Since the pushback controller doesn't strictly adhere to the congestion
window, it better belongs together with the congestion controller logic.
Also ensuring that it does not override the configured min bitrate.
Bug: webrtc:9586
Change-Id: I57dcfc946d470247e66c67adabddaafa3d9d83ad
Reviewed-on: https://webrtc-review.googlesource.com/c/105102
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25115}
This CL replaces CHECKs and crashes with DCHECKs and default values.
Bug: webrtc:9535
Change-Id: Ib4b16421699c633d0e9ef140189861c8179450f4
Reviewed-on: https://webrtc-review.googlesource.com/c/105003
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25111}
This CL removes outdated code for testing of platforms with clock-drift
Bug: webrtc:8671
Change-Id: Ie202c514609d9f3d2357107b0daf895331275797
Reviewed-on: https://webrtc-review.googlesource.com/c/105183
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25105}
This CL ensures that the default setting for the usage of stationary signal
properties is not overridden by mistake.
Bug: chromium:894243
Change-Id: I85ab65383ee82b5f3153864da7a0cede7776c146
Reviewed-on: https://webrtc-review.googlesource.com/c/105181
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25104}
The SDP attribute extmap-allow-mixed shows that the client supports
mixing of one- and two-byte header extensions within the same stream.
This is supported on the receive side since CL "Parse two-byte header
extensions", commit 07ba2b9445525da3eabf7c188d631abf32279d98.
Bug: webrtc:7990
Change-Id: I8419da48673f513fcca21a8722614f4601a075fc
Reviewed-on: https://webrtc-review.googlesource.com/c/103420
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25098}
Basic integration of media_transport in JSepTransportController.
- Creates media_transport if media transport factory provided in jsep transport controller configuration.
- Unittest that makes sure media_transport is created with correct caller or callee setting.
- Added fake_media_transport, for now simple implementation which only stores caller/callee, but in the future fake media transport will be expanded to pass frames between two fake media_transports, which will enable audio / video integration tests.
NEXT STEPS: Once integration of media_transport with PeerConnection (https://webrtc-review.googlesource.com/c/src/+/103860) lands, we can start passing media transport factory from peer connection to jsep transport controller.
NOTE: Includes missing include change from https://webrtc-review.googlesource.com/c/src/+/103540 (otherwise this change will not compile)
Bug: webrtc:9719
Change-Id: I1e8a521beab445aa9f7ea93c8d7a537dc137d11c
Reviewed-on: https://webrtc-review.googlesource.com/c/104400
Commit-Queue: Anton Sukhanov <sukhanov@google.com>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25096}
This change prevents resets unless someone actually set a FrameEncryptor
/ FrameDecryptor.
Bug: webrtc:9795
Change-Id: I29910b9ecc2f6f8eea371c5961ac7e9780de65d2
Reviewed-on: https://webrtc-review.googlesource.com/c/104901
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25095}
After landing https://webrtc-review.googlesource.com/c/src/+/104802, it
is finally possible to remove the complexity behind
rtc_base:rtc_base_approved and switch back to one build target.
The long term vision is to remove it too, in favor of smaller and more
focues build targets.
Bug: webrtc:9838
Change-Id: Ib98dfae103a20edb8c8b6706d376ad4f3c992886
Reviewed-on: https://webrtc-review.googlesource.com/c/105041
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25093}
compare_videos.py will now print the VMAF score for each frame.
The CL also removes some stale comments.
Bug: webrtc:9642
Change-Id: I5623588580dea06dd487d7763dc3a2511bd2cd3c
Reviewed-on: https://webrtc-review.googlesource.com/c/105103
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25092}
The logic in rtc::Buffer::OnMovedFrom was backwards w.r.t.
RTC_DCHECK_IS_ON. We intended to provoke bugs when DCHECKs are on and
play it safe when DCHECKs are off, but actually we did the reverse.
This CL fixes that.
It also adds a death test that would have caught the bug.
Bug: webrtc:9856
Change-Id: Ib6a4b07d12732e5a66e93b36b885abdab93e55c7
Reviewed-on: https://webrtc-review.googlesource.com/c/105040
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25091}
I don't think this has any impact, just wanted to have a first unit
test to play around with.
Bug: None
Change-Id: I892e2642f0243c5c9ee807cf71abcd96112b25f4
Reviewed-on: https://webrtc-review.googlesource.com/c/105000
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25089}
In order to implement a WebRTC component in Chromium, WebRTC needs to
export some symbols.
If RTC_EXPORT relies on COMPONENT_BUILD to mark symbols as exported
or imported, it will not be possible to incrementally add RTC_EXPORT
because the Chromium build will break (two Chromium components that
export a symbol or no component that exports it).
