This conversion function is useful for downstream native codecs
which want to take a VideoCodecInfo in the Java wrapper and use it
as a SdpVideoFormat in the C++ object.
Bug: webrtc:9495
Change-Id: Id2ece708eafd4be648fafff9e94aa13ace504bdf
Reviewed-on: https://webrtc-review.googlesource.com/87381
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23865}
Fix the build with libstdc++ after 496cedfe5 ("AEC3: Reverberation model:
Changes on the decay estimation") by including the missing header:
../../modules/audio_processing/aec3/reverb_model_estimator.cc: In constructor ‘webrtc::ReverbModelEstimator::ReverbModelEstimator(const webrtc::EchoCanceller3Config&)’:
../../modules/audio_processing/aec3/reverb_model_estimator.cc:61:21: error: ‘fabsf’ was not declared in this scope
reverb_decay_(fabsf(config.ep_strength.default_len)),
^~~~~
../../modules/audio_processing/aec3/reverb_model_estimator.cc:61:21: note: suggested alternative: ‘labs’
reverb_decay_(fabsf(config.ep_strength.default_len)),
^~~~~
labs
../../modules/audio_processing/aec3/reverb_model_estimator.cc: In member function ‘void webrtc::ReverbModelEstimator::UpdateReverbDecay(const std::vector<float>&)’:
../../modules/audio_processing/aec3/reverb_model_estimator.cc:206:15: error: ‘powf’ was not declared in this scope
decay = powf(2.0f, -exp_candidate * kFftLengthBy2);
^~~~
Spotted by Jose Dapena Paz <jose.dapena@lge.com>.
Bug: chromium:819294
Change-Id: If992e5e473b9d4d0c1b3c1006c3816b7c4eee296
Reviewed-on: https://webrtc-review.googlesource.com/87241
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Raphael Kubo da Costa (rakuco) <raphael.kubo.da.costa@intel.com>
Cr-Commit-Position: refs/heads/master@{#23863}
This clarifies the dependencies of utility/ a lot (spoiler:
there are very few) and makes it easier to separate the build
targets for aecm and aec2.
Bug: webrtc:9488
Change-Id: If916f86e80c19d1b650d0908fbe8343ea7c47bd7
Reviewed-on: https://webrtc-review.googlesource.com/87141
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23860}
The main filter is adapted at a lower rate which reduces the risk of
diverging during double talk. The change yields notable transparency
improvements.
Bug: webrtc:9497
Change-Id: Ib23b7a4055d313dede535d2b65dc7e023a2db042
Reviewed-on: https://webrtc-review.googlesource.com/87300
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23858}
The size limit is set to correspond to approximately 5 seconds of
decoded audio at the codecs' normal operating bitrates. This is to
avoid timeouts on the bots.
NOTRY=true
Bug: chromium:840115
Change-Id: I74b3c196259e03981aa2c4ef349e6e1334e9bf58
Reviewed-on: https://webrtc-review.googlesource.com/87302
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23857}
This CL is in preparation to change the RTPVideoTypeHeader into an absl::variant.
Bug: none
Change-Id: I1672d866df0395f3417d8e278cc67f017ab0ff98
Reviewed-on: https://webrtc-review.googlesource.com/87261
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23856}
All usage of these functions should be migrated to use
android.opengl.Matrix instead.
Bug: webrtc:9487
Change-Id: I023761b31cae7e7af9b537928b849657baa5bb8b
Reviewed-on: https://webrtc-review.googlesource.com/87263
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23853}
The current acknowledged bitrate (i.e. throughput) estimator takes 500ms
to initialize. This CL creates a field trial to experiment with this
arbitrary initialization time.
Bug: webrtc:9492, webrtc:7746
Change-Id: I8a803f7bc0ee78856e808e289f37bab57d763efa
Reviewed-on: https://webrtc-review.googlesource.com/87145
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23852}
This reverts commit 425193b4a92f0df1f3fbea3626b9abf6a38f67ec.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Revert "Unit test for case where the number of active and configured spatial"
>
> This reverts commit 5eb6045ce5754ce815929c54dd27ab0bf3ae62ba.
>
> Reason for revert: Test breaks downstream.
>
> Original change's description:
> > Unit test for case where the number of active and configured spatial
> > layers differ.
> >
> > Bug: webrtc:9472
> > Change-Id: I5cf292a12d73777ca0fd5771eb1a4756626f640c
> > Reviewed-on: https://webrtc-review.googlesource.com/85644
> > Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#23782}
>
> TBR=brandtr@webrtc.org,ssilkin@webrtc.org,mhoro@webrtc.org
>
> Change-Id: Ib97cdb127e79ee969f7cb3f931cb7bd533f13af0
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9472
> Reviewed-on: https://webrtc-review.googlesource.com/86320
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23785}
TBR=brandtr@webrtc.org,terelius@webrtc.org,ssilkin@webrtc.org,mhoro@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9472
Change-Id: I796909c553702a0fa19e5e16e4586f915569b134
Reviewed-on: https://webrtc-review.googlesource.com/87220
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23847}
Since the frame is processed on the same thread as the decoding happens
on, keeping a reference to the frame may cause deadlocks on some
implementations.
