I found one additional way a crash could occur: "OnRtpTransportChanged"
being called instead of "OnDtlsTransportChanged", due to a mixup of m=
section types. I could reproduce this by:
1. Applying description with RTP data channel m= section.
2. Applying description with both a rejected RTP data channel m=
section and rejected SCTP m= section.
This is a very strange scenario, but maybe there are other ways to
reproduce that I haven't thought of.
The solution is to combine "OnRtpTransportChanged" and
"OnDtlsTransportChanged", and not do anything with the content type.
This simplifies the code a bit as well.
Bug: chromium:827917
Change-Id: If6818ea0c41573255831534060b30c76a6544e04
Reviewed-on: https://webrtc-review.googlesource.com/70360
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22893}
The existing rule regards a candidate pair as not receiving if it does
not receive any data packet, connectivity check, or connectivity check
response for a timeout period since the last receipt of any packet
above. A backup candidate pair typically sends connectivity checks at a
slow pace to preserve the battery life, and the existing rule however
declares receiving timeout for backup candidate pairs as a side effect.
This is a result of the conflicting value of the receiving timeout
period and the longer default connectivity check interval for backup
candidate pairs.
The new rule regards any candidate pair that has its last connectivity
check acknowledged by a response as receiving.
Bug: webrtc:9145
Change-Id: Ie0171fd83aca3d6a0a465885be32f0854856be7f
Reviewed-on: https://webrtc-review.googlesource.com/69784
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Honghai Zhang <honghaiz@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22892}
It seems iOS trybots are the only ones that build "all". This causes
problems when using Abseil because some targets in
//third_party/abseil-cpp fail to build (because they depend on CCTZ).
Bug: webrtc:8821
Change-Id: I017ecb0527a7e3f3c59f41053fa1878d16cbe4e9
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/70140
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22890}
This is a reland of fc43d11717e16dd427ac84fee614e5511e43cefd
Original change's description:
> Add thread checker to PortAllocator and its subclasses and fix a bug
> causing memory contention by threads.
>
> PortAllocator and its subclasses assume all of their methods except the
> constructor must be called on the same thread (the network thread in
> practice). This CL adds a thread checker to PortAllocator and its
> subclasses for thread safety, and fixes bugs of invoking some of their
> methods in PeerConnection on the signaling thread.
>
> Bug: webrtc:9112
> Change-Id: I33ba9bae72ec09a45ec70435962f3f25cd31583c
> Reviewed-on: https://webrtc-review.googlesource.com/66945
> Commit-Queue: Qingsi Wang <qingsi@google.com>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22814}
Bug: webrtc:9112
Change-Id: I5c7377f05c0daccbe469e2fdbdfacabc5c222f4c
Reviewed-on: https://webrtc-review.googlesource.com/69422
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22889}
RFC4566 says to use this if a session has no meaningful name. But we
were rejecting it due to another rule that says "whitespace MUST NOT be
used on either side of the = sign".
Bug: chromium:590625
Change-Id: I5d632f2cb371060adee794febe19bdfe76cb20ed
Reviewed-on: https://webrtc-review.googlesource.com/70262
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22887}
This reverts commit aaa85ae565989f42b811c9a4858bb087319ba214.
Reason for revert: Breaks iOS64 Debug trybot: https://uberchromegw.corp.google.com/i/internal.client.webrtc/builders/iOS64%20Debug/builds/14014
The failure being at:
../../test/fpe_observer_unittest.cc:93: Failure
Expected equality of these values:
0x009f
Which is: 159
all_flags
Which is: 31
It looks like the missing flag may be "FE_FLUSHTOZERO"?
Original change's description:
> Reland "Floating-point exception observer for unit tests"
>
> This reverts commit e3d522dd6b52025191bacfab241f130e9870941f.
>
> Reason for revert: Disabling test failing in downstream projects.
>
> Original change's description:
> > Revert "Floating-point exception observer for unit tests"
> >
> > This reverts commit 3fb3939896f6270d48aff34eee2946bd7661bd63.
