This changes added a simple measurement of levels "close to the audio hardware"
both for playout and for recording. These levels are logged once each 10 seconds.
It also adds WebRTC.Audio.RecordedOnlyZeros UMA stat and it is updated at
destuction. It will report true iff all reported recording leves are zero.
BUG=NONE
R=peah@webrtc.org
Review URL: https://codereview.webrtc.org/2328433003 .
Cr-Commit-Position: refs/heads/master@{#14160}
In order to be able to clear out any potentially stashed old frames from
the RtpFrameReferenceFinder we can now clear frames that contain packets
older than |seq_num|.
BUG=webrtc:5514
Review-Url: https://codereview.webrtc.org/2304723004
Cr-Commit-Position: refs/heads/master@{#14156}
The right thing to do would be to ensure that the multiplication can't
overflow, but that'd be a major change bordering on a rewrite, and
would take too much time. So instead, we just instruct UBSan to look
the other way.
BUG=chromium:634834
Review-Url: https://codereview.webrtc.org/2318083002
Cr-Commit-Position: refs/heads/master@{#14154}
This will unblock planned work on libsrtp and reduce the exposed GYP surface.
BUG=webrtc:5949
NOTRY=True
Review-Url: https://codereview.webrtc.org/2314473002
Cr-Commit-Position: refs/heads/master@{#14152}
After the Android test framework in Chromium moved over to generated
scripts, these paths are no longer used, so let's clean them up.
BUG=webrtc:6323
NOTRY=True
Review-Url: https://codereview.webrtc.org/2327013002
Cr-Commit-Position: refs/heads/master@{#14151}
This is to avoid a naming conflict with webrtc::RTCStatsReport that is
surfaced if you try to include it in peerconnectioninterface.h.
Background: The current stats is very much non-spec-compliant. A new
stats collection API is underway that is meant to be spec-compliant.
Some classes in Chromium and webrtc/sdk/objc have spec-compliant names
but non-spec-compliant behavior. These are being renamed to "Legacy" so
that new spec-compliant classes can be added with the correct names.
BUG=chromium:627816
TBR=tkchin@webrtc.org
NOTRY=True
Review-Url: https://codereview.webrtc.org/2313943002
Cr-Commit-Position: refs/heads/master@{#14150}
I have written a large part of the code in these files and I feel I
should be an OWNER of them.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2322983002
Cr-Commit-Position: refs/heads/master@{#14148}
Integrate AvgCounter to be used for BWE stats in call.
Fixes for stats regression in:
WebRTC.Call.EstimatedSendBitrateInKbps
WebRTC.Call.PacerBitrateInKbps
Example:
BWE for a 15 seconds long call (with intervals of 1 sec):
|300|400|500|600|600|600|600| 0 | 0 | 0 | 0 | 0 |800|800|800| // 0 - network state down
Reported via OnNetworkChanged:
|300|400|500|600| x | x | x | 0 | x | x | x | x |800| x | x | // x - empty interval, 0 -> pauses stats
Stats:
|300|400|500|600|600|600|600| - | - | - | - | - |800|800|800| // x -> last value used (intervals during pause ignored)
AvgCounter uses the average of samples within an interval (interval length is 2 sec).
BUG=webrtc:6244
Review-Url: https://codereview.webrtc.org/2307913002
Cr-Commit-Position: refs/heads/master@{#14147}
Only has counts stats right now but enumeration stats can easily be added in the future if needed.
BUG=webrtc:6313
Review-Url: https://codereview.webrtc.org/2320473002
Cr-Commit-Position: refs/heads/master@{#14146}
Reason for revert:
Unfortunately this didn't help, so I'm reverting this to clean things up for now.
Original issue's description:
> Add a DEPS gclient hook to prune corrupt mockito remote.
>
> A third_party remote for third_party/mockito/src is corrupt. Run
> `git remote prune origin` in the checkout directory before syncing
> Chromium.
>
> Thanks to slan@ for providing this fix.
>
> TBR=kjellander@webrtc.org
> BUG=chromium:644722
> NOTRY=True
>
> Committed: https://crrev.com/3115b06957d5b8cba5d45cebc301795e94a926ec
> Cr-Commit-Position: refs/heads/master@{#14138}
TBR=ehmaldonado@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:644722
Review-Url: https://codereview.webrtc.org/2326473003
Cr-Commit-Position: refs/heads/master@{#14143}
the AEC. This solves the following issues:
-Even though the buffering was previously done using ringbuffers, those
were inefficiently used which caused a lot of hidden memcopys.
-The ringbuffers wasted a lot of space in the AEC state as they were too
long.
-The lowest and two upper bands were decoupled in the buffering, which
required extra code to handle.
-On top of the ringbuffers there was a second linear buffer that was
stored in the state which caused even more data to be stored on the
state.
-The incoming nearend frames were passed to the functions in the form
of buffers on the state, which made the code harder to read as it was
not immediately clear where the nearend signal was used, and when it
was modified.