By using `rtc_enable_symbol_export` (which makes GN define
WEBRTC_ENABLE_SYMBOL_EXPORT), WebRTC will be able to incrementally mark
symbols with RTC_EXPORT and flip the value of `rtc_enable_symbol_export`
in the Chromium build when everything will be ready.
Bug: webrtc:9419
Change-Id: I57ab63c53545c500eaaefd75c112b4674aa7cf19
Reviewed-on: https://webrtc-review.googlesource.com/c/104980
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25087}
This adds calls to the underlying RtpRtcp module to indicate when audio
is part of bitrate allocation. This information is propagated and set in
the packet info for each packet.
This is part of a series of CLs that allows GoogCC to track sent bitrate
that is included in bitrate allocation but without transport feedback.
Bug: webrtc:9796
Change-Id: I79b024cb7f2eb8c86421cfa34d38ef68467776c3
Reviewed-on: https://webrtc-review.googlesource.com/c/104882
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25086}
This CL removes the need of having rtc_base:rtc_base_approved_generic
and rtc_base:rtc_base_approved_objc since it removes the _objc build
target by moving the declaration of rtc::DescriptionFromOSStatus into
rtc_base/logging_mac.h in order to have a new target
rtc_base:logging_mac on which rtc_base:logging can depend on.
The target rtc_base:rtc_base_approved_generic will be removed in a
follow up CL.
Bug: webrtc:9838
Change-Id: Id93ac7bced213128e7d654694ff15337c26dab62
Reviewed-on: https://webrtc-review.googlesource.com/c/104802
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25085}
If the need arises, please use:
using std::swap;
swap(a, b);
which falls back to a generic std::swap.
Bug: webrtc:9855
Change-Id: I819839d160fc7ae289310a13e3988cdb3f0b3086
Reviewed-on: https://webrtc-review.googlesource.com/c/104100
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25084}
Since it is currently unused and not actively maintained, code under
ortc/ will be deleted by this CL.
Bug: webrtc:9824
Change-Id: I20f890b1a1e5e1dbd2b3949af916ae0a6bc8a032
Reviewed-on: https://webrtc-review.googlesource.com/c/102601
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25082}
Streams that are part of transport feedback are assumed to be part of
allocation. A SetAsPartOfAllocation method is introduced to be used by
media streams that are part of bitrate allocation but not included in
feedback.
This is part of a series of CLs that allows GoogCC to track sent bitrate
that is included in bitrate allocation but without transport feedback.
Bug: webrtc:9796
Change-Id: If7ac1ad3e6f3c28b2ada2aef1607a42689d899b2
Reviewed-on: https://webrtc-review.googlesource.com/c/104881
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25079}
In order to correctly implement RTCPeerConnectionState and RTCIceConnectionState the ice transports need to support RTCIceTransportState.
This CL adds an implementation parallel to the current non-standard IceTransportState. It's not currently used anywhere. The old implementation will remain in place until we're ready to switch RTCIceConnectionState over.
Bug: webrtc:9308
Change-Id: I30e2bbb5b4fafa410261bcd9d5e3b76c03435feb
Reviewed-on: https://webrtc-review.googlesource.com/c/103220
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25078}
This CL changes the tuning of the echo suppressor for the case when
there is echo only. The resulting effect is a slight increase of
transparency
Bug: webrtc:9844,chromium:893744
Change-Id: I5e6a867e0d03dc3a468a8f5cfa64103e001baae1
Reviewed-on: https://webrtc-review.googlesource.com/c/104760
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25075}
This CL utilizes the AEC3 ability to tailor the suppressor during
situations when the nearend dominates over the residual echo. This is
done by increasing the thresholds for transparent echo suppressor
behavior when the nearend is strong compared to the residual echo.
Bug: webrtc:9836, chromium:893744
Change-Id: Ic06569eefc7f2557b401db43b3ac84b299071294
Reviewed-on: https://webrtc-review.googlesource.com/c/104460
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25071}
This CL adds flags to the PacketOptions and PacktInfo struct that are
intended to be used to indicate if the packet belongs to a media stream
that is part of bitrate allocation as well as if it is included in
transport wide packet feedback.
This is part of a series of CLs that allows GoogCC to track sent bitrate
that is included in bitrate allocation but without transport feedback.
Bug: webrtc:9796
Change-Id: Icdf3e1e13d3f119574ee1b2c574f2d3329a7e303
Reviewed-on: https://webrtc-review.googlesource.com/c/104920
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25069}