> >
> > Reason for revert: Downstream projects failures.
> >
> > Original change's description:
> > > Floating-point exception observer for unit tests
> > >
> > > This CL adds a simple tool that let a unit test fail if a floating
> > > point exception occurs. It is possible to focus on specific exceptions.
> > > Note that FloatingPointExceptionObserver is only effective in debug
> > > mode. For this reason, the related unit tests only run in debug mode.
> > > Plus, due to some platform-specific limitations, not all the floating
> > > point exceptions are available on Android.
> > >
> > > Bug: webrtc:8948
> > > Change-Id: I0956e27f2f3aa68771dd647169fba7968ccbd771
> > > Reviewed-on: https://webrtc-review.googlesource.com/58097
> > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > > Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#22768}
> >
> > TBR=phoglund@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org
> >
> > Change-Id: I0fd3d114ab4a348fd46339e98273e19c1ac1c6dc
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: webrtc:8948
> > Reviewed-on: https://webrtc-review.googlesource.com/67380
> > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22769}
>
> TBR=phoglund@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:8948
> Change-Id: I7584d941b227277a271323b47bc70945af999758
> Reviewed-on: https://webrtc-review.googlesource.com/69060
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22848}
TBR=phoglund@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:8948
Change-Id: Ia377cea165211a0fad8f7ab29baae3eee64395c3
Reviewed-on: https://webrtc-review.googlesource.com/70280
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22886}
Since we always pass in the first audio channel, we should always pass 1 as the number of channels in the initialization function.
Bug: webrtc:8732
Change-Id: I978edb125d7cc701a5e07193256327908be00560
Reviewed-on: https://webrtc-review.googlesource.com/69660
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22885}
This reverts commit 64051d4975b5cee06ab36584f272ff97e35de357.
Reason for revert: Fix applied.
Original change's description:
> Revert "Android: Generalize and make TextureBufferImpl public"
>
> This reverts commit 28111d7fa0b94e37a5eeba616eb806c65b12560e.
>
> Reason for revert: Crashes video_quality_loopback_test.
>
> Original change's description:
> > Android: Generalize and make TextureBufferImpl public
> >
> > This CL generalizes TextureBufferImpl so it's useful from other contexts than
> > from a SurfaceTextureHelper, and fixes a bug in cropAndScale(). It also exposes
> > the class in the api so that clients don't have to duplicate the logic.
> >
> > Bug: None
> > Change-Id: Ib82aa8bee025ec14de74a7be9d91fd4e5298a248
> > Reviewed-on: https://webrtc-review.googlesource.com/69819
> > Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> > Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22875}
>
> TBR=magjed@webrtc.org,sakal@webrtc.org
>
> Change-Id: Ica7fc181fec70b8b79f39f0e114eef81a03aa116
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: None
> Reviewed-on: https://webrtc-review.googlesource.com/70240
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22878}
TBR=magjed@webrtc.org,sakal@webrtc.org
Change-Id: I173d1ccfe0baa80460f796ebaedc51731233108f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/70183
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22883}
The code using the units now depends on specific targets to make the
dependencies more clear
Bug: None
Change-Id: I3200d57a2974b6981db68f05d84391cbbb06e981
Reviewed-on: https://webrtc-review.googlesource.com/70181
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22882}
It's a module for applying a gain to the capture signal.
The gain is the first processing step in APM.
After this CL, these two features work:
* The PreAmplifier can be activated with
AudioProcessing::Config::pre_amplifier
* The PreApmlifier can be controlled after APM creation by
AudioProcessing::SetRuntimeSetting.
What's left is a change to AecDumps and to AecDump-replay.
NOTRY=True # 1-line change, tests just passed.
Bug: webrtc:9138
Change-Id: I85b3af511695b0a9cec2eed6fee7f05080305e1d
Reviewed-on: https://webrtc-review.googlesource.com/69811
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22881}
With rounding to the nearest the result can exceed the allocated
bitrate.
