For HDR codecs, we expect to receive input that has 10-bit color depth. But
currently, WebRTC assumes only 8-bit input and output. This CL adds k010
format that represent this input.
Bug: webrtc:9376
Change-Id: Ie7df64b0eddb0752b493e0457a49083a1e87ba51
Reviewed-on: https://webrtc-review.googlesource.com/81920
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23749}
This can happen with the following sequence of API calls:
1) AddTrack(track) + offer/answer
2) RemoveTrack(track's sender) + offer/answer
3) AddTrack(same track)
Since the first transceiver had already been used to send, it will
not get re-used by the second call to AddTrack. Another RtpSender
will be created with its ID = the track ID. But the code hits a
DCHECK when CreateOffer is later called since both m= sections will
offer the same track ID component of the MSID.
The fix implemented here is to randomly generate a sender ID if
there is already an RtpSender with the track's ID.
Bug: webrtc:8734
Change-Id: Ic2dda23d66e364e77ff7505e1c37e53105a17dae
Reviewed-on: https://webrtc-review.googlesource.com/84249
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23748}
The frequency shape of the echo path has been included in the reverberation model.
Bug: webrtc:9454,chromium:856636
Change-Id: Id2bc3096df31e29328936f94fe965ed1883d70f7
Reviewed-on: https://webrtc-review.googlesource.com/85370
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23746}
This removes call of av_register_all(), which is deprecated, and
related code.
Bug: webrtc:9352
Change-Id: Ib7de5931c900eaf1023ecf3046f560feaaeec8ef
Reviewed-on: https://webrtc-review.googlesource.com/85347
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23743}
common_audio/fft4g.c is third party codem that have to be moved into
third_party folder, so to be able to do it at first we have to extract
it into separate target. It is extracted with corresponding header file
and will be moved in futher CL.
Bug: webrtc:8366
Change-Id: I586ca94d4e9242c23163b987fa334dfa705020ed
Reviewed-on: https://webrtc-review.googlesource.com/85372
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23742}
When GetSvcConfig returned fewer spatial layers than the number
statically configured from the test, we would crash on a SIGFPE.
This is not a problem in the production code, since there we
reset the encoder with the correct number of spatial layers
whenever the resolution changes.
Bug: None
Change-Id: I339e4a3c0fa993c7c649533c0eae71e1314194e7
Reviewed-on: https://webrtc-review.googlesource.com/85374
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23741}
This slightly increases fuzzer coverage of the APM.
(.25 % points more line coverage.)
Bug: webrtc:9413
Change-Id: Ic992423f1dcf34fa0aa9649c8035a8e48b0ccdb2
Reviewed-on: https://webrtc-review.googlesource.com/85342
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23732}
Before this CL, the packetsLost and jitter stats (as returned by
GetStats, at the API level) were only being updated when an RTCP SR or
RR is generated. According to the stats spec, "local" stats like this
should be updated any time a packet is received.
This CL also fixes some minor issues with the calculation of packetsLost
(and fractionLost):
* Packets weren't being count as lost if lost over a sequence number
rollover.
* Temporary periods of "negative" loss (caused by duplicate or out of
order packets) weren't being accumulated into the cumulative loss
counter. Example:
Period 1: Received packets 1, 2, 4
Loss over that period: 1 (expected 4 packets, got 3)
Reported cumulative loss: 1
Period 2: Received packets 3, 5
Loss over that period: -1 (expected 1 packet, got 2)
Reported cumulative loss: 1 (should be 0!)
Landing with NOTRY because Android compile bots are broken for an
unrelated reason.
NOTRY=True
Bug: webrtc:8804
Change-Id: I840ba34de8957b1276f6bdaf93718f805629f5c8
Reviewed-on: https://webrtc-review.googlesource.com/50020
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23731}
This change also standardizes the RtpSender to a single constructor
and moves the |track| and |stream_ids| arguments to setter methods.
