Before this CL for SVC streams (e.g VP9) still 3 separate RTP_RTCP senders
were created. The RTCP target bitrate messages were treated as simulcast
and were split and send for each separate spatial layer in a separate SSRC.
To fix that an svc flag is now wired to VideoSendStream config
and filled based on the encoder config in WebrtcVideoEngine. This flag is
used to differentiate between simulcast and SVC mode in RtpVideoSender.
Bug: webrtc:10485
Change-Id: Ifa01d12a7d4f01fcbe448ad11e0cc39ab2d1df55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129929
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27345}
With visibility restricted to modules/video_coding/.
Also drop some unneeded dependencies on system_wrappers.
Bug: webrtc:3380
Change-Id: If3b64396953a026bede09c9fb5eb06cfc4c29f3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130104
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27344}
Chromium requires that all code that waits on a sync primitive be
annotated with ScopedAllowBaseSyncPrimitives(ForTesting). Webrtc
already imports ScopedAllowBaseSyncPrimitives.
ScopedAllowBaseSyncPrimitivesForTesting is equivalent but can only
be used in tests and doesn't required adding a friend declaration to
thread_restrictions.h.
Previously, the code that is annotated with
ScopedAllowBaseSyncPrimitivesForTesting in this CL didn't fail because
it ran on a TaskRunner annotated with the deprecated
WithBaseSyncPrimitives() trait (cf.
https://cs.chromium.org/chromium/src/content/renderer/media/webrtc/task_queue_factory_unittest.cc?l=23&rcl=362f3723ac358d932ea2e3af65512a1243697a31).
Change-Id: Id7cfa2ea108870de86dc887458ae783c807791cc
Bug: chromium:889029
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128823
Commit-Queue: Francois Pierre Doray <fdoray@chromium.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27339}
When a bandwidth decrease to the estimated throughput would lead to
the "critical low" region we allow dropping to the link capacity
estimate instead (if it is higher).
Also moved BweInitialBackOffInterval config to the same field trial
string.
Bug: webrtc:10462
Change-Id: I4d6ee020a9ab8cede035b64253e3b3b1e2fb92b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129920
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27325}
Previously, legacy GetStats would look up the track ID by querying the
local/remote SDP by SSRC. This doesn't work with Unified Plan since the
RtpSender/RtpReceiver track IDs may not correspond to the track ID
stored in the SDP.
This CL changes legacy GetStats to pull the track ID directly from the
RtpSenders and RtpReceivers as it generates the stats. This has a few
additional benefits:
1) Unsignaled receive SSRC stats should now get correctly matched to
the unsigneled RtpReceiver track ID for both Plan B and Unified
Plan.
2) Removes a couple methods on PeerConnection that were only used by
the legacy StatsCollector.
3) Keeps the SSRC -> track ID mapping more localized which should make
the code easier to understand.
Bug: chromium:943493
Change-Id: I43ecde8c3a3d1c5f9c749ba6c8dfb11e8c4950fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129782
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27324}
This adds the bindings for rid in RtpParameters.Encoding and bindings
for send_encodings in RtpTransceiverInit to allow creating a transceiver
with multiple send encodings.
Bug: webrtc:10464
Change-Id: I4c205dc0f466768c63b7efcb3c68e93277236da0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128960
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27323}
Problem seems to be that once the estimate drops, "sample_uncertainty"
becomes very large, and it therefore takes a long time to recover.
Fix is under config for further downstream verification.
Bug: webrtc:10462
Change-Id: I5c2035f06e8a5088db0f0cb6ca511ef900e07645
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128902
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27320}
This change prevents FilterAnalyzer from accessing memory out-of-bounds
when the filter is resized.
Bug: chromium:946439
Change-Id: I7e2392c8b1ff0ff55566c663bf6d7a40d7754501
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129928
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27318}
This CL allows us to control how many probes we send when the bandwidth
allocation is updated, and how big they are.
Bug: webrtc:10394
Change-Id: I19e40740a528f83384b65d7509295034cc9a3031
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129904
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27317}
The pointer-to-submodule interfaces are being removed.
This CL:
1) introduces AudioProcessing::Config::GainController1 with most config,
2) adds functions to APM for setting and getting analog gain,
3) creates a temporary GainControlConfigProxy to support the transition
to the new config.
4) Moves the lock references in GainControlForExperimentalAgc and
GainControlImpl into the GainControlConfigProxy, as it becomes the
sole AGC object with functionality exposed to the client.
Bug: webrtc:9947, webrtc:9878
Change-Id: Ic31e15e9bb26d6497a92b77874e0b6cab21ff2b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126485
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27316}
This CL changes the API for webrtc::VideoEncoder.
There is a legacy method called SetRates(). This is indicated as being
deprecated, but there seem to be a number of usages still left.
Then there is the new SetRateAllocation() method which takes a
VideoBitrateAllocation instance instead of a single target bitrate.
This CL adds a new version of SetRates() which moves all the existing
parameters in a RateControlParameters struct, and adds a bandwidth
allocation signal. The intent of this signal is to allow the encoder
to know how close to the target it needs to stay. If the encoder rate
is network restricted, it will need to be more aggressive in keep the
rate down and possibly drop frames. If there is some margin for
overshoot, it might do different trade-offs.
Furthermore, the frame rate signal is changes from an integer to a
floating point type in order to get more precise rates at low frame
rates and the return type has been changed to void since the only use
of the previous values to log it and that is better done inside encoder
where the error condition originates.
The intent is to properly deprecate the "old" SetRates() /
SetRateAllocation() methods, send out a PSA and then remove them in two
weeks. Changes required by users should be trivial, as using the new
headroom signal is optional.
Bug: webrtc:10155, webrtc:10481
Change-Id: I4f797b0b0c73086111869ef4ee5f42bf530f5fde
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129724
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27314}
As a library, WebRTC should not assume UNICODE and _UNICODE to be
defined globally.
This CL explicitly selects wide character functions and types in
order to build WebRTC with /UUNICODE and /U_UNICODE.
Bug: None
Change-Id: Ie4e2bcb4c5c34aee6f68dc7b5b54b76f088ee3e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128904
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Noah Richards <noahric@chromium.org>
Cr-Commit-Position: refs/heads/master@{#27313}
This lets us change how many bytes and packets goes into the probes, as
well as some other things.
Bug: webrtc:10394
Change-Id: I26bb26a644e6f00366e9275228760c8744d63735
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128424
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27312}
Even if neither frame height nor frame width is <=0 we can end up
with <=0 dimensions in renderHeight or renderWidth. With this change,
we perform the check on the latter.
Bug: webrtc:10367
Change-Id: I9672672659ad7d12cf1e7ccab5b5cde583ae7e8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129760
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27307}
- Add GetFrameStatistics API:
This is useful for downstream test users that want to read frame-level stats.
- Remove other APIs that are not used by downstream tests:
* AddFrame
* GetFrame
* GetFrameWithTimestamp
* SliceAndCalcAggregatedVideoStatistic
* PrintFrameStatistics
* Size
* Clear
The implementations, which are used by the fixture implementation, are kept.
Bug: webrtc:10349
Change-Id: Id2f6fa5a36b8341a5ccb365725f71ebe0c0f1570
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128779
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27306}
