Reason for revert:
Downstream apps should now be fixed.
Original issue's description:
> Revert of Remove the old AndroidVideoCapturer stack code. (patchset #2 id:20001 of https://codereview.webrtc.org/2235893003/ )
>
> 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
>
> Committed: https://crrev.com/e39f251dacf66e50153bcda615f06b7c59e5856b
> Cr-Commit-Position: refs/heads/master@{#13958}
TBR=magjed@webrtc.org,glaznev@webrtc.org,kjellander@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
Review-Url: https://codereview.webrtc.org/2298063003
Cr-Commit-Position: refs/heads/master@{#13988}
Also, make explicit that the GYP build has limited support.
BUG=webrtc:6281
NOTRY=True
Review-Url: https://codereview.webrtc.org/2298143002
Cr-Commit-Position: refs/heads/master@{#13984}
This is step 2 of the plan below.
The modified plan
1. First land unmodified task_queue.h into webrtc_override in Chrome
2. Modify build files in the webrtc repo to include the task_queue.h and task_queue.cc from webrtc_overrides. This will breaks webrtc Chrome FYI.
3. Combine a roll of webrtc to Chrome and a the cl in https://codereview.chromium.org/2293913003/ into one cl.
The combined cl will roll in build files from 2 and add task_queue.cc in webrtc_overrides and build task_queue_unittest.cc as part of content_unittests to test task_queue.cc in webrtc_overrides.
4... Start using task queues in webrtc........
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/2289203002
Cr-Commit-Position: refs/heads/master@{#13983}
hbos and hta are webrtc/stats/ OWNERS. Public api headers relating to
rtcstats are placed in webrtc/api/ and implementations are placed in
webrtc/stats/. This ownership allows the rtcstats owners to own both .cc
and .h files.
For example, rtcstats.[h/cc] and rtcstatsreport.[h/cc].
(Soon there will also be rtcstats_objects.[h/cc] and more.)
BUG=chromium:627816
Review-Url: https://codereview.webrtc.org/2294693002
Cr-Commit-Position: refs/heads/master@{#13981}
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
TBR=ehmaldonado@webrtc.org
Review-Url: https://codereview.webrtc.org/2291443002
Cr-Commit-Position: refs/heads/master@{#13980}
This is the stats collector for the new stats types, RTCStats[1] and
RTCStatsReport[2]. It so far only produces RTCPeerConnectionStats[3] as
an example of how it would collect stats. Each RTCStats subclass will
get a corresponding RTCStatsCollector::ProduceFooStats().
Stats reports are cached and returned as const references (ref
counting). This allows stats to be inspected by multiple observers and
across multiple threads. No copies will have to be made when surfacing
this to Blink or other places.
The current implementation of ProducePeerConnectionStats() only look at
existing DataChannels. This might be incorret if data channels can be
removed? Will investigate in a follow-up, crbug.com/636818.
[1] https://www.w3.org/TR/2016/WD-webrtc-20160531/#idl-def-rtcstats
[2] https://www.w3.org/TR/2016/WD-webrtc-20160531/#rtcstatsreport-object
[3] https://w3c.github.io/webrtc-stats/archives/20160526/webrtc-stats.html#pcstats-dict*
BUG=chromium:627816, chromium:636818
Review-Url: https://codereview.webrtc.org/2242043002
Cr-Commit-Position: refs/heads/master@{#13979}
when building the code for ARM.
The intention is to follow up this CL with other CLs that
further addresses the internal resampling in APM
BUG=webrtc:6181
Review-Url: https://codereview.webrtc.org/2265473003
Cr-Commit-Position: refs/heads/master@{#13974}
GetCopyWithRotationApplied is not yet deleted; downstream projects
must be updated first.
BUG=webrtc:5682
Review-Url: https://codereview.webrtc.org/2285693002
Cr-Commit-Position: refs/heads/master@{#13973}
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}