Bug: webrtc:8734
Change-Id: I227a84868a80797f6cc2a1af6eec6d76da8ea159
Reviewed-on: https://webrtc-review.googlesource.com/84248
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@{#23730}
The FrameCombiner sub-module of the AudioMixer uses one of two
limiters. One is an AudioProcessingModule with AGC1 enabled and
configured as a limiter. The other is the limiter part of AGC2. This
change removes the APM-AGC1 limiter. This requires small changes to
FrameCombiner, AudioMixerImpl and tests.
We also stop using the finch experiment flag.
Bug: webrtc:8925
Change-Id: Id7b8349ec4720b6417b15eaf70ed1a850b6ddbed
Reviewed-on: https://webrtc-review.googlesource.com/84620
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23727}
Also move StringStream to the only test using it.
Bug: webrtc:6424
Change-Id: Iad79c7becaa2764ac954c18711eaae4faf46ae72
Reviewed-on: https://webrtc-review.googlesource.com/84320
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23721}
This reverts commit 056a68da896d9a578b9ea83e56d261648ea0adc6.
Reason for revert: Trying to reland.
Original change's description:
> Revert "Enable any address ports by default."
>
> This reverts commit f04148c810aad2a0809dc8978650c55308381c47.
>
> Reason for revert: Speculative revert. I suspect this is breaking a
> downstream test (I'll reland if it is not the culprit).
>
> Original change's description:
> > Enable any address ports by default.
> >
> > Ports not bound to any specific network interface are allocated by
> > default. These any address ports are pruned after allocation,
> > conditional on the allocation results of normal ports that are bound to
> > the enumerated interfaces.
> >
> > Bug: webrtc:9313
> > Change-Id: I3ce12eeab0cf3547224e5f8c188d061fc530e145
> > Reviewed-on: https://webrtc-review.googlesource.com/78383
> > Commit-Queue: Qingsi Wang <qingsi@google.com>
> > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#23673}
>
> TBR=deadbeef@webrtc.org,pthatcher@webrtc.org,qingsi@google.com
>
> Change-Id: I3b3dc42c7de46d198d4b9c270020dcf1100dd907
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9313
> Reviewed-on: https://webrtc-review.googlesource.com/84300
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23678}
TBR=deadbeef@webrtc.org,mbonadei@webrtc.org,pthatcher@webrtc.org,qingsi@google.com
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9313
Change-Id: I98442346babb5d8953d37dc5825efaf79804ed7f
Reviewed-on: https://webrtc-review.googlesource.com/85000
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23720}
Added distinction between number of configured and number of actively
encoded spatial layers and include number of actively encoded spatial
layers in ssData. Modified layer_filtering_transport.cc test to
parse from the RTP header and use the number of actively encoded
spatial layers for filtering spatial video layers.
Bug: webrtc:9425
Change-Id: Ic9f8895ab08b0626f9bb53a75ec33d8e7eb8706e
Reviewed-on: https://webrtc-review.googlesource.com/84243
Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23716}
* Move SimulcastEncoderAdapter out under modules/video_coding
* Move SimulcastRateAllocator back out to modules/video_coding/utility
* Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility
* Move any VP8 specific code - such as temporal layer bitrate budgeting -
under codec type dependent conditionals.
* Plumb the simulcast index for H264 in the codec specific and RTP format data structures.
TBR=sprang@webrtc.org,stefan@webrtc.org,titovartem@webrtc.org
Bug: webrtc:5840
Change-Id: I2d3b130622dd7ceec5528f3ab6c46f109e6bafb8
Reviewed-on: https://webrtc-review.googlesource.com/84743
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23715}
The purpose is to make the fixture reusable in downstream
projects. The CL adds the following things to API:
- api/test/video_quality_test_fixture.h
- api/test/create_video_quality_test_fixture.h
The following things are moved to API:
- call/bitrate_constraints.h (api/bitrate_constraints.h)
- call/simulated_network.h (api/test/simulated_network.h)
- call/media_type.h (api/mediatypes.h)
These are required by the params struct passed to the
fixture. I didn't attempt to split the params struct into
an internal-only and public version in this CL, and as
a result we need to pull in the above things. They are
quite harmless though, so I think it's worth it in order
to avoid splitting up the test config struct.
