[This CL is a rebase of an original CL by solenberg@:
https://codereview.webrtc.org/2948763002/ which in turn was a
rebase of an original CL by peah@:
https://chromium-review.googlesource.com/c/527032/]
Allow an external audio processing module to be used in WebRTC
This CL adds support for optionally using an externally created audio
processing module in a peerconnection. The ownership is shared
between the peerconnection and the external creator of the module.
As part of this the internal ownership of the audio processing module
is moved from VoiceEngine to WebRtcVoiceEngine.
BUG=webrtc:7775
Review-Url: https://codereview.webrtc.org/2961723004
Cr-Commit-Position: refs/heads/master@{#18837}
Reland the base->rtc_base without adding stub headers (will be
done in follow-up CL). This preserves git blame history of all files.
BUG=webrtc:7634
NOTRY=True
TBR=kwiberg@webrtc.org
Change-Id: Iea3bb6f3f67b8374c96337b63e8f5aa3e6181012
Reviewed-on: https://chromium-review.googlesource.com/554611
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18821}
This refactoring takes a careful approach to avoid rushing the change:
* stub headers are left in all the old locations of webrtc/base
* existing GN targets are kept and now just forward to the moved ones
using public_deps.
The only exception to the above is the base_java target and its .java files,
which were moved to webrtc/rtc_base right away since it's not possible
to use public_deps for android_library.
To avoid breaking builds, a temporary Dummy.java file was added to
the new intermediate target in webrtc/rtc_base:base_java as well to avoid
hitting a GN assert in the android_library template.
The above approach should make the transition smooth without breaking
downstream.
A helper script was created (https://codereview.webrtc.org/2879203002/)
and was run like this:
stub-headers.py -s webrtc/base -d webrtc/rtc_base -i 7634
stub-headers.py -s webrtc/base/numerics -d webrtc/rtc_base/numerics -i 7634
Fixed invalid header guards in the following files:
webrtc/base/base64.h
webrtc/base/cryptstring.h
webrtc/base/event.h
webrtc/base/flags.h
webrtc/base/httpbase.h
webrtc/base/httpcommon-inl.h
webrtc/base/httpcommon.h
webrtc/base/httpserver.h
webrtc/base/logsinks.h
webrtc/base/macutils.h
webrtc/base/nattypes.h
webrtc/base/openssladapter.h
webrtc/base/opensslstreamadapter.h
webrtc/base/pathutils.h
webrtc/base/physicalsocketserver.h
webrtc/base/proxyinfo.h
webrtc/base/sigslot.h
webrtc/base/sigslotrepeater.h
webrtc/base/socket.h
webrtc/base/socketaddresspair.h
webrtc/base/socketfactory.h
webrtc/base/stringutils.h
webrtc/base/testbase64.h
webrtc/base/testutils.h
webrtc/base/transformadapter.h
webrtc/base/win32filesystem.h
Added new header guards to:
sslroots.h
testbase64.h
BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True
R=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2877023002 .
Cr-Commit-Position: refs/heads/master@{#18816}
Patch set 1 is a reland + trivial rebase.
Patch set >= 2 contains bug fixes.
> Original issue's description:
> > Fix bug in vie_encoder.cc which caused channel parameters not to be updated at regular intervals, as it was intended.
> >
> > That however exposes a bunch of failed test, so this CL also fixed a few other things:
> > * FakeEncoder should trust the configured FPS value rather than guesstimating itself based on the realtime clock, so as not to completely undershoot targets in offline mode. Also, compensate for key-frame overshoots when outputting delta frames.
> > * FrameDropper should not assuming incoming frame rate is 0 if no frames have been seen.
> > * Fix a bunch of test cases that started failing because they were relying on the fake encoder undershooting.
