The packetization parts of this class are accessed from the
encoder thread, which might change under different occasions.
The use of a sequenced task checker here is thus incorrect, since
that requires the access to always be on the same thread, whenever
a task queue is not used.
The access to the instantiated object of this class, at least when
it comes to the RTP packetization parts, is however synchronized
using the lock in PayloadRouter::OnEncodedImage. We can therefore
safely remove the sequenced task checker.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2562983002
Cr-Commit-Position: refs/heads/master@{#15549}
The file was aldready pruned down to the point where it only included
webrtc/typedefs.h. Therefore, all includes of
voice_engine_configurations.h are replaced with typedefs.h, except on
two occasions where it was obvously not needed.
BUG=webrtc:6506
Review-Url: https://codereview.webrtc.org/2553583002
Cr-Commit-Position: refs/heads/master@{#15547}
This also deletes unused features of the video_capturer interface, the classes
VideoCaptureFeedBack, VideoCaptureEncodeInterface and related methods,
and the module id which used to be passed as an argument to the
VideoCaptureDataCallback.
In theory the module id could have been used to let a single
VideoCaptureDataCallback serve several capturers, and demultiplex
on the id, but in practice, it was unused. With this change, it is
required to use a separate VideoSinkInterface for each capturer.
BUG=webrtc:6789
Review-Url: https://codereview.webrtc.org/2534553002
Cr-Commit-Position: refs/heads/master@{#15540}
ScreenCapturerWinGdi randomly returns black frames in test environment. The root
cause is still unknown, so change ScreenCapturerWinGdi tests into MANUAL mode to
execute in test environment, but unblock other developers. We can eventually get
a failure ratio and more samples for debugging.
BUG=webrtc:6666, webrtc:6843
Review-Url: https://codereview.webrtc.org/2564173002
Cr-Commit-Position: refs/heads/master@{#15518}
Theil and Sen's estimator essentially looks at the line through every pair of points and selects the median slope. This is robust to corruption of up to 29% of the data points.
Wire up new estimator to field trial experiment. Add unit and integration tests. Results are promising.
BUG=webrtc:6728
Review-Url: https://codereview.webrtc.org/2512693002
Cr-Commit-Position: refs/heads/master@{#15508}
orders of the ProcessStream and ProcessReverseStream API calls.
This CL adds the ability to specify that call order in a file.
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2561843003
Cr-Commit-Position: refs/heads/master@{#15503}
is the one stored in the aecdump.
During AEC development, it is handy to be able to simulate different
doubletalk scenarios. This CL adds the ability to add an
artificial nearend on top of that present in the aecdump, which
allows for the developer to artificially customize the scenario
being tested
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2562593003
Cr-Commit-Position: refs/heads/master@{#15502}
Reason for revert:
Increase in encode time larger than expected.
Original issue's description:
> Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabled.
>
> Error resilience is currently always enabled for VP9 which reduces quality.
>
> BUG=webrtc:6783
>
> Committed: https://crrev.com/4eb03c76fa2320534d669fda2aabf800e7a6f579
> Cr-Commit-Position: refs/heads/master@{#15390}
TBR=stefan@webrtc.org,marpan@webrtc.org,mflodman@webrtc.org,marpan@google.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6783
Review-Url: https://codereview.webrtc.org/2554403006
Cr-Commit-Position: refs/heads/master@{#15501}
GDI capturer may randomly return a blank frame. So this change enables tests for
Linux / DirectX capturer / magnifier capturer.
BUG=webrtc:6666
Review-Url: https://codereview.webrtc.org/2559583002
Cr-Commit-Position: refs/heads/master@{#15489}
This moves some GN check configurations out of .gn to individual targets.
The now checked target is:
"//webrtc/modules/video_capture/*",
"//webrtc/modules/video_coding/*",
BUG=webrtc:6828
NOTRY=True
R=kjellander@webrtc.org
Review-Url: https://codereview.webrtc.org/2555333004
Cr-Commit-Position: refs/heads/master@{#15488}
For example, zero rtt may be reported to:
BitrateControllerImpl::OnReceivedRtcpReceiverReport:
- SendSideBandwidthEstimation::UpdateReceiverBlock
- SendSideBandwidthEstimation::UpdateUmaStats
BitrateAllocator::OnNetworkChanged:
- ProtectionBitrateCalculator::SetTargetRates
Re-add check that was removed in https://codereview.webrtc.org/2422063002.