Bug: none
Change-Id: I0260a1640a1454951ca8e48fd447e047ef0271ee
Reviewed-on: https://webrtc-review.googlesource.com/69982
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22879}
This reverts commit 28111d7fa0b94e37a5eeba616eb806c65b12560e.
Reason for revert: Crashes video_quality_loopback_test.
Original change's description:
> Android: Generalize and make TextureBufferImpl public
>
> This CL generalizes TextureBufferImpl so it's useful from other contexts than
> from a SurfaceTextureHelper, and fixes a bug in cropAndScale(). It also exposes
> the class in the api so that clients don't have to duplicate the logic.
>
> Bug: None
> Change-Id: Ib82aa8bee025ec14de74a7be9d91fd4e5298a248
> Reviewed-on: https://webrtc-review.googlesource.com/69819
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22875}
TBR=magjed@webrtc.org,sakal@webrtc.org
Change-Id: Ica7fc181fec70b8b79f39f0e114eef81a03aa116
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/70240
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22878}
Having only one name for seconds makes the interface more consistent.
The non-abbreviated was chosen since it's used less frequently than
ms() and us().
Bug: None
Change-Id: Ia29ff2f9f18f3dddcde9bac4f041695cef2c8f0f
Reviewed-on: https://webrtc-review.googlesource.com/69817
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22877}
Add configuration fields for the pre-amplifier in the Audio Processing
Module. Also add flags and settings for the pre-amplifier in
audioproc_f.
Also make the setting stored in Aec Dumps. And make the setting
applied when playing back Aec Dumps in audioproc_f.
Bug: webrtc:9138
Change-Id: I4e59df200e1ebc56f06fae74ebf17d85858958a3
Reviewed-on: https://webrtc-review.googlesource.com/69560
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22876}
This CL generalizes TextureBufferImpl so it's useful from other contexts than
from a SurfaceTextureHelper, and fixes a bug in cropAndScale(). It also exposes
the class in the api so that clients don't have to duplicate the logic.
Bug: None
Change-Id: Ib82aa8bee025ec14de74a7be9d91fd4e5298a248
Reviewed-on: https://webrtc-review.googlesource.com/69819
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22875}
This reduces the number of files.
Bug: None
Change-Id: Ia32b7b7cc3260fbecc2b9a3c75723dd4a76c6d5b
Reviewed-on: https://webrtc-review.googlesource.com/69816
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22874}
This CL includes the following changes:
- APM runtime setting (ID + float payload) and unit tests
- Swap queue of APM runtime settings used in AudioProcessingImpl
- runtime settings handler that forwards the settings to APM
sub-modules when a message is retrieved from the queue
- Unit test placeholder to check that the pre-gain update message
is correctly delivered
Bug: webrtc:9138
Change-Id: Id22704af15fde2b87a4431f5ce64ad1aeafc5280
Reviewed-on: https://webrtc-review.googlesource.com/69320
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22873}
This fixes inconsistency in names of variables and fields which
represent spatial/temporal index of layer:
simulcast_svc_idx -> spatial_idx
spatial_layer_idx -> spatial_idx
temporal_layer_idx -> temporal_idx
Also, this adds printing of spatial/temporal index and target bitrate
to RD report.
Bug: none
Change-Id: Ic4dfdadc57a1577bb3d35d1782a152a9dbef0280
Reviewed-on: https://webrtc-review.googlesource.com/69981
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22869}
This file was still reflecting the old structure of the repository.
This CL updates all the paths and removes configs to track deleted
directories.
Bug: webrtc:9152
Change-Id: Iaed184d9e7100361676015d7c6ddbd04439e0a45
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/69818
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@google.com>
Cr-Commit-Position: refs/heads/master@{#22868}
BundleFilter is replaced by RtpDemuxer in RtpTransport for payload
type-based demuxing. RtpTransport will support MID-based demuxing later.
Each BaseChannel has its own RTP demuxing criteria and when connecting
to the RtpTransport, BaseChannel will register itself as a demuxer sink.
