I decided to make one webrtc/sdk/android/BUILD.gn for both tests and Java/jni src.
External dependencies needs to be updated after this CL.
Future work is required to clean up the Android api and move
implementation details to /webrtc/sdk/android/src.
BUG=webrtc:5882,webrtc:6804
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2547483003
Cr-Commit-Position: refs/heads/master@{#15443}
Reason for revert:
The downstream problem is now fixed, and this should be good to land again.
Original issue's description:
> Revert of APM: Change 3 UMA metrics to fewer but linearly distributed buckets (patchset #2 id:20001 of https://codereview.webrtc.org/2547593002/ )
>
> Reason for revert:
> Breaks down-stream dependencies.
>
> Original issue's description:
> > APM: Change 3 UMA metrics to fewer but linearly distributed buckets
> >
> > In this change WebRTC.Audio.ApmCaptureInputLevel{Average,Peak} are
> > changed to 64 buckets, while WebRTC.Audio.AgcLevel is changed to 50
> > buckets. All three are changed to have linear spacing between buckets.
> >
> > Also, the metrics are renamed to avoid stats conflicts because of different bucket schemes:
> > - WebRTC.Audio.AgcLevel -> WebRTC.Audio.AgcSetLevel
> > - WebRTC.Audio.ApmCaptureInputLevelAverage -> WebRTC.Audio.ApmCaptureInputLevelAverageRms
> > - WebRTC.Audio.ApmCaptureInputLevelPeakRms -> WebRTC.Audio.ApmCaptureInputLevelPeakRms
> >
> > BUG=webrtc:6622
> > CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_rel_ng
> >
> > Committed: https://crrev.com/49715fe3be17d8579586d5bc954d626126d53415
> > Cr-Commit-Position: refs/heads/master@{#15418}
>
> TBR=peah@webrtc.org,rkaplow@chromium.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6622
>
> Committed: https://crrev.com/63407a9b6ae6f3fc096e01d64e46c6d21d86b517
> Cr-Commit-Position: refs/heads/master@{#15420}
TBR=peah@webrtc.org,rkaplow@chromium.org
BUG=webrtc:6622
CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_rel_ng
Review-Url: https://codereview.webrtc.org/2551863003
Cr-Commit-Position: refs/heads/master@{#15442}
Reason for revert:
Failures on the Linux Memcheck bot
Original issue's description:
> This approach passes packetization mode to the encoder as part of
> a cricket::VideoCodec structure, rather than as part of struct VideoCodecH264 inside webrtc::VideoCodec.
>
> BUG=600254
>
> Committed: https://crrev.com/e59647b991f61cf1cf61b020356705e6c0f81257
> Cr-Commit-Position: refs/heads/master@{#15437}
TBR=hbos@webrtc.org,sprang@webrtc.org,mflodman@webrtc.org,magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=600254
Review-Url: https://codereview.webrtc.org/2558453002
Cr-Commit-Position: refs/heads/master@{#15441}
flaky, with many false results and with a huge
variance.
This CL addresses that by changing the way the
API call durations are measured, using a warmup
period and a longer interval for computing the
timing estimates.
Furthermore, this CL reduces the number of tests
to compensate for the fact that the tests now are
more expensive, as well as to reduce the number
of regressions further.
BUG=webrtc:6614,webrtc:6685,666725
Review-Url: https://codereview.webrtc.org/2549403002
Cr-Commit-Position: refs/heads/master@{#15440}
a cricket::VideoCodec structure, rather than as part of struct VideoCodecH264 inside webrtc::VideoCodec.
BUG=600254
Review-Url: https://codereview.webrtc.org/2528343002
Cr-Commit-Position: refs/heads/master@{#15437}
The Bazel build format doesn't support having separate
lists of compilation flags for C and C++; it just has a single
copts list for cc_library:
https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts
This makes it hard to convert our GN targets to Bazel when there are
compiler warnings that aren't supported for C (like -Woverloaded-virtual
being added in bugs.webrtc.org/6653).
The solution for this is to move all .c files to their own targets
and remove C++-only compiler flags during conversion.
New targets:
//webrtc/common_audio:common_audio_c
//webrtc/common_audio:common_audio_neon_c
//webrtc/modules/audio_coding:g711_c
//webrtc/modules/audio_coding:g722_c
//webrtc/modules/audio_coding:ilbc_c
//webrtc/modules/audio_coding:isac_c
//webrtc/modules/audio_coding:isac_fix_c
//webrtc/modules/audio_coding:isac_test_util
//webrtc/modules/audio_coding:pcm16b_c
//webrtc/modules/audio_coding:webrtc_opusj_c
//webrtc/modules/audio_device:mac_portaudio
//webrtc/modules/audio_procssing:audio_processing_c
//webrtc/modules/audio_procssing:audio_processing_neon_c
This CL also adds a PRESUBMIT.py check that will throw an error
if targets are mixing .c and .cc files, to preven this from regressing.
