The "has_input_video_ logic" is simplified to abort AdaptUp() and
AdaptDown() directly rather than in each calling place of the
VideoSourceRestrictor. The intent is no change in behavior.
The degradation_preference_ is removed from the VideoSourceRestrictor
as its only usage was DCHECKing (not worth it).
ResourceAdaptationModuleInterface gets SetHasInputVideo() and
SetDegradationPreference(), making these things controllable without
knowing implementation details.
StartCheckForOveruse() and StopCheckForOveruse() are renamed to
StartResourceAdaptation() and StopResourceAdaptation().
Bug: webrtc:11222
Change-Id: Id2d7f34d427dfb3ecd4831b1a245d07becae6520
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166173
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30296}
This is a reland of f3aa6326b8e21f627b9fba72040122723251999b
Original change's description:
> Replace the ExperimentalAgc config with the new config format
>
> This CL replaces the use of the ExperimentalAgc config with
> using the new config format.
>
> Beyond that, some further changes were made to how the analog
> and digital AGCs are initialized/called. While these can be
> made in a separate CL, I believe the code changes becomes more
> clear by bundling those with the replacement of the
> ExperimentalAgc config.
>
> TBR: saza@webrtc.org
> Bug: webrtc:5298
> Change-Id: Ia19940f3abae048541e6716d0184b4caafc7d53e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163986
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30149}
Bug: webrtc:5298
Change-Id: I6db03628ed3fa2ecd36544fe9181dd8244d7e2df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165760
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30295}
This is a reland of b70c5c5ce97e7dcf2e1d8453f5ea0639d4b60453
Original change's description:
> Using simulated rtc::Thread for peer connection scenario tests.
>
> Bug: webrtc:11255
> Change-Id: I5d29e997a7209ffc64595082358cca9b2115d07a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165689
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30258}
Bug: webrtc:11255
Change-Id: If65cd56b59158cebec5609407a721fbdb47cfd1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166046
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30294}
This reverts commit c0f25cf762a6946666c812f7a3df3f0a7f98b38d.
Reason for revert: breaks negotiation with send-only clients
(webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264]
(peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER)
(peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters.
Original change's description:
> Distinguish between send and receive codecs
>
> Even though send and receive codecs may be the same, they might have
> different support in HW. Distinguish between send and receive codecs
> to be able to keep track of which codecs have HW support.
>
> Bug: chromium:1029737
> Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30284}
TBR=steveanton@webrtc.org,kron@webrtc.org
Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30292}
The test previously assumed that doing an offer/answer exchange would
leave the ICE state completed which is unlikely in practice but probably
worked most of the time in test since the network components were faked.
Bug: webrtc:11280
Change-Id: I9bc0e1490b0b8401cc832b73da9dc7fe870bc9fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166400
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30289}
Even though send and receive codecs may be the same, they might have
different support in HW. Distinguish between send and receive codecs
to be able to keep track of which codecs have HW support.
Bug: chromium:1029737
Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30284}
The idea to communicate spatial dependencies with spatial layers bitmask
wasn't fully implemented and was dropped in later version of the descriptor.
Bug: webrtc:10342
Change-Id: I1ed191c3a2a9d2e1e9ddf313f781ca8257c34dfa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166165
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30278}
The test is likely timing out on iOS simulator (see bug). Maybe I'm
going a bit overboard here :) if you want to keep all the cases I
removed, you can run some cases in one test method and the others in
another test method. Are the cases I removed particularly important?
Bug: webrtc:11284
Change-Id: I8f2e8830f931594c3471d1c20a2654e258b9fcf0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166169
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30277}
This reduces indirection and makes it easier to follow code. It also
fits into a long term strategy of reducing the scope of ChannelSend.
Bug: webrtc:9883
Change-Id: I2661c4aa6c561f7691beaaa289636254f7a58b72
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166042
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30273}
This reverts commit d61338fa6ed957dd992f25da4844db34b14f89c7.
Reason for revert: Causing a build break:
webrtc/call/BUILD:300:1: Undeclared inclusion(s) in rule 'webrtc/call:rtp_sender':
this rule is missing dependency declarations for the following files included by 'call/rtp_transport_controller_send.cc':
'webrtc/modules/congestion_controller/rtp/transport_feedback_demuxer.h'
Original change's description:
> Reland "Extracts ssrc based feedback tracking from feedback adapter."
>
> This is a reland of 08c46adc1e9f9a8d74357fe132a68906ae6e6974
>
> Original change's description:
> > Extracts ssrc based feedback tracking from feedback adapter.
> >
> > This prepares for moving TransportFeedbackAdapter to TaskQueue.
> >
> > Bug: webrtc:9883
> > Change-Id: Ib333f6a6837ff6dd8b10813e8953e4d8cd5d8633
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162040
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30076}
>
> Bug: webrtc:9883
> Change-Id: Ia74a3b1fba4d83eece9b0eb6475d6e6aecb65700
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162201
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30266}
TBR=sprang@webrtc.org,srte@webrtc.org
Change-Id: I7f3f872c7ff863a37ad8dca08051fe1e04671bfb
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166182
Reviewed-by: JT Teh <jtteh@webrtc.org>
Commit-Queue: JT Teh <jtteh@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30270}
This patch improves handshake wrt GOOG_PING support so that
- if goog_ping_enable: sender send it's goog-ping version until it gets
STUN_BINDING_RESPONSE
- receiver only sends it's goog-ping-version if getting a
goog-ping-version in the request
This means that the overhead of STUN_ATTR_GOOG_MISC_INFO is only
- added on STUN_BINDING_REQUEST until a response is received.
