This changeset adds the ability for API users to enable or disable GCM Cipher
suites from objective-c.
Bug: chromium:713701
Change-Id: I0ac7b60f55dd56bebbcfb315a542ef4843099802
Reviewed-on: https://webrtc-review.googlesource.com/89263
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24028}
WebRTC internal headers are always included starting from the root
(e.g. #include "modules/audio_coding/..."), so there is no need to
specify the include_dirs removed by this CL.
Bug: webrtc:9538
Change-Id: I91e70508c67020bbf70304df5e48ca757ad43221
Reviewed-on: https://webrtc-review.googlesource.com/89385
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24026}
Previously, this would crash with UnsupportedOperationException. Allows
still calling this while the method is deprecated.
Bug: webrtc:9536, webrtc:7925
Change-Id: I7b88cecca7a4e6f505c7211cf2eb576c394973f8
Reviewed-on: https://webrtc-review.googlesource.com/89381
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24025}
The current default behavior is unchanged and points to MAINTAIN_FRAMERATE,
meaning there is no way to currently use BALANCED as we can't detect
when the value as been set or not.
Updating this is an API change that should be done in another CL and
properly communicated first.
Bug: webrtc:7607
Change-Id: Ic3877ad8dd7bc418296f21a04bc37f59ec55934a
Reviewed-on: https://webrtc-review.googlesource.com/88766
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24024}
This reverts commit 06f66c72600e58438ba9caf9f523e00a519ef3c0.
Reason for revert: Breaks downstream project.
Original change's description:
> Removing unneeded dependency.
>
> The //audio build target does not depend on the
> builtin_audio_encoder_factory, this CL removes it from the dependency
> list in order to avoid to propagate symbols that are not supposed to
> be there.
>
> Bug: webrtc:9528
> Change-Id: Ib3868ee93f61057f61283faaa83e0633ebfdea90
> Reviewed-on: https://webrtc-review.googlesource.com/89002
> Reviewed-by: Yves Gerey <yvesg@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24019}
TBR=mbonadei@webrtc.org,ossu@webrtc.org,yvesg@webrtc.org
Change-Id: Icf8f0ad4e7f5cce96fa1c0491a281ef2fd2e713f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9528
Reviewed-on: https://webrtc-review.googlesource.com/89400
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24023}
After detecting overuse of the network capacity, the target
bitrate is reduced. Normally, this should happen at most once
per RTT to prevent repeated reductions from the same overuse
signal. This CL fixes a bug that allowed repeated reductions
if an overuse was detected before it had the first reliable
throughput measurement.
The fix is guarded by a field trial. To enable the fix, use
WebRTC-BweInitialBackOffInterval/Enabled-200/
Bug: webrtc:9493
Change-Id: Iae566227fd94ebb8a4449406572158a8b79d9c53
Reviewed-on: https://webrtc-review.googlesource.com/88765
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24021}
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.
Bug: webrtc:9251, webrtc:163
Change-Id: I118156a4f9b00d8c4c4f199a5af50c494e31c34a
Reviewed-on: https://webrtc-review.googlesource.com/89343
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24020}
The //audio build target does not depend on the
builtin_audio_encoder_factory, this CL removes it from the dependency
list in order to avoid to propagate symbols that are not supposed to
be there.
Bug: webrtc:9528
Change-Id: Ib3868ee93f61057f61283faaa83e0633ebfdea90
Reviewed-on: https://webrtc-review.googlesource.com/89002
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24019}
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.
Bug: webrtc:9251, webrtc:163
Change-Id: I6d3b45de9dca3a5a04f0cdd5583919d35a585a7e
Reviewed-on: https://webrtc-review.googlesource.com/89043
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24018}
PeerConnectionInterface.
This allows the implementations of PeerConnectionInterface to deprecate
this method.
Bug: None
Change-Id: I54b56206ebac2486f112e09137c9def225683297
Reviewed-on: https://webrtc-review.googlesource.com/89261
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24011}
- Move PayloadRouter to RtpTransportControllerInterface.
- Move RetransmissionLimiter inside RtpTransportControllerSend from
VideoSendStreamImpl.
- Move video RTP specifics into PayloadRouter, in particular ownership
of the RTP modules.
- PayloadRouter now contains all video specific RTP code, and will be
renamed in a follow-up to VideoRtpSender.
- Introduce VideoRtpSenderInterface.
Bug: webrtc:9517
Change-Id: I1c7b293fa6f9c320286c80533b3c584498034a38
Reviewed-on: https://webrtc-review.googlesource.com/88240
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24009}
Full path is specified because otherwise the inner class from
VideoCapturer is used instead.
Bug: webrtc:9496
Change-Id: I122e6525101594863d506eb3c12359b5648d935e
Reviewed-on: https://webrtc-review.googlesource.com/89042
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24006}
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.
Bug: webrtc:9251
Change-Id: I1607df2a3ad177e2f3023156eb8cf37857ae06ba
Reviewed-on: https://webrtc-review.googlesource.com/89041
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24002}
Previously, the fuzzer read a int16_t and converted to float. That is
how float audio samples were generated. This CL changes the fuzzer to
read floats directly, and then sanitize them.
Bug: webrtc:7820
Change-Id: Icc526611466c10dd4222b19a4d4b4fd26643812a
Reviewed-on: https://webrtc-review.googlesource.com/85343
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24001}
The fuzzer figured out that 3 bytes is enough to fuzz a package.
2 bytes for packet length, and 1 byte of actual packet. A 20K test case
can generate > 6000 packets. It does not seem like efficient fuzzing.
This CL simply stops execution when 200 packets have been generated.
