This reverts commit 126648763184b7e224d6c4a2f85efb4a9307378f.
Reason for revert: Partial Capture API is not needed, according to new info from the Chrome team.
Original change's description:
> Partial frame capture API part 3
>
> Implement utility for applying partial updates to video frames.
>
> Bug: webrtc:10152
> Change-Id: I295fa9f792b96bbf1140a13f1f04e4f9deaccd5c
> Reviewed-on: https://webrtc-review.googlesource.com/c/120408
> 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@{#26522}
TBR=ilnik@webrtc.org,nisse@webrtc.org,sprang@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10152
Change-Id: I9d7c79ca571a44a419102871d3106e7065638433
Reviewed-on: https://webrtc-review.googlesource.com/c/122089
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26630}
This reverts commit 1f0a84a2ecea59f86adc1af70eed974a3c6d59ac.
Reason for revert: Partial Capture API is not needed, according to new info from the Chrome team.
Original change's description:
> Partial frame capture API part 5
>
> Wire up partial video frames in video quality tests
>
> Bug: webrtc:10152
> Change-Id: Ifa13bb308258c8d3930a6cfbcc97c95b132cecf3
> Reviewed-on: https://webrtc-review.googlesource.com/c/120410
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26549}
TBR=ilnik@webrtc.org,sprang@webrtc.org,stefan@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10152
Change-Id: I32017b1a7109a3615598a976f4b0e61edf4e8757
Reviewed-on: https://webrtc-review.googlesource.com/c/122088
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26628}
Logging will not stop after 10 minutes but after the max size of the
file is reached.
Bug: None
No-Try: True
Change-Id: I18c064cc60b52d892d5070b3eddf08749b60604c
Reviewed-on: https://webrtc-review.googlesource.com/c/121959
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26625}
Hard-coding default values forces IDs over 14 to be used even
when we offer less than 15 different extensions.
Note that the code relies on MergeRtpHdrExts for making sure
that extension IDs are kept consistent and non-colliding between
different streams (audio/video).
Bug: webrtc:10288
Change-Id: I3e59f7ddc8ca43cea91084a6b7f36df70fb6be4a
Reviewed-on: https://webrtc-review.googlesource.com/c/121646
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26622}
The new parameter simplify TaskQueueFactory lifetime and
allows insertion of per TestCase state
Bug: webrtc:10191
Change-Id: If4948df2756580957052b2b333b5c12cf4914d55
Reviewed-on: https://webrtc-review.googlesource.com/c/121648
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26583}
Adds methods AddNetworkChangeCallback and RemoveNetworkChangeCallback,
to replace SetNetworkChangeCallback. Needed because both VideoChannel and
VoiceChannel register such a callback.
Bug: webrtc:9719
Change-Id: Ic592b2d775d721a0f44ba0af88ed963bf02d73a3
Reviewed-on: https://webrtc-review.googlesource.com/c/121460
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26575}
This CL is a manual spin-off of [1], which tried to apply clang-tidy's
performance-move-const-arg [1] to the WebRTC codebase.
Since there were some wrong fixes to correct, this CL lands a few
different fixes, like adding a constructor overload to take an rvalue
reference or remove 'const' to make std::move effective.
[1] - https://webrtc-review.googlesource.com/c/src/+/120350
[2] - https://clang.llvm.org/extra/clang-tidy/checks/performance-move-const-arg.html
Bug: webrtc:10252
Change-Id: I42a777247fee2cb788efcd7c2035148330056b7a
Reviewed-on: https://webrtc-review.googlesource.com/c/120928
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26553}
Wire up partial video frames in video quality tests
Bug: webrtc:10152
Change-Id: Ifa13bb308258c8d3930a6cfbcc97c95b132cecf3
Reviewed-on: https://webrtc-review.googlesource.com/c/120410
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26549}
Adds methods AddNetworkChangeCallback and RemoveNetworkChangeCallback,
to replace SetNetworkChangeCallback. Needed because both VideoChannel
and VoiceChannel register such a callback.
This cl is step 1, it just adds the methods to the interface, without
calling them.
Bug: webrtc:9719
Change-Id: I39f1748706d4369ca71d594ca5e2f1380de5ce66
Reviewed-on: https://webrtc-review.googlesource.com/c/121462
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26541}
This allows us to bind to temporaries.