> > * Fix test
> >
> > BUG=7664
> >
> > Review-Url: https://codereview.webrtc.org/2883963002
> > Cr-Commit-Position: refs/heads/master@{#18473}
> > Committed: 6431e21da6
BUG=webrtc:7664
Review-Url: https://codereview.webrtc.org/2953053002
Cr-Commit-Position: refs/heads/master@{#18782}
An automatic undef checker noticed an error in the new
audio_processing_fuzzer. An array was default-initialized, meaning its
contents had indeterminate values. This array was later filtered for
forbidden values, which happened or did not happen based on the
uninitialized values. Hence the error.
TBR=henrik.lundin@webrtc.org
NOTRY=true
Bug: 735718, webrtc:7820
Change-Id: Ib50a6f7a5f054472fef07a27d37e02cea432a8ca
Reviewed-on: https://chromium-review.googlesource.com/544937
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18713}
This is a high-level fuzzer that creates an AudioProcessing instance.
All possible combinations of publicly visible components are
fuzzed. Input and output sample rate, call order and use of the
float/fix interface is fuzzed. Sample rate may change between calls.
To fuzz floating point numbers, raw data is converted to floats, and
filtered for special values like 'inf', 'nan' and very large
values. Note that the default use case of APM is to only allow values
between +/- 2^15.
BUG=webrtc:7820
Review-Url: https://codereview.webrtc.org/2876793002
Cr-Commit-Position: refs/heads/master@{#18678}
Timing information is gathered in EncodedImage,
starting at encoders. Then it's sent using RTP header extension. In the
end, it's gathered at the GenericDecoder. Actual reporting and tests
will be in the next CLs.
BUG=webrtc:7594
Review-Url: https://codereview.webrtc.org/2911193002
Cr-Commit-Position: refs/heads/master@{#18659}
This is to make it possible to override the rtc_task_queue target only.
BUG=none
Review-Url: https://codereview.webrtc.org/2931273002
Cr-Commit-Position: refs/heads/master@{#18534}
Fix issue where running tests on iOS would get killed after a certain
time had passed. This seems to be due to springboard killing apps
that don't have a GUI running. Creating a UIApplication to wrap
the test suite seems to solve this problem in chromium.
This CL adds a class for this purpose. Most of the code was copied
from chromium with bits taken out.
Bug: webrtc:7161, webrtc:7758
Change-Id: I10f9bc8914e73f2870a9b0a2703cde496af8db2f
Reviewed-on: https://chromium-review.googlesource.com/528173
Reviewed-by: Henrik Andreasson <henrika@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Kári Tristan Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18509}
Reason for revert:
Looks like there's still one failing perf test:
RampUpTest.UpDownUpTransportSequenceNumberPacketLoss
Original issue's description:
> Reland of Periodically update codec bit/frame rate settings. (patchset #1 id:1 of https://codereview.webrtc.org/2923993002/ )
>
> Reason for revert:
> Create reland cl that we can patch with fix.
>
> Original issue's description:
> > Revert of Periodically update codec bit/frame rate settings. (patchset #8 id:140001 of https://codereview.webrtc.org/2883963002/ )
> >
> > Reason for revert:
> > Breaks some Call perf tests that are not run by the try bots....
> >
> > Original issue's description:
> > > Fix bug in vie_encoder.cc which caused channel parameters not to be updated at regular intervals, as it was intended.
> > >
> > > That however exposes a bunch of failed test, so this CL also fixed a few other things:
> > > * FakeEncoder should trust the configured FPS value rather than guesstimating itself based on the realtime clock, so as not to completely undershoot targets in offline mode. Also, compensate for key-frame overshoots when outputting delta frames.
> > > * FrameDropper should not assuming incoming frame rate is 0 if no frames have been seen.
> > > * Fix a bunch of test cases that started failing because they were relying on the fake encoder undershooting.