BUG=webrtc:6692
Review-Url: https://codereview.webrtc.org/2552883010
Cr-Commit-Position: refs/heads/master@{#15486}
Also deletes one call to CongestionController::pacer.
BUG=None
Review-Url: https://codereview.webrtc.org/2542113003
Cr-Commit-Position: refs/heads/master@{#15479}
This moves some GN check configurations out of .gn to individual targets.
The now checked target is:
"//webrtc/modules/audio_processing/*",
BUG=webrtc:6828
NOTRY=True
R=kjellander@webrtc.org
Review-Url: https://codereview.webrtc.org/2558813003
Cr-Commit-Position: refs/heads/master@{#15475}
This CL breaks out the output sample rate calculation from
webrtc::AudioMixerImpl. A new OutputRateCalculator interface is added
to make the sample rate configurable. There are at least three reasons
for this change:
1. The mixer will be used for an internal project, in which no
resampling is done after the mixing. There the sample rate should
be static. Currently, it can differ across mix iterations and
depends on the number of audio sources. If there are no sources,
the WebRTC mixer behavior is to produce silence at 48 kHz.
2. A planned change to WebRTC will make audio processing steps
happen at constant sample rates. A configurable sample rate
calculator will make the transition simpler for the mixer.
3. The current mixer design is a single large file. Behavior is not
always simple to change (e.g. as in this case to mix at a
constant rate), unrelated behavior can be broken, reusing the
mixer in internal projects is tricky. Using DI for the sample
rate calculation solves parts of these issues.
Changes:
The protected mixer c-tor now takes
unique_ptr<OutputRateCalculator>. The current output rate calculation
is moved to DefaultOutputRateCalculator. A new factory method
AudioMixerImpl::CreateWithOutputRateCalculator is added. The old
factory method passes the default rate calculator.
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2557713006
Cr-Commit-Position: refs/heads/master@{#15472}
BitBlt returns a BOOL value, which should be taken care in ScreenCapturerWinGdi.
Meanwhile, this change also replaces assert() / abort() with RTC_DCHECK() /
RTC_CHECK() / RTC_NOTREACHED().
This change cannot fix the bug, the reason of the issue is still unknown, but it
is still the right thing to do.
In ScreenCapturerIntegrationTest, each frame will be captured at most 600 times.
Since the test case fails, which means the ScreenCapturerWinGdi consistently
returns a white frame for 600 times under a certain state. With this change,
instead of returning white frame, ScreenCapturerWinGdi will return a temporary
error. But I do not think a ScreenCapturerWinGdi can automatically recover by
retrying.
BUG=webrtc:6843
Review-Url: https://codereview.webrtc.org/2553353002
Cr-Commit-Position: refs/heads/master@{#15465}
This moves some GN check configurations out of .gn to individual targets.
The now checked targets are:
"//webrtc/api/*",
"//webrtc/audio/*",
"//webrtc/modules/audio_coding/*",
Many targets were fixed by adding dependencies, but the ones that
requires more refactorings are left with the check_includes attribute
set to false instead.
Make //webrtc/test:test_support a public dep of //webrtc/test:test_main
to avoid having to add that to all users of it.
BUG=webrtc:6828
NOTRY=True
Review-Url: https://codereview.webrtc.org/2556943003
Cr-Commit-Position: refs/heads/master@{#15461}
This makes it possible to use << and RTC_CHECK_EQ with this class.
BUG=none
Review-Url: https://codereview.webrtc.org/2554003002
Cr-Commit-Position: refs/heads/master@{#15456}
A trivial change to remove a deprecated comment.
BUG=chromium:314516
Review-Url: https://codereview.webrtc.org/2553283002
Cr-Commit-Position: refs/heads/master@{#15454}
Specifically, reject any bitrate allocated for a layer not representable
by the BitrateAllocation struct.
BUG=chromium:671312
Review-Url: https://codereview.webrtc.org/2549233005
Cr-Commit-Position: refs/heads/master@{#15447}
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}