The CL addresses this by replacing all the buffers by two linear buffers:
-One buffer before the AEC processing for producing nearend
blocks of size 64 that can be processed by the AEC.
-One inside the AEC processing that buffers the current
nearend block until the next block is processed.
The changes have been tested to be bitexact.
This CL will be followed by several other CLs, that refactor the other
buffers in the AEC.
BUG=webrtc:5298, webrtc:6018
Review-Url: https://codereview.webrtc.org/2311833002
Cr-Commit-Position: refs/heads/master@{#14141}
This is a simple refactoring of VideoQualityTest. It will help in adding audio related settings to VideoQualityTest.
BUG=
Review-Url: https://codereview.webrtc.org/2312613003
Cr-Commit-Position: refs/heads/master@{#14139}
A third_party remote for third_party/mockito/src is corrupt. Run
`git remote prune origin` in the checkout directory before syncing
Chromium.
Thanks to slan@ for providing this fix.
TBR=kjellander@webrtc.org
BUG=chromium:644722
NOTRY=True
Review-Url: https://codereview.webrtc.org/2326523002
Cr-Commit-Position: refs/heads/master@{#14138}
Reason for revert:
Breaks Chromium FYI bots because of link issue
Original issue's description:
> Don't use VoE legacy APIs in force_mic_volume_max tool.
>
> BUG=webrtc:4690
>
> Committed: https://crrev.com/ae9f2bdcef7f3a338ece6f57744c8c8f74d15483
> Cr-Commit-Position: refs/heads/master@{#14135}
TBR=henrika@webrtc.org,phoglund@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/2320113002
Cr-Commit-Position: refs/heads/master@{#14137}
Right now, RuntimeExceptions thrown in an invoke function will kill the handler thread. This CL propagates the exceptions to the calling thread instead, to give the caller a change to handle them.
BUG=webrtc:6327
Review-Url: https://codereview.webrtc.org/2316223002
Cr-Commit-Position: refs/heads/master@{#14132}
Reason for revert:
Broke all Chromium libFuzzer builds
https://bugs.chromium.org/p/chromium/issues/detail?id=645069
Original issue's description:
> Setting up an RTP input fuzzer for NetEq
>
> This CL introduces a new fuzzer target neteq_rtp_fuzzer that
> manipulates the RTP header fields before inserting the packets into
> NetEq. A few helper classes are also introduced.
>
> BUG=webrtc:5447
> NOTRY=True
>
> Committed: https://crrev.com/2d273f1e97cd5030ed1686f27ce1118291b66395
> Cr-Commit-Position: refs/heads/master@{#14103}
TBR=ivoc@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:5447
Review-Url: https://codereview.webrtc.org/2328483002
Cr-Commit-Position: refs/heads/master@{#14131}
I could not find a single place it was used, outside of the unittests
for the sync packet support itself.
Review-Url: https://codereview.webrtc.org/2309303002
Cr-Commit-Position: refs/heads/master@{#14130}
libjingle_peerconnection_so is not including common_config, which is
causing some differences is the defines.
We'd like to prevent that happening in the future.
NOTRY=True
BUG=webrtc:5949
Review-Url: https://codereview.webrtc.org/2325603002
Cr-Commit-Position: refs/heads/master@{#14127}
Reason for revert:
Downstream build is fixed.
Original issue's description:
> Revert of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #3 id:80001 of https://codereview.webrtc.org/2280703002/ )
>
> Reason for revert:
> Breaks downstream build.
>
> Original issue's description:
> > Ignore Camera and Flip bits in CVO when parsing video rotation
> >
> > Currently, if WebRTC receives a CVO byte where the Camera or Flip bit is
> > set, then rotation is incorrectly parsed as 0. This CL fixes that issue.
> > The Camera and Flip bit is still unimplemented and will just be ignored
> > though.
> >
> > BUG=webrtc:6120
> > R=danilchap@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org
> >
> > Committed: f9e1b922ef
>
> TBR=pthatcher@webrtc.org,danilchap@webrtc.org,tommi@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6120
>
> Committed: https://crrev.com/97667c7746282704acccd896e26175decee349c0
> Cr-Commit-Position: refs/heads/master@{#14035}
TBR=pthatcher@webrtc.org,danilchap@webrtc.org,tommi@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6120
Review-Url: https://codereview.webrtc.org/2320913003
Cr-Commit-Position: refs/heads/master@{#14124}
With this CL, the NetEqReplacementInput class handles reordered and
missing packets in a better way than before, by storing the last
confirmed packet size and using that when the next packet size cannot
be calculated.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2319553003
Cr-Commit-Position: refs/heads/master@{#14122}
1. Use of const in all variable declarations where it is possible
2. Variable names and function arguments changed from CamelCase to match code style
3. A few stale comments removed.
4. Chromium clang plugin check added (now possible thanks to kwiberg@'s work on common.h)
5. Disallow constructor macros added.
NOTRY=true
Review-Url: https://codereview.webrtc.org/2294263002
Cr-Commit-Position: refs/heads/master@{#14120}
"WebRTC.Call.BitrateReceivedInKbps"
"WebRTC.Call.VideoBitrateReceivedInKbps"
"WebRTC.Call.AudioBitrateReceivedInKbps"
"WebRTC.Call.RtcpBitrateReceivedInBps"
Reports the average of periodically computed bitrates over a call (2 sec interval). Adds possibility to later modify the stats and use/report the periodic bitrates.