> > > * Fix test
> > >
> > > BUG=7664
> > >
> > > Review-Url: https://codereview.webrtc.org/2883963002
> > > Cr-Commit-Position: refs/heads/master@{#18473}
> > > Committed: 6431e21da6
> >
> > TBR=stefan@webrtc.org,holmer@google.com
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=7664
> >
> > Review-Url: https://codereview.webrtc.org/2923993002
> > Cr-Commit-Position: refs/heads/master@{#18475}
> > Committed: 5390c4814d
>
> TBR=stefan@webrtc.org,holmer@google.com
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=7664
>
> Review-Url: https://codereview.webrtc.org/2924023002
> Cr-Commit-Position: refs/heads/master@{#18497}
> Committed: cdafeda1cbTBR=stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=7664
Review-Url: https://codereview.webrtc.org/2926283002
Cr-Commit-Position: refs/heads/master@{#18500}
Reason for revert:
Create reland cl that we can patch with fix.
Original issue's description:
> Revert of Periodically update codec bit/frame rate settings. (patchset #8 id:140001 of https://codereview.webrtc.org/2883963002/ )
>
> Reason for revert:
> Breaks some Call perf tests that are not run by the try bots....
>
> Original issue's description:
> > Fix bug in vie_encoder.cc which caused channel parameters not to be updated at regular intervals, as it was intended.
> >
> > That however exposes a bunch of failed test, so this CL also fixed a few other things:
> > * FakeEncoder should trust the configured FPS value rather than guesstimating itself based on the realtime clock, so as not to completely undershoot targets in offline mode. Also, compensate for key-frame overshoots when outputting delta frames.
> > * FrameDropper should not assuming incoming frame rate is 0 if no frames have been seen.
> > * Fix a bunch of test cases that started failing because they were relying on the fake encoder undershooting.
> > * Fix test
> >
> > BUG=7664
> >
> > Review-Url: https://codereview.webrtc.org/2883963002
> > Cr-Commit-Position: refs/heads/master@{#18473}
> > Committed: 6431e21da6
>
> TBR=stefan@webrtc.org,holmer@google.com
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=7664
>
> Review-Url: https://codereview.webrtc.org/2923993002
> Cr-Commit-Position: refs/heads/master@{#18475}
> Committed: 5390c4814dTBR=stefan@webrtc.org,holmer@google.com
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=7664
Review-Url: https://codereview.webrtc.org/2924023002
Cr-Commit-Position: refs/heads/master@{#18497}
Reason for revert:
Breaks some Call perf tests that are not run by the try bots....
Original issue's description:
> Fix bug in vie_encoder.cc which caused channel parameters not to be updated at regular intervals, as it was intended.
>
> That however exposes a bunch of failed test, so this CL also fixed a few other things:
> * FakeEncoder should trust the configured FPS value rather than guesstimating itself based on the realtime clock, so as not to completely undershoot targets in offline mode. Also, compensate for key-frame overshoots when outputting delta frames.
> * FrameDropper should not assuming incoming frame rate is 0 if no frames have been seen.
> * Fix a bunch of test cases that started failing because they were relying on the fake encoder undershooting.
> * Fix test
>
> BUG=7664
>
> Review-Url: https://codereview.webrtc.org/2883963002
> Cr-Commit-Position: refs/heads/master@{#18473}
> Committed: 6431e21da6TBR=stefan@webrtc.org,holmer@google.com
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=7664
Review-Url: https://codereview.webrtc.org/2923993002
Cr-Commit-Position: refs/heads/master@{#18475}
That however exposes a bunch of failed test, so this CL also fixed a few other things:
* FakeEncoder should trust the configured FPS value rather than guesstimating itself based on the realtime clock, so as not to completely undershoot targets in offline mode. Also, compensate for key-frame overshoots when outputting delta frames.
* FrameDropper should not assuming incoming frame rate is 0 if no frames have been seen.
* Fix a bunch of test cases that started failing because they were relying on the fake encoder undershooting.
* Fix test
BUG=7664
Review-Url: https://codereview.webrtc.org/2883963002
Cr-Commit-Position: refs/heads/master@{#18473}
Return false if ReadBits fails.
Prevents GetQp from returning true with a qp of zero.
BUG=webrtc:7662
Review-Url: https://codereview.webrtc.org/2911013002
Cr-Commit-Position: refs/heads/master@{#18462}
Reason for revert:
Fixing and re-landing.