This CL doesn't solve all the problems we need to
implement downstream tests; we probably need to upstream
tracing variants of FakeNetworkPipe for instance, but
that will come later. This puts in place the basic
structure for now.
Bug: None
Change-Id: I35e26ed126fad27bc7b2a465400291084f6ac911
Reviewed-on: https://webrtc-review.googlesource.com/69601
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23714}
This is a reland of 21219a0e43446701810236fb9fdd59be072c12df
The default implementation of OnLogMessage(msg, sev, tag) discarded
the tag, resulting in FileRotatingLogSink not receiving tags.
Since the revert the default implementation of
OnLogMessage(msg, sev, tag) has been updated to add the tag to the log
message. A more efficient implementation of it has also been added for
FileRotatingLogSink.
Unit tests are added for the default implementation and for Loggable
injection.
Original change's description:
> Reland "Injectable logging"
>
> Any injected loggable or NativeLogger would be deleted if PCFactory
> was reinitialized without calling setInjectableLogger. Now native
> logging is not implemented as a Loggable, so it will remain active
> unless a Loggable is injected.
>
> This is a reland of 59216ec4a4151b1ba5478c8f2b5c9f01f4683d7f
>
> Original change's description:
> > Injectable logging
> >
> > Allows passing a Loggable to PCFactory.initializationOptions, which
> > is then injected to Logging.java and logging.h. Future log messages
> > in both Java and native will then be passed to this Loggable.
> >
> > Bug: webrtc:9225
> > Change-Id: I2ff693380639448301a78a93dc11d3a0106f0967
> > Reviewed-on: https://webrtc-review.googlesource.com/73243
> > Commit-Queue: Paulina Hensman <phensman@webrtc.org>
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#23241}
>
> Bug: webrtc:9225
> Change-Id: I2fe3fbc8c323814284bb62e43fe1870bdab581ee
> TBR: kwiberg
> Reviewed-on: https://webrtc-review.googlesource.com/77140
> Commit-Queue: Paulina Hensman <phensman@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23310}
Bug: webrtc:9225
Change-Id: I67a5728fe772f0bedc9509713ed8b8ffdc31af81
TBR: kwiberg
Reviewed-on: https://webrtc-review.googlesource.com/80860
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23711}
This reverts commit 07efe436c9002e139845f62486e3ee4e29f0d85b.
Reason for revert: Breaks downstream project.
cricket::GetSimulcastConfig method signature has been updated.
I think you can get away with a default value for temporal_layers_supported (and then you can remove it after a few days when projects will be updated).
Original change's description:
> Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8.
>
> * Move SimulcastEncoderAdapter out under modules/video_coding
> * Move SimulcastRateAllocator back out to modules/video_coding/utility
> * Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility
> * Move any VP8 specific code - such as temporal layer bitrate budgeting -
> under codec type dependent conditionals.
> * Plumb the simulcast index for H264 in the codec specific and RTP format data structures.
>
> Bug: webrtc:5840
> Change-Id: Ieced8a00e38f273c1a6cfd0f5431a87d07b8f44e
> Reviewed-on: https://webrtc-review.googlesource.com/64100
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23705}
TBR=sprang@webrtc.org,stefan@webrtc.org,mflodman@webrtc.org,hta@webrtc.org,sergio.garcia.murillo@gmail.com,titovartem@webrtc.org,agouaillard@gmail.com
Change-Id: Ic9d3b1eeaf195bb5ec2063954421f5e77866d663
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:5840
Reviewed-on: https://webrtc-review.googlesource.com/84760
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23710}
Fuzzes the config and audio inputs to GainControlImpl.
Seems able to cover a few hundred lines of code that the APM fuzzer hasn't been able to reach.