The inheritance model is changed. New inheritance chain:
DtlsSrtpTransport->SrtpTransport->RtpTranpsort
The JsepTransport2 is renamed to JsepTransport.
NOTE:
When RTCP packets are received, Call::DeliverRtcp will be called for
multiple times (webrtc:9035) which is an existing issue. With this CL,
it will become more of a problem and should be fixed.
Bug: webrtc:8587
Change-Id: Ibd880e7b744bd912336a691309950bc18e42cf62
Reviewed-on: https://webrtc-review.googlesource.com/65786
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22867}
While investigating some screen-capture-track-end-in-meeting issues, the
relevant rtc error logs are not uploaded to server as other webrtc
modules do, which cause great hardness to identify the reason.
This cl is to use existing trace event methods to store error logs of
desktop capturers.
Bug: chromium:831756
Change-Id: Id0c1b439f9b63916fb9417cf4e6f2b8f3c556fcd
Reviewed-on: https://webrtc-review.googlesource.com/69783
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22866}
This flag is unused.
Bug: None
Change-Id: I1ad52feca1db8e669f4e7c7c5b45a4cb245c1c55
Reviewed-on: https://webrtc-review.googlesource.com/69780
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22865}
This adds confusion to the native API and is only needed for
Chromium UMA metrics, so the appropriate metrics have been moved
upstream and kDefault option removed.
Bug: chromium:811683
Change-Id: I666d7f7793765b8d6edcd99416c8b6c957766f00
Reviewed-on: https://webrtc-review.googlesource.com/59261
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22864}
Adding SVC rate allocator and layering configurator caused regression
for VP9 non-SVC senders. SVC bitrate limits, which were supposed to
be used only when spatial layering is enabled, are applied when
encoding single spatial layer. E.g. for VP9 360p sender maximum bitrate
is limited to 500kbps.
This fixes the regression. If sender is configured to send VP9 single
layer then codec's bitrate limits are applied to this layer.
Bug: webrtc:9151, chromium:831093
Change-Id: Ia1ae4087155ad7917a3443304a21532f1e68ea65
Reviewed-on: https://webrtc-review.googlesource.com/69813
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22862}
This CL moves the network units files into a separate folder with a
separate BUILD file. It also splits the units into separate files.
This prepares for moving all or some of the units to somewhere that
can be accessed by more components.
Bug: None
Change-Id: I4ebbc19088b024ba920b0b3c64e5f57431f4f955
Reviewed-on: https://webrtc-review.googlesource.com/68660
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22861}
This simplifies configuration, and it is a preparation for replaceing
encoder instance with an encoder factory in
VideoSendStream::Config::EncoderSettings.
Bug: webrtc:8830
Change-Id: Iaf4f6ad9e7cfaa76d8600c4fa68f393e2f3ea331
Reviewed-on: https://webrtc-review.googlesource.com/69809
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22859}
The SendSideCongestionController version is toggled by an experiment in
RtpTransportControllerSend. This CL adds a log statement of which
version is used, to make debugging easier.
Bug: webrtc:8415
Change-Id: I6201cf5f03e097cc07c6ae120dcff075c046c414
Reviewed-on: https://webrtc-review.googlesource.com/69808
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22858}
UpdatePacingRates required that a bandwidth estimate was available and
would otherwise crash. This CL ensures that there is an initial bandwidth
estimate available from the beginning.
Bug: webrtc:8415
Change-Id: I20c3b444eac42326a78cfebee70b4c1aa370c867
Reviewed-on: https://webrtc-review.googlesource.com/69802
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22857}
This disables toggling between transport feedback based controller and
the fall back controller in SendSideCongestionController. The toggling
seems to cause issues with the probing in certain circumstances. Since
it's feasible to run experiments without the toggling, disable it for now.
Bug: webrtc:8415
Change-Id: Ia4a827e95d730d651eaf3facbee7e9a5b0cb2562
Reviewed-on: https://webrtc-review.googlesource.com/69803
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22856}
The paced sender did not update the time out clock before the first
packet was send in paused state. This caused it to incorrectly log
warnings about elapsed time. This CL fixes this.