Original issue's description:
> Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ )
>
> Reason for revert:
> ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here.
> #include "webrtc/base/checks.h"
> ^-------------------
> The target:
> //webrtc/test:fileutils_unittests
> is including a file from the target:
> //webrtc/base:rtc_base_approved
>
> It's usually best to depend directly on the destination target.
> In some cases, the destination target is considered a subcomponent
> of an intermediate target. In this case, the intermediate target
> should depend publicly on the destination to forward the ability
> to include headers.
>
> Dependency chain (there may also be others):
> //webrtc/test:fileutils_unittests -->
> //webrtc/test:fileutils --[private]-->
> //webrtc/base:rtc_base_approved
>
>
> Original issue's description:
> > Enabling `gn check` on webrtc/test
> >
> > BUG=webrtc:6828
> > NOTRY=True
> >
> > Review-Url: https://codereview.webrtc.org/2911203002
> > Cr-Commit-Position: refs/heads/master@{#18372}
> > Committed: db5bb404b0
>
> TBR=kjellander@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6828
>
> Review-Url: https://codereview.webrtc.org/2920763002
> Cr-Commit-Position: refs/heads/master@{#18375}
> Committed: 1a6f143d07TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6828
Review-Url: https://codereview.webrtc.org/2918793002
Cr-Commit-Position: refs/heads/master@{#18376}
Reason for revert:
ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here.
#include "webrtc/base/checks.h"
^-------------------
The target:
//webrtc/test:fileutils_unittests
is including a file from the target:
//webrtc/base:rtc_base_approved
It's usually best to depend directly on the destination target.
In some cases, the destination target is considered a subcomponent
of an intermediate target. In this case, the intermediate target
should depend publicly on the destination to forward the ability
to include headers.
Dependency chain (there may also be others):
//webrtc/test:fileutils_unittests -->
//webrtc/test:fileutils --[private]-->
//webrtc/base:rtc_base_approved
Original issue's description:
> Enabling `gn check` on webrtc/test
>
> BUG=webrtc:6828
> NOTRY=True
>
> Review-Url: https://codereview.webrtc.org/2911203002
> Cr-Commit-Position: refs/heads/master@{#18372}
> Committed: db5bb404b0TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6828
Review-Url: https://codereview.webrtc.org/2920763002
Cr-Commit-Position: refs/heads/master@{#18375}
Before this CL, the RTP state would be re-randomized after a
recreation of VideoSendStream. That might lead to us sending
a non-compliant RTP stream, which is avoided after the
changes in this CL.
BUG=webrtc:5654
TBR=pbos@webrtc.org # Trivial change to fuzzer.
Review-Url: https://codereview.webrtc.org/2912713002
Cr-Commit-Position: refs/heads/master@{#18322}
The change is now compatible with the old JVM::Initialize API. The
context is passed to the ContextUtils class when calling its deprecated
signature.
BUG=webrtc:7665
NOTRY=True # Only comment changes since the last patchset.
Review-Url: https://codereview.webrtc.org/2903253004
Cr-Commit-Position: refs/heads/master@{#18268}
This functionality already exists for video streams, so not having it
for audio is unexpected and has lead to problems.
BUG=webrtc:7631
Review-Url: https://codereview.webrtc.org/2887733002
Cr-Commit-Position: refs/heads/master@{#18231}
probing_interval as a name is used for the period that BWE attempt to increase its estimate. The name is confusing since it is not related to "probing" which is a special mechanism for estimating BWE.
BUG=None
Review-Url: https://codereview.webrtc.org/2888893002
Cr-Commit-Position: refs/heads/master@{#18203}
Reason for revert:
Downstream project now fixed.
Original issue's description:
> Revert of Reduce dependencies on rtc::FileSystem in FileRotatingStream tests, adding helpers in webrtc::test:: (patchset #7 id:120001 of https://codereview.webrtc.org/2872283002/ )
>
> Reason for revert:
> Fails to compile successfully.