Longer term, we should probably move the frame processing to a separate
thread but this is an easy fix for now.
Bug: b/110246814
Change-Id: I251737e2188e1755d45b35165586d1b0daf14595
Reviewed-on: https://webrtc-review.googlesource.com/87104
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23843}
This CL also adds a test to test the behavior when StartRecording()
fails, which is the case when e.g. the microphone is already in use.
Bug: webrtc:9491
Change-Id: Ifce60ce5e9b7fa7521ca5c9fe20794233456b9ce
Reviewed-on: https://webrtc-review.googlesource.com/87105
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23842}
This calls webrtc::AudioProcessing::set_stream_key_pressed, which
opens up a lot of code paths in the transient suppressor.
The change breaks historical fuzzer test cases.
Bug: webrtc:9413
Change-Id: I1f593a98286c7e7c0fc6751d16df40ad813dbd70
Reviewed-on: https://webrtc-review.googlesource.com/86950
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23840}
In this CL we have introduced changes on the estimation of the decay involved in the exponential modeling of the reverberation. Specifically, the instantaneous ERLE has been tracked and used for adapting faster in the regions when the linear filter is performing well. Furthermore, the adaptation is just perform during render activity.
Change-Id: I974fd60e4e1a40a879660efaa24457ed940f77b4
Bug: webrtc:9479
Reviewed-on: https://webrtc-review.googlesource.com/86680
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23836}
Also adjust to base-layer fraction for the shortened 3-tl pattern to be
60%, just like the 2-tl setting.
This CL removes direct use of the allocation matrix and moves it behind
a static getter.
Bug: webrtc:9477
Change-Id: Ifd7d1edffa0555024fd252834357b926997d13b5
Reviewed-on: https://webrtc-review.googlesource.com/86681
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23834}
This PRESUBMIT check will ensure that WebRTC does not regress on
the amount of warning suppression flags used.
At the moment it only checks for //build/config/clang:extra_warnings.
Error message:
** Presubmit ERRORS **
Usage of //build/config/clang:extra_warnings is discouraged in WebRTC.
If you are not adding this code (e.g. you are just moving existing code) or you want to add an exception,
you can add a comment on the line that causes the problem:
"-Wno-odr" # no-presubmit-check TODO(bugs.webrtc.org/BUG_ID)
Affected files:
api/BUILD.gn (line 30)
Bug: webrtc:9251
Change-Id: I059cbc648ca6f6806cf5e936e0b83b72ec4f3f50
Reviewed-on: https://webrtc-review.googlesource.com/86942
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23831}
This change simplifies the calculation of the suppression gains.
It also contains a new tuning of the suppressor.
The suppressor behavior is tuned by setting echo-to-nearend ratios
for when the suppressor is to be fully transparent and for when to
fully suppress. An echo-to-masker value determines when the signal
is masked by noise. These three values are specified for low and
high frequencies.
Change-Id: I108e83c8f2a35462085a3fabaebcc02fa3103607
Bug: webrtc:9482
Reviewed-on: https://webrtc-review.googlesource.com/86021
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23830}
With deadbeef removed from these OWNERS files, Steve is the only OWNER
on our team. I'm adding myself, because I have worked in these
directories and it makes sense to be able to distribute the code
reviews.
NOTRY=True
Bug: None
Change-Id: I48e88a07ee42254d937391f500f273656853d98b
Reviewed-on: https://webrtc-review.googlesource.com/86980
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23826}
This CL makes NetEq handle nested RED packets without crashing. Nested
RED packets mean that the block PT (see
https://tools.ietf.org/html/rfc2198.html#section-3) in the RED packet
is also set to the RED PT. This implies a nested RED packet, which is
not supported. Instead, all payloads in a RED packet that have the RED
PT will be discarded.
Bug: chromium:851662
Change-Id: I86ec257e60fb8076e3574ac5a4a1ca50196f6b34
Reviewed-on: https://webrtc-review.googlesource.com/86824
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23825}
These are useful for plotting creating data files that can be used to
visualize and debug congestion controller behavior.
Bug: webrtc:9467
Change-Id: I75b03a309b4b7d562fefe82a828ae1e6a9f069c8
Reviewed-on: https://webrtc-review.googlesource.com/86126
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23822}
This simplifies the logic, prevents emitting code for every pair of
argument types to RTC_CHECK_OP and partially unblocks removing streams from
the check code altogether.
Bug: webrtc:8982
Change-Id: Ib6652ac9a342e4471c12574a79872833cc943407
Reviewed-on: https://webrtc-review.googlesource.com/86544
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23821}