- added on STUN_BINDING_RESPONSE if remote peer request it.
This is wire compatible with older versions so that
- new sender will enable GOOG_PING with new/old receiver.
- old sender will enable GOOG_PING with old receiver.
- old version will not enable GOOG_PING with new receiver
(receiver expecting sender to announce first).
BUG: webrtc:11100
Change-Id: Ib3434c593988188150f4c7506918139aaf138d0c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165787
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30269}
This avoids integer overflows when size_t is 32 bits, and conforms
to style guide recommendations to avoid unsigned integers.
Also add tests for overflow on RateStatistics accumulator.
Bug: webrtc:11247
Change-Id: Ifa0db567f41bbcf3ec46d89ab888f2ed9d03f3f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163991
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#30268}
The test sometimes failed because thread creation on some platforms on
internal tests may take too much work.
Now checks are less strict.
Bug: none
Change-Id: Ibd3df02bda26b0c5e804360a909c61afa760b3bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165960
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30267}
This is a reland of 08c46adc1e9f9a8d74357fe132a68906ae6e6974
Original change's description:
> Extracts ssrc based feedback tracking from feedback adapter.
>
> This prepares for moving TransportFeedbackAdapter to TaskQueue.
>
> Bug: webrtc:9883
> Change-Id: Ib333f6a6837ff6dd8b10813e8953e4d8cd5d8633
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162040
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30076}
Bug: webrtc:9883
Change-Id: Ia74a3b1fba4d83eece9b0eb6475d6e6aecb65700
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162201
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30266}
This reverts commit b0e0728159f07269a875c5b53658603cf6733480.
Reason for revert:
Causes Chromium tests to timeout, preventing rolls into Chromium.
Original change's description:
> Replaces SynchronousMethodCall with rtc::Thread::Invoke.
>
> Given that we already have Thread:.Invoke that can be used with lambda,
> SynchronousMethodCall doesn't add any value.
>
> This simplification prepares for simulated time peer connection tests.
>
> Bug: webrtc:11255
> Change-Id: I478a11f15e30e009dae4a3fee2120f6d7a03355f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165683
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30217}
TBR=steveanton@webrtc.org,srte@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:11255
Change-Id: I9d3aa218013129db7a09a77500a0547ce9ae341a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166047
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30265}
This moves SetHasInputVideoAndDegradationPreference() to the encoder
queue. OveruseFrameDetectorResourceAdaptationModule is now entirely
single-threaded, including its inner class VideoSourceRestrictor.
VideoStreamEncoder now protects the module with RTC_GUARDED_BY. This
ensures it is safely used, even without a SequenceChecker inside of the
module. The module's |encoder_queue_| is removed.
The one task queue reference that is needed - passing down the current
task queue to StartCheckForOveruse() - is replaced by a TaskQueueBase*
(instead of rtc::TaskQueue*), enabling obtaining the current queue with
TaskQueueBase::Current(). (There is no rtc::TaskQueue::Current().)
Furthermore, the only uses of VideoSourceSinkController that isn't on
the encoder queue are documented, with a TODO saying if these are moved
the VideoSourceSinkController could also be made single-threaded.
However since this requires introducing a delay to
VideoStreamEncoder::SetSource() and VideoStreamEncoder::Stop(),
arguably a more risky change, if this is to be attempted that should be
in a separate CL.
Bug: webrtc:11222
Change-Id: I448ca5125708d5f66b95b0b180d6d24cc356dfa9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165783
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30263}
After experimentation, not pacing audio is better. This is controlled
by the field trial WebRTC-Pacer-BlockAudio. This change keeps the flag,
but changes the behaviour such that it defaults to Disabled. However,
audio can still be paced if one chooses by enabling the field trial.
Bug: webrtc:11257
Change-Id: I5b23a82bb6708c007cf8dfb40065c821eefdc4e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165381
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30262}
This .wav file is an implicit data dependency, this CL adds this
information to the build system.
Bug: None
Change-Id: Ia953e63d4658debce3cecb93bb1f3e749fe52f54
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166044
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30257}
This prepares for introducing a simulated time rtc::ThreadManager
implementation that will run on a single underlying thread.
Bug: webrtc:11255
Change-Id: I793128cc0b8e649a3675914de67dfee3298b446a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165765
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30256}
This ensures more end to end test coverage of the feature and captures
a wider class of regression then the existing unit test.
Bug: webrtc:9883
Change-Id: I6e74e571500c5c5d74caf8f661cac08bee8934f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164461
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30252}
This CL was generated by running:
git ls-files | grep ".cc" | xargs perl -i -ne 'BEGIN {undef $/}; s/("[\s\n]*<<[\s\n]*")/" "/g; print;'; git cl format
After that I manually edited modules/audio_processing/gain_controller2.cc to preserve its original
formatting.
This primary benefit of this change is a small reduction in binary size.
Bug: None
Change-Id: I689fa7ba9c717c314bb167e5d592c3c4e0871e29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165961
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30251}
Using RTX SSRCs and payload type for retransmission of video. This
corresponds to the behavior when using the peer connection API.
Bug: webrtc:9883
Change-Id: Ic0e3964d097f42219ca225513a4bc771d70ccaa4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164460
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30248}
