrate depending on if we support 120 ms frames for
Opus.
BUG=b/35415435
Review-Url: https://codereview.webrtc.org/2691343008
Cr-Commit-Position: refs/heads/master@{#16643}
After https://codereview.webrtc.org/2340773002,
the path from webrtc::test::ResourcePath in
webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc is wrong.
It is
/path/to/repos/resources/audio_coding/testfile32kHz.pcm
It should be
/path/to/repos/webrtc-temp/src/resources/audio_coding/testfile32kHz.pcm.
The middle part is missing.
The reason this target is affected is because
webrtc::test::SetExecutablePath(argv[0]);
was not called.
That call is necessary for us to know that the test is being run from src/
and not from out/Default (as is assumed, when that function is not called.)
BUG=chromium:497757
R=kjellander@webrtc.org, henrik.lundin@webrtc.org
Review-Url: https://codereview.webrtc.org/2698743002
Cr-Commit-Position: refs/heads/master@{#16641}
The frame_analyzer which is used by compare_videos.py needs to handle
barcode errors. As before the reference and the test video can contain
repeated frames. When there are barcode decode errors in the test video,
then we should not let that contribute to the skipped frames score. When
there are barcode decode errors in the reference video, then we need to
take proper care to still calculate skipped barcodes when the
corresponding frames are not present in the test video and the test
video does not have a frame with a barcode decode error that could have
been the same frame as the one in the reference.
A new metric total number of skipped frames and for number of decode
errors is introduced. Barcodes that appears in the test video, but not
in the reference video are also listed.
BUG=webrtc:6967
Review-Url: https://codereview.webrtc.org/2666333003
Cr-Commit-Position: refs/heads/master@{#16638}
Previously, was only checking the Android SDK version. But it also needs
to check for the presence of the connectivity manager service.
BUG=webrtc:7026
Review-Url: https://codereview.webrtc.org/2697943002
Cr-Commit-Position: refs/heads/master@{#16631}
No point in measuring the time needed to write dropped frames to disk.
BUG=webrtc:6634
Review-Url: https://codereview.webrtc.org/2696503003
Cr-Commit-Position: refs/heads/master@{#16629}
For them implemeted upscaling in libyuv metrics calculation.
Updated maximum number of SL in vp9 encoder to 3.
Refactored names of some fields in Video_quality_check analyzer.
BUG=webrtc:7095
Review-Url: https://codereview.webrtc.org/2681683003
Cr-Commit-Position: refs/heads/master@{#16625}
Add explicit conversions to/from uint64_t
uint64_t is natural type for NtpTime, including wrap on overflow.
BUG=None
Review-Url: https://codereview.webrtc.org/2695683002
Cr-Commit-Position: refs/heads/master@{#16624}
The removed tests are covered by cases in call_perf_tests.
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/2672583002
Cr-Commit-Position: refs/heads/master@{#16621}
Reason for revert:
Breaks AppRTCMobile interoperability. The ICE candidate URL shouldn't be signaled between endpoints, it's only there for informational purposes.
Original issue's description:
> Add the url attribute to the IceCandidate (Java Wrapper)
>
> The url of the ICE server is added to the IceCandiate class.
> This can be used to tell which server this candidate was gathered from.
>
> BUG=webrtc:7128
>
> Review-Url: https://codereview.webrtc.org/2690593002
> Cr-Original-Commit-Position: refs/heads/master@{#16593}
> Committed: 8586c8ee88
> Review-Url: https://codereview.webrtc.org/2690593002
> Cr-Commit-Position: refs/heads/master@{#16615}
> Committed: 45efce01c7TBR=magjed@webrtc.org,zhihuang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7128
Review-Url: https://codereview.webrtc.org/2699533002
Cr-Commit-Position: refs/heads/master@{#16616}
The url of the ICE server is added to the IceCandiate class.
This can be used to tell which server this candidate was gathered from.
BUG=webrtc:7128
Review-Url: https://codereview.webrtc.org/2690593002
Cr-Original-Commit-Position: refs/heads/master@{#16593}
Committed: 8586c8ee88
Review-Url: https://codereview.webrtc.org/2690593002
Cr-Commit-Position: refs/heads/master@{#16615}
The flaky test was introduced in ad9010c9836, and is essentially a race
where the ViE Encoder has already configured the quality scaler on the
encoder thread before we've updated the ScalingSettings. This CL adds
a forced reconfiguration of the quality scaler to avoid this issue.
BUG=None
TBR=sprang@webrtc.org
Review-Url: https://codereview.webrtc.org/2695873004
Cr-Commit-Position: refs/heads/master@{#16612}
This enables tighter integration with XCode tooling and is a prereq
for adding UI tests.
BUG=webrtc:7150
Review-Url: https://codereview.webrtc.org/2697603002
Cr-Commit-Position: refs/heads/master@{#16609}
The test program modules/audio_processing/test/audioproc_float.cc
defined the flag 'agc_compression_gain' and had checks if the
parameter was valid (audioproc_float). The flag was also copied to
webrtc::test::SimulationSettings of audio_processing_simulator.h. The
setting was however never applied to APM.
This change applies the setting on the GainControl submodule in the
same way as the agc_target_level is applied.
This is needed for e.g. testing the AGC fixed digital limiter with the
same configuration as it is (currently) used with in AudioMixerImpl.
Also added new flag '-experimental_agc'. This flag allows disabling the
experimental AGC, which is how the AGC is used in AudioMixerImpl.
ExperimentalAgc is enabled by default, exactly as it was prior to this change.
The change has been tested locally by listening tests and diff comparisons.