Bug: webrtc:9413
Change-Id: I32776505be9c416ec03113c12437a92dcfadd827
Reviewed-on: https://webrtc-review.googlesource.com/84589
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23709}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameter 'api'
Then undo changes to optional target itself and optional_unittests
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I44093da213369d6a502e33792c694f620f53b779
Reviewed-on: https://webrtc-review.googlesource.com/84621
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23707}
This reverts commit 80c4cca4915dbc6094a5bfae749f85f7371eadd1.
Reason for revert: Breaks downstream tests.
Original change's description:
> NetEq: Deprecate playout modes Fax, Off and Streaming
>
> The playout modes other than Normal have not been reachable for a long
> time, other than through tests. It is time to deprecate them.
>
> The only meaningful use was that Fax mode was sometimes set from
> tests, in order to avoid time-stretching operations (accelerate and
> pre-emptive expand) from messing with the test results. With this CL,
> a new config is added instead, which lets the user specify exactly
> this: don't do time-stretching.
>
> As a result of Fax and Off modes being removed, the following code
> clean-up was done:
> - Fold DecisionLogicNormal into DecisionLogic.
> - Remove AudioRepetition and AlternativePlc operations, since they can
> no longer be reached.
>
> Bug: webrtc:9421
> Change-Id: I651458e9c1931a99f3b07e242817d303bac119df
> Reviewed-on: https://webrtc-review.googlesource.com/84123
> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23704}
TBR=henrik.lundin@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org
Change-Id: I555aae8850fc4ac1ea919bfa72c11b5218066f30
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9421
Reviewed-on: https://webrtc-review.googlesource.com/84680
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23706}
* Move SimulcastEncoderAdapter out under modules/video_coding
* Move SimulcastRateAllocator back out to modules/video_coding/utility
* Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility
* Move any VP8 specific code - such as temporal layer bitrate budgeting -
under codec type dependent conditionals.
* Plumb the simulcast index for H264 in the codec specific and RTP format data structures.
Bug: webrtc:5840
Change-Id: Ieced8a00e38f273c1a6cfd0f5431a87d07b8f44e
Reviewed-on: https://webrtc-review.googlesource.com/64100
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23705}
The playout modes other than Normal have not been reachable for a long
time, other than through tests. It is time to deprecate them.
The only meaningful use was that Fax mode was sometimes set from
tests, in order to avoid time-stretching operations (accelerate and
pre-emptive expand) from messing with the test results. With this CL,
a new config is added instead, which lets the user specify exactly
this: don't do time-stretching.
As a result of Fax and Off modes being removed, the following code
clean-up was done:
- Fold DecisionLogicNormal into DecisionLogic.
- Remove AudioRepetition and AlternativePlc operations, since they can
no longer be reached.
Bug: webrtc:9421
Change-Id: I651458e9c1931a99f3b07e242817d303bac119df
Reviewed-on: https://webrtc-review.googlesource.com/84123
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23704}
rtc_base/base64.h and rtc_base/base64.cc are third_party files in webrtc
repo and we are planning to move them into third_party folder. To be
able to do it, we need to eliminate all dependencies from this files
on webrtc code base. So thic CL will remove dependency on
rtc_base.checks.h. After it we will be able to move base64 files.
Bug: webrtc:8366
Change-Id: I79d252daef974e95bb43a1566d05e603922aea3f
Reviewed-on: https://webrtc-review.googlesource.com/84126
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23702}
This involves treating it just like "detailed", for now.
At a later stage we might want to modify codec parameters for it.
Bug: chromium:852701
Change-Id: I24678e1f7711bf03ca22273afaaf338e9e3ba1fe
Reviewed-on: https://webrtc-review.googlesource.com/83582
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Peter Boström <pbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23701}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameter 'rtc_base'
Then manually fix where Optional was used without rtc prefix (patchset#3)
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I825f80cc8089747876ba6316d9e7c30e05716974
Reviewed-on: https://webrtc-review.googlesource.com/84585
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23700}