That corresponds to 4 seconds of 20 ms packets.
Bug: chromium:840115
Change-Id: Id2742a6f8021134bacd8a6e8c71b32f20c7f1086
Reviewed-on: https://webrtc-review.googlesource.com/88566
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24000}
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.
Bug: webrtc:9251
Change-Id: I6278b69f4a009fd1d0e265ebcaa3734d33cfc2e7
Reviewed-on: https://webrtc-review.googlesource.com/88764
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23998}
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.
Bug: webrtc:9251
Change-Id: Ibdafc0bb08de1be7189af7053a67a24e3a26bd6b
Reviewed-on: https://webrtc-review.googlesource.com/89001
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23997}
This reverts commit c2406e4eaf7703c6c64d21318186adda791e09fd.
Reason for revert: Reland by removing the conflict with the broken CL.
Original change's description:
> Revert "Move allocation and rtp conversion logic out of payload router."
>
> This reverts commit 1da4d79ba3275b3fa48cad3b2c0949e0d3b7afe7.
>
> Reason for revert: Need to revert https://webrtc-review.googlesource.com/c/src/+/88220
>
> This causes a merge conflict. So need to revert this first.
>
> Original change's description:
> > Move allocation and rtp conversion logic out of payload router.
> >
> > Makes it easier to write tests, and allows for moving rtp module
> > ownership into the payload router in the future.
> >
> > The RtpPayloadParams class is split into declaration and definition and
> > moved into separate files.
> >
> > Bug: webrtc:9517
> > Change-Id: I8700628edff19abcacfe8d3a20e4ba7476f712ad
> > Reviewed-on: https://webrtc-review.googlesource.com/88564
> > Commit-Queue: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#23983}
>
> TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org
>
> Change-Id: I342c4bf483d975c87c706fe7f76f44e2dc60fe4c
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9517
> Reviewed-on: https://webrtc-review.googlesource.com/88821
> Reviewed-by: JT Teh <jtteh@webrtc.org>
> Commit-Queue: JT Teh <jtteh@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23991}
TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,lliuu@webrtc.org,jtteh@webrtc.org,tkchin@webrtc.org
Change-Id: I154145cdbc668feee86dbe78860147a6954fee6c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9517
Reviewed-on: https://webrtc-review.googlesource.com/89020
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23996}
In current state, if you want to do something with the capturer (eg. switch to next camera again) it fails with an exception that camera switch is already in progress.
Change-Id: I908eb590b54fdf3346441097b39f1f2a2eb56ce8
Bug: webrtc:9527
Change-Id: I908eb590b54fdf3346441097b39f1f2a2eb56ce8
Reviewed-on: https://webrtc-review.googlesource.com/88700
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23995}
This reverts commit 1da4d79ba3275b3fa48cad3b2c0949e0d3b7afe7.
Reason for revert: Need to revert https://webrtc-review.googlesource.com/c/src/+/88220
This causes a merge conflict. So need to revert this first.
Original change's description:
> Move allocation and rtp conversion logic out of payload router.
>
> Makes it easier to write tests, and allows for moving rtp module
> ownership into the payload router in the future.
>
> The RtpPayloadParams class is split into declaration and definition and
> moved into separate files.
>
> Bug: webrtc:9517
> Change-Id: I8700628edff19abcacfe8d3a20e4ba7476f712ad
> Reviewed-on: https://webrtc-review.googlesource.com/88564
> Commit-Queue: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23983}
TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Change-Id: I342c4bf483d975c87c706fe7f76f44e2dc60fe4c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9517
Reviewed-on: https://webrtc-review.googlesource.com/88821
Reviewed-by: JT Teh <jtteh@webrtc.org>
Commit-Queue: JT Teh <jtteh@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23991}
Currently the codec specific max bitrate that is set in the SDP
gets overridden by the value set with the "b=AS" attribute
(WebRtcVideoChannel::SetSendParameters). But at the
WebRtcVideoSendStream level it does the opposite - the codec
specific max bitrate value overrides the values that could be
set by RtpParameters or the "b=AS" value
(in WebRtcVideoSendStream::CreateVideoEncoderConfig). This change
updates the logic to be consistent with what happens at the
WebRtcVideoChannel level, and allows the RtpParameter max bitrate
to override the codec specific max bitrate.
Bug: webrtc:8655
Change-Id: I3f0347cb7cffcfc577484231b061ab0712453e69
Reviewed-on: https://webrtc-review.googlesource.com/88520
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23989}
This CL is the first step for introducing color space information in webrtc.
- Add ColorSpace class listing color profiles.
- Add ColorSpace as a member of webrtc::VideoFrame.
- Make use of this class by extracting info from VP9 decoder.
Bug: webrtc:9522
Change-Id: I5e2514efee2a193bddb4459261387f2d40e936ad
Reviewed-on: https://webrtc-review.googlesource.com/88540
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23988}
This CL re-activates the explicit handling of microphone
gain changes in the AEC3 code. The implementation is done
beneath a kill-switch so that when that switch is active
the changes in this CL are bitexact.
Bug: webrtc:9526,chromium:863826
Change-Id: I58e93d8bc0bce7bec91e102de9891ad48ebc55d8
Reviewed-on: https://webrtc-review.googlesource.com/88620
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23986}
Makes it easier to write tests, and allows for moving rtp module
ownership into the payload router in the future.
The RtpPayloadParams class is split into declaration and definition and
moved into separate files.
Bug: webrtc:9517
Change-Id: I8700628edff19abcacfe8d3a20e4ba7476f712ad
Reviewed-on: https://webrtc-review.googlesource.com/88564
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23983}