BUG=None
NOTRY=True # win_dbg bot not cooperating
Review-Url: https://codereview.webrtc.org/2684983004
Cr-Commit-Position: refs/heads/master@{#16603}
The files socketpool.h and diskcache.h also become unused, and are
deleted together with their sources.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2694753002
Cr-Commit-Position: refs/heads/master@{#16601}
downstream application depends on it.
Mark the old Port::AddAddress deprecated and will be removed after the
applications stop replying on it.
BUG=None.
R=deadbeef@webrtc.org
Review-Url: https://codereview.webrtc.org/2694103003 .
Cr-Commit-Position: refs/heads/master@{#16598}
If android_setsocknetwork() is available, and it fails, then bind()
should *not* be called, and an error should be returned.
If it succeeds, then bind should be called, but with an "any" address.
This is to prevent cases where sockets are sent with a source address
that doesn't match the network interface they're sent on. See bug below.
This CL also changes "NetworkBinderResults" to an enum class, and
renames it to "NetworkBinderResult".
BUG=webrtc:7026
Review-Url: https://codereview.webrtc.org/2646863005
Cr-Commit-Position: refs/heads/master@{#16597}
To ensure compliance with older version high profile should appear in local SDP
before baseline profile.
BUG=b/34816463
Review-Url: https://codereview.webrtc.org/2696733002
Cr-Commit-Position: refs/heads/master@{#16596}
Reason for revert:
Breaks downstream application's build
Original issue's description:
> Add the url attribute to the IceCandidate (Java Wrapper)
>
> The url of the ICE server is added to the IceCandiate class.
> This can be used to tell which server this candidate was gathered from.
>
> BUG=webrtc:7128
>
> Review-Url: https://codereview.webrtc.org/2690593002
> Cr-Commit-Position: refs/heads/master@{#16593}
> Committed: 8586c8ee88TBR=magjed@webrtc.org,zhihuang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7128
Review-Url: https://codereview.webrtc.org/2692993002
Cr-Commit-Position: refs/heads/master@{#16595}
Makes sure video decoder software fallback handles InitDecode()
failures, and properly releases the pointer after ::Release() so that
another decode failure will properly reinitialize the decoder.
Also makes sure to not call Decode() without a previous InitDecode()
succeeding.
BUG=webrtc:7154
R=noahric@chromium.org, sophiechang@chromium.org
Review-Url: https://codereview.webrtc.org/2690183004 .
Cr-Commit-Position: refs/heads/master@{#16594}
The url of the ICE server is added to the IceCandiate class.
This can be used to tell which server this candidate was gathered from.
BUG=webrtc:7128
Review-Url: https://codereview.webrtc.org/2690593002
Cr-Commit-Position: refs/heads/master@{#16593}
The URL of the ICE server will be reconstructed by the Port and the URL
attribute is added to the cricket::Candidate struct so that we can tell
which ICE server the candidate was gathered from.
This CL only changes the native C++ code. The Java and Objc wrapper will
be created in separate CLs.
BUG=webrtc::7128
Review-Url: https://codereview.webrtc.org/2685053004
Cr-Commit-Position: refs/heads/master@{#16591}
Lateness is determined by the length of the send-side history, currently
set to 60 seconds.
BUG=webrtc:5079
Review-Url: https://codereview.webrtc.org/2684353004
Cr-Commit-Position: refs/heads/master@{#16588}
This avoids redoing RTP header parsing already done in Call, for video.
The next step is to convert other types of receive streams, i.e.,
audio and flexfec, to use a compatible OnRtpPacket method. We can then
introduce a shared base interface, and simplify media-independent
receive processing in Call.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2681673004
Cr-Commit-Position: refs/heads/master@{#16583}
The only implementation which used a nullptr was a mock used in tests,
so add a dummy instance there instead.
Remove tests for stats_proxy_ in vie_encoder and just dcheck in the
constructor instead.
BUG=None
Review-Url: https://codereview.webrtc.org/2695643002
Cr-Commit-Position: refs/heads/master@{#16577}
SslSocketFactory is unused since https://codereview.webrtc.org/2506983002, and it's the last
user of AutoDetectProxy.
Also move HttpListenServer and SocksProxyServer to the rtc_base_tests_utils gn target, since they're used by tests only.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2541453002
Cr-Commit-Position: refs/heads/master@{#16576}
Updated comment.
Don't call AdaptUp/AdaptDown in tests without first emitting a frame.
Handle frame received precondition in AdaptUp/AdaptDown with DCHECK
instead of return.
BUG=webrtc:4172, webrtc:6850
Review-Url: https://codereview.webrtc.org/2690023002
Cr-Commit-Position: refs/heads/master@{#16572}
Other minor changes:
- Define locks after stuff it is protecting
- Use explicit default dtors
- Replace unnecessary lock in DelayedEncoder with SequencedTaskChecker
BUG=webrtc:7130
Review-Url: https://codereview.webrtc.org/2686103002
Cr-Commit-Position: refs/heads/master@{#16554}
There are currently two webrtc::CircularBuffers defined:
- modules/audio_coding/test/utility.{h,cc}
- modules/audio_processing/echo_detector/circular_buffer.{h,cc}
This CL moves the former definition to the webrtc::test namespace,
to avoid link errors in a future build target.
BUG=None
Review-Url: https://codereview.webrtc.org/2667383008
Cr-Commit-Position: refs/heads/master@{#16553}
"bundle_transport_name" is no longer relevant here, and
"rtcp_mux_required" is implied by whether or not an RTCP transport is
passed in.
BUG=None
Review-Url: https://codereview.webrtc.org/2689503002
Cr-Commit-Position: refs/heads/master@{#16551}