BUG=webrtc:5283
Review-Url: https://codereview.webrtc.org/2303763002
Cr-Commit-Position: refs/heads/master@{#14119}
Avoids reporting a growing delay (i.e. time b/w current time and oldest packet in the pacer).
BUG=webrtc:6253
Review-Url: https://codereview.webrtc.org/2279283002
Cr-Commit-Position: refs/heads/master@{#14118}
RecreateWebRtcStream was checking the "sending_" flag, but wasn't
checking rtp_parameters_.encodings[0].active. As a result, if an
application calls "RtpSender.setParameters(inactive_params)" then later
the stream is recreated due to some change in SDP parameters, the stream
would incorrectly start sending.
R=pthatcher@webrtc.org, skvlad@webrtc.org
Review URL: https://codereview.webrtc.org/2246313002 .
Cr-Commit-Position: refs/heads/master@{#14116}
AudioNetworkAdaptor is supposed to facilitate AudioEncoder to adapt to varying network conditions.
This is the first of a sequence of CLs that are to add one implementation of AudioNetworkAdaptor.
This CL illustrates the interfaces of the AudioNetworkAdaptor.
BUG=webrtc:6303
Review-Url: https://codereview.webrtc.org/2308573002
Cr-Commit-Position: refs/heads/master@{#14115}
Reason for revert:
Resubmit capturer tests
Original issue's description:
> Revert of [WebRTC] A real ScreenCapturer test (patchset #8 id:240001 of https://codereview.webrtc.org/2268093002/ )
>
> Reason for revert:
> ScreenCapturerTest.CaptureUpdatedRegion fails on Win DrMemory Full.
>
> Original issue's description:
> > [WebRTC] A real ScreenCapturer test
> >
> > We do not have a real ScreenCapturer test before. And after CL 2210443002, a new
> > ScreenDrawer interface is added to the code base to draw various shapes on the
> > screen. This change is to use ScreenDrawer to test ScreenCapturer. Besides test
> > cases, some other changes are included,
> >
> > 1. A WaitForPendingPaintings() function in ScreenDrawer, to wait for a
> > ScreenDrawer to finish all the pending draws. This function now only sleeps 50
> > milliseconds on X11 and 100 milliseconds on Windows.
> >
> > 2. A Color structure to help handle a big-endian or little-endian safe color and
> > provide functions to compare with DesktopFrame::data(). Both ScreenDrawer and
> > DesktopFrameGenerator (in change 2202443002) can use this class to create colors
> > and compare with or paint to a DesktopFrame.
> >
> > 3. ScreenDrawer now uses Color structure instead of uint32_t.
> >
> > BUG=314516
> >
> > TBR=kjellander@chromium.org
> >
> > Committed: https://crrev.com/9d1c54ace0dc9f68da0152aa1ded2a8dba0a43ae
> > Cr-Commit-Position: refs/heads/master@{#14058}
>
> TBR=sergeyu@chromium.org,jamiewalch@chromium.org,kjellander@chromium.org,zijiehe@chromium.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=314516
>
> Committed: https://crrev.com/4c44202dc348613695a4b529bbd7c9bdab6195ec
> Cr-Commit-Position: refs/heads/master@{#14071}
TBR=sergeyu@chromium.org,jamiewalch@chromium.org,kjellander@chromium.org,asapersson@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=644130
Review-Url: https://codereview.webrtc.org/2313653003
Cr-Commit-Position: refs/heads/master@{#14113}
Methods are named more consistently and have a more consistent
signatures. The call structure of mixing is slightly
simplified. Anonymous participants are also ramped up.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2298163002
Cr-Commit-Position: refs/heads/master@{#14110}
This file defines webrtc::Config which was mostly used by modules/audio_processing. The files webrtc/common.h, webrtc/common.cc and webrtc/test/common_unittests.cc are moved to modules/audio_processing and the few remaining uses of webrtc::Config are replaced with simpler code.
- For NetEq and pacing configuration, a VoEBase::ChannelConfig is passed to VoEBase::CreateChannel().
- Removes the need for VoiceEngine::Create(const Config& config). No need to store the webrtc::Config in VoE shared state.
BUG=webrtc:5879
Review-Url: https://codereview.webrtc.org/2307533004
Cr-Commit-Position: refs/heads/master@{#14109}
There is no clear reason to have them in build_overrides, and
webrtc/build seems to be a better place.
Also, delete build_overrides/webrtc.gni
NOTRY=True
BUG=webrtc:5949
Review-Url: https://codereview.webrtc.org/2309253004
Cr-Commit-Position: refs/heads/master@{#14108}