BUG=webrtc:6653
NOTRY=True
Review-Url: https://codereview.webrtc.org/2550563003
Cr-Commit-Position: refs/heads/master@{#15433}
This change enables experimentation with the clipping minimum level
parameter in the gain control.
BUG=webrtc:6622
CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_rel_ng
Review-Url: https://codereview.webrtc.org/2543753006
Cr-Commit-Position: refs/heads/master@{#15426}
Reason for revert:
Breaks down-stream dependencies.
Original issue's description:
> APM: Change 3 UMA metrics to fewer but linearly distributed buckets
>
> In this change WebRTC.Audio.ApmCaptureInputLevel{Average,Peak} are
> changed to 64 buckets, while WebRTC.Audio.AgcLevel is changed to 50
> buckets. All three are changed to have linear spacing between buckets.
>
> Also, the metrics are renamed to avoid stats conflicts because of different bucket schemes:
> - WebRTC.Audio.AgcLevel -> WebRTC.Audio.AgcSetLevel
> - WebRTC.Audio.ApmCaptureInputLevelAverage -> WebRTC.Audio.ApmCaptureInputLevelAverageRms
> - WebRTC.Audio.ApmCaptureInputLevelPeakRms -> WebRTC.Audio.ApmCaptureInputLevelPeakRms
>
> BUG=webrtc:6622
> CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_rel_ng
>
> Committed: https://crrev.com/49715fe3be17d8579586d5bc954d626126d53415
> Cr-Commit-Position: refs/heads/master@{#15418}
TBR=peah@webrtc.org,rkaplow@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6622
Review-Url: https://codereview.webrtc.org/2548333002
Cr-Commit-Position: refs/heads/master@{#15420}
In this change WebRTC.Audio.ApmCaptureInputLevel{Average,Peak} are
changed to 64 buckets, while WebRTC.Audio.AgcLevel is changed to 50
buckets. All three are changed to have linear spacing between buckets.
Also, the metrics are renamed to avoid stats conflicts because of different bucket schemes:
- WebRTC.Audio.AgcLevel -> WebRTC.Audio.AgcSetLevel
- WebRTC.Audio.ApmCaptureInputLevelAverage -> WebRTC.Audio.ApmCaptureInputLevelAverageRms
- WebRTC.Audio.ApmCaptureInputLevelPeakRms -> WebRTC.Audio.ApmCaptureInputLevelPeakRms
BUG=webrtc:6622
CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_rel_ng
Review-Url: https://codereview.webrtc.org/2547593002
Cr-Commit-Position: refs/heads/master@{#15418}
This push decision if Marker bit should be set into packetizers fixing
issue where returned last_packet flag was ambiguous for some VP9 packets.
Added test for VP9 where last_packet != marker_bit
BUG=webrtc:6723
Review-Url: https://codereview.webrtc.org/2522553002
Cr-Commit-Position: refs/heads/master@{#15415}
This CL is in preparation to move the AudioFrame into webrtc/api. The
AudioFrame is a POD type used for representing 10ms of audio. It
appears as a parameter and return value of interfaces being migrated
to webrtc/api, in particular AudioMixer.
Here, methods operator+=, operator>>=, Mute are
moved into a new target webrtc/audio/utility/audio_frame_operations,
and dependencies are changed to use
the new versions. The old AudioFrame methods are marked deprecated.
The audio frame utilities in webrtc/modules/utility:audio_frame_operations
are also moved to the new location.
TBR=kjellander@webrtc.org
BUG=webrtc:6548
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2424173003
Cr-Commit-Position: refs/heads/master@{#15413}
https://codereview.webrtc.org/2504023002 broke exponential probing.
After that change ProbeController stops exponential probes prematurely:
it goes to kProbingComplete state if SetEstimatedBitrate() is called
with bitrate lower than min_bitrate_to_probe_further_bps_, which always
happens with the first pair of probes. As result it wasn't sending
repeated probes as it should. This change fixes that issue by moving
probe expieration logic to ProbeContoller::Process(). This also ensures
that the controller goes to kProbingComplete state as soon as probing
timeout expired, without waiting for the next SetEstimatedBitrate()
call.
BUG=669421
Review-Url: https://codereview.webrtc.org/2546613003
Cr-Commit-Position: refs/heads/master@{#15392}
Error resilience is currently always enabled for VP9 which reduces quality.
BUG=webrtc:6783
Review-Url: https://codereview.webrtc.org/2532053002
Cr-Commit-Position: refs/heads/master@{#15390}
Before this the BWE was allowed to operate freely up to 100 kbps. This isn't a good idea for audio BWE.
