Use different media ssrcs for different RtcpBandwidthObservers
Bug: None
Change-Id: I1733ddfa5dcd378b700e31fd805d8930ec69064f
Reviewed-on: https://chromium-review.googlesource.com/517798
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18536}
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}
For devices with multiple cameras, all supported resolutions from both
the front-facing and back cameras are listed.
Bug: webrtc:7783
Change-Id: I228eda28ea48181c86d344413dda9f3a71b0864f
Reviewed-on: https://chromium-review.googlesource.com/529045
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Daniela Jovanoska Petrenko <denicija@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18533}
WebRtcVideoChannel and and WebRtcVideoEngine seem to have been removed, and only WebRtcVideoChannel2 and WebRtcVideoEngine2 remain, which removes the need for the "2" postfix.
BUG=None
Review-Url: https://codereview.webrtc.org/2932073002
Cr-Commit-Position: refs/heads/master@{#18531}
After landing https://chromium-review.googlesource.com/528173
only one test needs to be disabled: VirtualSocketServerTest.delay_v4
BUG=webrtc:7727
NOTRY=True
TESTED=gn gen out/x64-Debug --args='target_os="ios" ios_enable_code_signing=false is_component_build=false target_cpu="x64"'
ninja -C out/x64-Debug webrtc_nonparallel_tests
out/x64-Debug/iossim -d "iPhone 6s" -s 10.3 out/x64-Debug/webrtc_nonparallel_tests.app
Review-Url: https://codereview.webrtc.org/2909073002
Cr-Commit-Position: refs/heads/master@{#18519}
The default implementations of AudioProcessing::{AttachAecDump,
DetachAecDump} are removed and audio_processing.cc is decoupled from
aec_dump.h. After this CL, the two methods are pure virtual. The
default implementations were added because doing otherwise would break
internal projects.
Bug: webrtc:7404
Change-Id: If94f60aeefe4ad1eefed3744f857692cc645bdf4
Reviewed-on: https://chromium-review.googlesource.com/528132
Commit-Queue: Alex Loiko <aleloi@google.com>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18517}
When using the SetEncoder interface, there's no actual CodecInst to return from Channel::GetSendCodec. Before this CL, this was done by calling the ACM, which has functionality for generating a CodecInst with the necessary values even when handed an external encoder. Unfortunately, this call takes a lock and does some extra processing which isn't strictly necessary in this case. Since GetSendCodec is called inside the audio input callback code, this can cause problems.
This CL instead generates a CodecInst in the SetEncoder call and has GetSendCodec simply return that one if it's available. If it isn't the value from codec_manager_ is returned instead, as was the case before injectable audio codec related changes were added to Channel.
BUG=b/38018041
Review-Url: https://codereview.webrtc.org/2924363004
Cr-Commit-Position: refs/heads/master@{#18515}
This decoder is only used for iOS/Mac and it will simplify to make it
ObjC++ instead of C++, similar to how the encoder is .mm already.
Bug: None
Change-Id: I13f62f018432e9c23e7277eea29258a73e1590e1
Reviewed-on: https://chromium-review.googlesource.com/529084
Reviewed-by: Kári Tristan Helgason <kthelgason@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18513}
This solves an issue where if the first packet happens to be an RTX packet, it is recovered with an incorrect SSRC.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2928123002
Cr-Commit-Position: refs/heads/master@{#18512}
Make sure to call ensureGLContext before calling OpenGL functions
BUG=webrtc:7751
Review-Url: https://codereview.webrtc.org/2916583005
Cr-Commit-Position: refs/heads/master@{#18511}
This class is video-specific, and we want to free the name
"RtpStreamReceiver" so it can be reused for a media-independent RTP
receive class.
Also renames related files.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2926253002
Cr-Commit-Position: refs/heads/master@{#18510}
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}
One of our toolchains does not expose |errno| in the global namespace.
BUG=none
Review-Url: https://codereview.webrtc.org/2926273002
Cr-Commit-Position: refs/heads/master@{#18506}
This CL ensures that it is not possible to run several echo canceller
solutions in cascade inside the audio processing module.
Bug: webrtc:7776
Change-Id: I1777f97aeacb8cdf5c6c3be4bf13eefcde7d69fb
Reviewed-on: https://chromium-review.googlesource.com/527053
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18505}
This CL adds a MockAecDump and integration tests that inject the mock
into AudioProcessingImpl. The tests check the call pattern between
AudioProcessingImpl and AecDump. The existing tests ApmTest.* and
DebugDumpTest.* (not touched by this CL) check that the data written
by AecDumpImpl is valid.
The tests check that the protobuf-writing methods for the different
protobuf message types in audio_processing/debug.proto are indeed
called for the different modes of AudioProcessingImpl operation.
BUG=webrtc:7404
Review-Url: https://codereview.webrtc.org/2888533005
Cr-Commit-Position: refs/heads/master@{#18501}
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}
Reland is done after fixes in downstream projects.
TBR=sprang@webrtc.org,asapersson@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=chrome:718738
Review-Url: https://codereview.webrtc.org/2922953002
Cr-Commit-Position: refs/heads/master@{#18498}
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}
AudioMixerImpl::CreateWithOutputRateCalculator has become
deprecated. Instead, either Create() or Create(OutputRateCalculator,
bool use_limiter) should be used. The first uses sensible default
values for missing arguments. The second takes all arguments. The old
CreateWithOutputRateCalculator is deprecated so that we don't have
different Create:s with all possible combinations of parameters.
Note that the factory methods may change in the future. The reason for
adding 'use_limiter' was that the limiter that was used had
questionable benefit and was very computationally expensive. Now work
is going on to replace it with a much cheaper version. After
the change, the factory method may change again to not allow for
disabling the limiter.
Bug: webrtc:7167
Change-Id: I0f9005e27e726fa552ee38dcbe965274e5006544
Reviewed-on: https://chromium-review.googlesource.com/528074
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18496}
Make RtpDemuxer able to demux RTP packets according to RSID (RTP Stream ID), as well as the (pre-existing) ability to demux according to SSRC.
BUG=None
Review-Url: https://codereview.webrtc.org/2920993002
Cr-Commit-Position: refs/heads/master@{#18495}
Make resilience configurable in video processor integration tests.
BUG=webrtc:6783
Review-Url: https://codereview.webrtc.org/2919803002
Cr-Commit-Position: refs/heads/master@{#18493}
This reflects currently active developers of the module.
NOTRY=True
Bug: None
Change-Id: Ibc0810b08db753404fcb94038a4bd857d5585ef9
Reviewed-on: https://chromium-review.googlesource.com/528075
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18492}
This is a minor clean-up made possible by simplifications done in the
past.
Bug: none
Change-Id: Id0ea167572f8da36db5de949441f67a2a18555be
Reviewed-on: https://chromium-review.googlesource.com/528073
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18490}
Track perf for a test using 200kbps link, 5% packet loss and queue
length of 30 packets. This currently performs poorly.
BUG=webrtc:7694
Review-Url: https://codereview.webrtc.org/2930703002
Cr-Commit-Position: refs/heads/master@{#18488}