Bug: None
Change-Id: I240d169464a708c12eb580d57bc385330b8dd6b1
Reviewed-on: https://webrtc-review.googlesource.com/69561
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22852}
Fixes potential crash in SendSideCongestionController when route is
changed before network is available.
Bug: webrtc:8415
Change-Id: I781f0e342e5bb42fedbf96c9c5c6d2c199ab3192
Reviewed-on: https://webrtc-review.googlesource.com/69801
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22851}
Added a new sub-module 'GainApplier'. The build target is
'modules/audio_processing/agc2:gain_applier'. A small refactoring
makes the GainApplier used in adaptive-digital AGC2.
The AGC2 now multiplies samples with a gain in 3 places. It's the
GainApplier, the GainCurveApplier, and the FixedGainController. The
GainApplier is used in AdaptiveDigitalGainApplier and will be used as
a pre-amplifier.
Bug: webrtc:9138
Change-Id: Ibc4c0ea109c6757f159d4adb6e3d8614179c9bc6
Reviewed-on: https://webrtc-review.googlesource.com/69321
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22849}
This reverts commit e3d522dd6b52025191bacfab241f130e9870941f.
Reason for revert: Disabling test failing in downstream projects.
Original change's description:
> Revert "Floating-point exception observer for unit tests"
>
> This reverts commit 3fb3939896f6270d48aff34eee2946bd7661bd63.
>
> Reason for revert: Downstream projects failures.
>
> Original change's description:
> > Floating-point exception observer for unit tests
> >
> > This CL adds a simple tool that let a unit test fail if a floating
> > point exception occurs. It is possible to focus on specific exceptions.
> > Note that FloatingPointExceptionObserver is only effective in debug
> > mode. For this reason, the related unit tests only run in debug mode.
> > Plus, due to some platform-specific limitations, not all the floating
> > point exceptions are available on Android.
> >
> > Bug: webrtc:8948
> > Change-Id: I0956e27f2f3aa68771dd647169fba7968ccbd771
> > Reviewed-on: https://webrtc-review.googlesource.com/58097
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22768}
>
> TBR=phoglund@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org
>
> Change-Id: I0fd3d114ab4a348fd46339e98273e19c1ac1c6dc
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8948
> Reviewed-on: https://webrtc-review.googlesource.com/67380
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22769}
TBR=phoglund@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:8948
Change-Id: I7584d941b227277a271323b47bc70945af999758
Reviewed-on: https://webrtc-review.googlesource.com/69060
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22848}
The issue is visible when reconfiguring the screen arrangement while
sharing the displays. Can sometimes be seen right after starting the
screen sharing.
Indeed CaptureFrame can be called at any time so TakeLatestFrameForDisplay
should always return a valid frame and the call should not empty the
internal container.
Also add missing teardown in the provider on failure case.
Bug: webrtc:8652
Change-Id: Ice151c1da92b9ad2b86ca9368d30d9d21114e53e
Reviewed-on: https://webrtc-review.googlesource.com/69420
Commit-Queue: Zijie He <zijiehe@chromium.org>
Reviewed-by: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#22846}
- Directly include api/audio/audio_frame.h everywhere AudioFrame is used.
- This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM.
- audio_frame.h still included from module_common_types.h for backwards compatibility with clients.
Bug: webrtc:9139, webrtc:7504
Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897
Reviewed-on: https://webrtc-review.googlesource.com/62464
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22845}
This CL fixes issues when applying a description with an empty BUNDLE
group (previously it would fail, after recent refactoring it started
crashing).
This CL also will cause an empty BUNDLE group to be generated when it
should be. Namely, when responding to an offer that had a BUNDLE group,
rejecting everything in it.
Bug: chromium:831996
Change-Id: I4e705a328daef4e81f8f1ace6aa73ddfa13c0107
Reviewed-on: https://webrtc-review.googlesource.com/69720
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22844}
