Fix some out-of-sync differences between GYP and GN targets for
desktop capture.
Remove sources that aren't used on iOS for that platform, to solve
linking errors that surfaced when flipping iOS to GN by default.
BUG=webrtc:5949
NOTRY=True
TBR=sergeyu@chromium.org
Review-Url: https://codereview.webrtc.org/2289103002
Cr-Commit-Position: refs/heads/master@{#13971}
The invalid condition made the test be included for iOS, which
fails linking.
BUG=webrtc:5949, webrtc:5544
NOTRY=True
Review-Url: https://codereview.webrtc.org/2291023002
Cr-Commit-Position: refs/heads/master@{#13970}
In order to get resource files to be properly packaged into
the .app for a unit test on iOS, the resource files needs
to be listed as sources in a bundle_data target.
BUG=webrtc:5949
NOTRY=True
Review-Url: https://codereview.webrtc.org/2292853002
Cr-Commit-Position: refs/heads/master@{#13968}
Instead of full RtpRtcpImpl takes interface of all functions it needs from it.
Added single function for parsing packets and sending feedback, moving that
logic from RtpRtcpImpl to RtcpReceiver.
BUG=webrtc:5260
Review-Url: https://codereview.webrtc.org/2274573002
Cr-Commit-Position: refs/heads/master@{#13960}
Reason for revert:
Breaks downstream.
Original issue's description:
> Remove the old AndroidVideoCapturer stack code.
>
> This code is no longer needed. Apps should be using the new API introduced here: https://codereview.webrtc.org/2127893002/
>
> Committed: https://crrev.com/1b365a8db070f9cdcbf35ec871f758dcd909e51d
> Cr-Commit-Position: refs/heads/master@{#13950}
TBR=magjed@webrtc.org,glaznev@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/2291583002
Cr-Commit-Position: refs/heads/master@{#13958}
This would make it possible to gather stats on multiple threads, store
the results in multiple reports and to merge the results.
Added rtcstatsreport_unittest.cc, moving a RTCStatsReport-related test
from rtcstats_unittest.cc. Added more unittests covering the order of
stats and TakeMembersFrom.
Also changed RTCStatsReport[] to RTCStatsReport::Get to avoid
confusion with other usages of the [] operator.
BUG=chromium:627816
NOTRY=True
Review-Url: https://codereview.webrtc.org/2278433003
Cr-Commit-Position: refs/heads/master@{#13957}
This adds a new file, webrtc/modules/audio_coding/neteq/tools/packet_source.cc, so that I'll have somewhere to put the new non-inlined methods.
NOTRY=true
BUG=webrtc:163
Review-Url: https://codereview.webrtc.org/2290593002
Cr-Commit-Position: refs/heads/master@{#13956}
2 fixes: When running tool on log with no packet losses, the tool no
longer crashes. When providing relative path to log, the tool no
longer searches in out/target, but instead in current directory.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2291473003
Cr-Commit-Position: refs/heads/master@{#13954}
This is still a tiny lie, since it checks for kCodecArbitrary to avoid
scaling a codec with an external decoder, because of missing information
in that case. The main point is still true, though. Once the next CL is
in, removing NetEqDecoder usage completely in DecoderDatabase, another
workaround will be in place for external decoders until we can fully
deprecate them.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2270063006
Cr-Commit-Position: refs/heads/master@{#13952}
Change the previous GN configs to build GYP instead
(since we'll keep GYP around for a while) but exclude tests
and examples for that config, since we'll only support the production
code for GYP.
Add new configs for upcoming rename of those bots to GYP instead
of GN.
BUG=webrtc:5949
NOTRY=True
Review-Url: https://codereview.webrtc.org/2277253002
Cr-Commit-Position: refs/heads/master@{#13946}
The only real difference between the two is that SetRtcpTransportChannel
had a workaround to prevent a signal from being emitted early.
Basically, in SetTransport, we want to switch the transport channels and
*then* update the state, rather than updating the state after changing
only one transport channel.
But this can be accomplished more easily by simply updating the state in
SetTransport directly.
Review-Url: https://codereview.webrtc.org/2274283004
Cr-Commit-Position: refs/heads/master@{#13945}
Log the DTLS handshake error code in OpenSSLStreamAdapter.
Forward the error code to WebRTCSession with the Signals.
This part is only for the WebRTC native code.
To make it work, need another CL for Chromium.
BUG=webrtc:5959
Review-Url: https://codereview.webrtc.org/2167363002
Cr-Commit-Position: refs/heads/master@{#13940}
Reason for revert:
This caused build breakage due to upstream dependencies.
These dependencies need to be resolved before landing the CL.
Original issue's description:
> This CL adds functionality in the level controller to
> receive a signal level to use initially, instead of the
> default initial signal level.
>
> BUG=
>
> Committed: https://crrev.com/57fec1d828113241186e78710ec5e851cc1a0e81
> Cr-Commit-Position: refs/heads/master@{#13931}
TBR=henrik.lundin@webrtc.org,aleloi@webrtc.org,solenberg@webrtc.org,henrika@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/2283793002
Cr-Commit-Position: refs/heads/master@{#13936}
Change the previous GN configs to build GYP instead
(since we'll keep GYP around for a while) but exclude tests
and examples for that config, since we'll only support the production
code for GYP.
Add new configs for upcoming rename of those bots to GYP instead
of GN.
The Linux32 Debug/Release bots were removed a while back, so
their configs are removed as well.
BUG=webrtc:5949
NOTRY=True
Review-Url: https://codereview.webrtc.org/2277633005
Cr-Commit-Position: refs/heads/master@{#13935}
Since this conversion is used in multiple place and extension seems
right place to keep it in.
BUG=webrtc:1994
NOTRY=true
Review-Url: https://codereview.webrtc.org/2272563010
Cr-Commit-Position: refs/heads/master@{#13934}
the number of points that need to be mocked for testing.
For the now non-virtual methods, DecoderDatabase now does a lookup
through GetDecoderInfo and then delegates to the appropriate method in
the DecoderInfo object, if one is found.
A few other methods were also changed to look up through GetDecoderInfo.
Also moved the audio decoder factory into DecoderInfo, so that
DecoderInfo::GetDecoder can be used directly.
Review-Url: https://codereview.webrtc.org/2276913002
Cr-Commit-Position: refs/heads/master@{#13933}
receive a signal level to use initially, instead of the
default initial signal level.
BUG=
Review-Url: https://codereview.webrtc.org/2254973003
Cr-Commit-Position: refs/heads/master@{#13931}
We have RTC_CHECK and RTC_DCHECK for C now, so we should use it. It's
one fewer difference between our C and C++ code.
NOPRESUBMIT=true
Review-Url: https://codereview.webrtc.org/2274083002
Cr-Commit-Position: refs/heads/master@{#13930}
Unit test would fail in default configuration (e.g. rtc_use_h264=0), cause it tests instantiating H264 specifics.
BUG=webrtc:6194, webrtc:6198
Review-Url: https://codereview.webrtc.org/2228733004
Cr-Commit-Position: refs/heads/master@{#13929}
Checksums were updated for NetEq and ACM bitexactness tests (after
verifying the audio quality).
BUG=webrtc:5447
Review-Url: https://codereview.webrtc.org/2266293005
Cr-Commit-Position: refs/heads/master@{#13928}
There were 3 different meanings for "ReadyToSend", for example, so it
was difficult to understand the meaning at first glance.
Also switching ASSERTs to RTC_DCHECKs.
Review URL: https://codereview.webrtc.org/2269173004 .
Cr-Commit-Position: refs/heads/master@{#13926}