BUG=webrtc:5079
Review-Url: https://codereview.webrtc.org/2542083003
Cr-Commit-Position: refs/heads/master@{#15389}
It is not safe to call vpx_codec_destroy if vpx_codec_dec_init failed,
because the |decoder_| memory will be uninitialized. See the bug for
more info.
BUG=chromium:663293
Review-Url: https://codereview.webrtc.org/2541163007
Cr-Commit-Position: refs/heads/master@{#15381}
The packet size was only used to control how often to output DTMF
packets. However, it likely did not work as intended, since that
interval was only set during initialization. No changes to the packet
size, like what AudioEncoder::Num10MsFramesInNextPacket could
indicate, were picked up. The value was instead taken from an entry in
ACMCodecDB.
Since it was not-fully-functional, its exact value didn't seem to
matter and it was getting in the way of making it possible to supply
an external audio encoder factory, I've decided to remove it
altogether. The DTMF code now uses an interval of 50 ms regardless,
which is a value recommended by the RFC.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2545753002
Cr-Commit-Position: refs/heads/master@{#15380}
AdaptiveVideoSource is used in testing/simulations of the bandwidth estimator.
Nada's reaction to delay depends on the current bitrate and the configured max rate in a non-intuituve way. Increase the starting bitrate to compensate for the increased max bitrate. This is only used in unit tests.
BUG=webrtc:6807
# Presubmit warns about a lint error in bwe.h that's unrelated to my change. Fixing it is beyond the scope of this CL.
NOPRESUBMIT=true
Review-Url: https://codereview.webrtc.org/2542843003
Cr-Commit-Position: refs/heads/master@{#15364}
These tests have been a little flaky on the bots.
Hopefully increasing the timeout by 200% will help.
BUG=webrtc:6799
Review-Url: https://codereview.webrtc.org/2541743006
Cr-Commit-Position: refs/heads/master@{#15355}
It is unused since cl https://codereview.webrtc.org/2386573002.
The new denoiser implementation and its tests are kept for now. This
code is also unused, but there are some plans to take this code into
use in the not too distant future.
BUG=None
Review-Url: https://codereview.webrtc.org/2496153002
Cr-Commit-Position: refs/heads/master@{#15338}
These tests have been causing a large number of false alerts, which is
tons of work for the perf sheriff. It's infeasible to test every single
permutation, so we must choose carefully. This CL reduces the number
of RESULT lines from the test from ~450 to ~50. I attempted to choose
interesting permutations, but you probably know better what's
interesting...
https://chromeperf.appspot.com/report?
sid=a7193c96f708018848ca07ad9c78ac657cadab3c70b3939c42bd7d70a092d61a
also suggests most of the metrics have enormous standard deviations,
so maybe you could look into how stable the metrics really are
and remove/stabilize the ones that aren't?
BUG=chromium:666725
Review-Url: https://codereview.webrtc.org/2529393006
Cr-Commit-Position: refs/heads/master@{#15329}
And delete the method CongestionController::packet_router.
BUG=None
Review-Url: https://codereview.webrtc.org/2516983004
Cr-Commit-Position: refs/heads/master@{#15323}
JitterBufferDelayInMs is used for the WebRTC-NewVideoJitterBuffer finch
experiment, and therefore needs to be calculated.
BUG=webrtc:5514
Review-Url: https://codereview.webrtc.org/2534093003
Cr-Commit-Position: refs/heads/master@{#15313}
To support rotation in DirectX capturer, several other changes are also
required.
1. Removing AddRect in RotateDesktopFrame, this is a performance improvement.
DxgiOutputDuplicator creates a rotated DesktopRegion, which can be directly
add to updated_region.
2. DxgiOutputDuplicator::SourceRect() is not accurate, the rectangle in source
is controlled by |offset| or |rotation_| + |offset|, instead of desktop_rect().
3. The |region| in DxgiTexture::CopyFrom() is not accurate. It needs an
unrotated DesktopRegion which offsets by |offset| instead of desktop_rect(). To
avoid generating both rotated and unrotated updated_region, this parameter has
been removed. This impacts DxgiTextureStagning performance a little bit (1.5ms).
Refer to bug for details.
BUG=webrtc:6646
Review-Url: https://codereview.webrtc.org/2530303002
Cr-Commit-Position: refs/heads/master@{#15308}
This CL adds WebRTC.Audio.ApmCaptureInputLevelAverage and
WebRTC.Audio.ApmCaptureInputLevelPeak. The metrics are updated once
every 10 seconds.
BUG=webrtc:6622
Review-Url: https://codereview.webrtc.org/2534473004
Cr-Commit-Position: refs/heads/master@{#15300}