>
>
> Original issue's description:
> > Reduce dependencies on rtc::FileSystem in FileRotatingStream tests.
> >
> > Use webrtc::test::OutputPath instead of Filesystem::GetAppTempFolder.
> > Added functions RemoveFile and RemoveDir in the webrtc::test namespace,
> > to replace use of Filesystem::DeleteFolderAndContents.
> >
> > This makes Filesystem::DeleteFolderAndContents unused, to be deleted
> > together with related code in a followup cl.
> >
> > BUG=webrtc:7345
> >
> > Review-Url: https://codereview.webrtc.org/2872283002
> > Cr-Commit-Position: refs/heads/master@{#18173}
> > Committed: dd7b5f32b5
>
> TBR=pthatcher@webrtc.org,kjellander@webrtc.org,tommi@webrtc.org,nisse@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7345
>
> Review-Url: https://codereview.webrtc.org/2885393002
> Cr-Commit-Position: refs/heads/master@{#18180}
> Committed: deaa33d2f5TBR=pthatcher@webrtc.org,kjellander@webrtc.org,tommi@webrtc.org,ehmaldonado@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7345
Review-Url: https://codereview.webrtc.org/2885413002
Cr-Commit-Position: refs/heads/master@{#18193}
Reason for revert:
Fails to compile successfully.
Original issue's description:
> Reduce dependencies on rtc::FileSystem in FileRotatingStream tests.
>
> Use webrtc::test::OutputPath instead of Filesystem::GetAppTempFolder.
> Added functions RemoveFile and RemoveDir in the webrtc::test namespace,
> to replace use of Filesystem::DeleteFolderAndContents.
>
> This makes Filesystem::DeleteFolderAndContents unused, to be deleted
> together with related code in a followup cl.
>
> BUG=webrtc:7345
>
> Review-Url: https://codereview.webrtc.org/2872283002
> Cr-Commit-Position: refs/heads/master@{#18173}
> Committed: dd7b5f32b5TBR=pthatcher@webrtc.org,kjellander@webrtc.org,tommi@webrtc.org,nisse@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7345
Review-Url: https://codereview.webrtc.org/2885393002
Cr-Commit-Position: refs/heads/master@{#18180}
Since the RtpStreamId and RepairedRtpStreamId extensions can have variable
length, it makes no sense for them to have a constant valueSize field.
The header length calculation in RtpHeaderExtensionMap needed to be changed
for this because it previously worked with the assumption that all header
types have a constant size. Now it's the caller's job to specify the length
of the extensions that it might use.
BUG=webrtc:7433
Review-Url: https://codereview.webrtc.org/2867713003
Cr-Commit-Position: refs/heads/master@{#18179}
Use webrtc::test::OutputPath instead of Filesystem::GetAppTempFolder.
Added functions RemoveFile and RemoveDir in the webrtc::test namespace,
to replace use of Filesystem::DeleteFolderAndContents.
This makes Filesystem::DeleteFolderAndContents unused, to be deleted
together with related code in a followup cl.
BUG=webrtc:7345
Review-Url: https://codereview.webrtc.org/2872283002
Cr-Commit-Position: refs/heads/master@{#18173}
Reason for revert:
Breaking downstream projects.
Original issue's description:
> Split iOS sdk in to separate targets
>
> This CL splits the iOS sdk into separate static libraries for video,
> audio, ui, common, and peerconnection-related code. This will in the
> future make it easier to compile WebRTC without unneeded components.
>
> BUG=webrtc:4867
>
> Review-Url: https://codereview.webrtc.org/2862543002
> Cr-Commit-Position: refs/heads/master@{#18166}
> Committed: 52c83fe710TBR=magjed@webrtc.org,denicija@webrtc.org,tkchin@webrtc.org,henrika@webrtc.org,kthelgason@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4867
Review-Url: https://codereview.webrtc.org/2890513002
Cr-Commit-Position: refs/heads/master@{#18170}