Bug: none
Change-Id: Ic84ad378f344776bef38f9dc81a6fe0dee74400f
Reviewed-on: https://webrtc-review.googlesource.com/c/120901
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26530}
This adds the following non-standardized metrics to video receiver
stats:
- freezeCount
- pauseCount
- totalFreezesDuration
- totalPausesDuration
- totalFramesDuration
- sumOfSquaredFrameDurations
For description of these metrics see
https://henbos.github.io/webrtc-provisional-stats/#RTCVideoReceiverStats-dict*
Bug: webrtc:10145
Change-Id: I4c76d5651102e73b1592ffed561e6224f2badeb6
Reviewed-on: https://webrtc-review.googlesource.com/c/114840
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26523}
Reland with fixes to undefined behavior.
Define new optional struct in VideoFrame to signal that the frame is a
changed part of a whole picture and add a flag to signal that partial
update may be issued by the VideoFrame source.
Also, fix too strict assumptions in FrameBuffers PasteFrom methods.
Also, add ability to set a new buffer in video frame.
Original Reviewed-on: https://webrtc-review.googlesource.com/c/120405
Bug: webrtc:10152
Change-Id: I85790dfc7cec2f23abfe9d6cd18dc76a0c343bc0
Reviewed-on: https://webrtc-review.googlesource.com/c/120780
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26493}
This reverts commit 8a21e1c9c95c9b9b570c84bdfeda0315ede9dc29.
Reason for revert: breaks buildbots
Original change's description:
> Partial frame capture API part 1
>
> Define new optional struct in VideoFrame to signal that the frame is a
> changed part of a whole picture and add a flag to signal that partial
> update may be issued by the VideoFrame source.
>
> Also, fix too strict assumptions in FrameBuffers PasteFrom methods.
> Also, add ability to set a new buffer in video frame.
>
>
> Bug: webrtc:10152
> Change-Id: Ie0da418fd60bc7a34334329292e0b860ec388788
> Reviewed-on: https://webrtc-review.googlesource.com/c/120405
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26489}
TBR=ilnik@webrtc.org,nisse@webrtc.org,sprang@webrtc.org
Change-Id: Ibf61f28e529a444882962b984474d4849bb44e4b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10152
Reviewed-on: https://webrtc-review.googlesource.com/c/120760
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26490}
Define new optional struct in VideoFrame to signal that the frame is a
changed part of a whole picture and add a flag to signal that partial
update may be issued by the VideoFrame source.
Also, fix too strict assumptions in FrameBuffers PasteFrom methods.
Also, add ability to set a new buffer in video frame.
Bug: webrtc:10152
Change-Id: Ie0da418fd60bc7a34334329292e0b860ec388788
Reviewed-on: https://webrtc-review.googlesource.com/c/120405
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26489}
This is useful for internal calculations in bitrate control code as we
can skip conversion constants.
DataRate Example(TimeDelta time, DataSize size) {
double time_seconds = time.seconds<double>();
double size_bytes = size.bytes<double>();
double rate_bytes_per_sec = size_bytes/time_seconds;
return DataRate::bytes_per_sec(std::max(0.0,rate_bytes_per_sec));
}
Bug: webrtc:9709
Change-Id: I8eefed578b6e8eee67fc36af723216407e0d0323
Reviewed-on: https://webrtc-review.googlesource.com/c/120720
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26488}
This creates the API for an ICE transport object, and lets it
be accessible from a DTLS transport object.
Bug: chromium:907849
Change-Id: Ieb24570217dec75ce0deca8420739c1f116fbba4
Reviewed-on: https://webrtc-review.googlesource.com/c/118703
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26472}
This CL is spawned from [1] and it introduces RTCError(const T&) in
order to remove an unneeded std::move.
[1] - https://webrtc-review.googlesource.com/c/src/+/120350
Bug: webrtc:10252
Change-Id: Ibd5aa1c901fd920549e9437908178c786019a328
Reviewed-on: https://webrtc-review.googlesource.com/c/120560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26468}
This is a followup, deleting the things marked as deprecated in
https://webrtc-review.googlesource.com/c/113180.
Bug: webrtc:9719
Change-Id: I64dc31c6918f575599fc6b0bbfa47c5b1f2f3019
Reviewed-on: https://webrtc-review.googlesource.com/c/113521
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26465}
This simplifies the design by making simulated network more self
sufficient. It also prepares for removing network node specific
configuration (The behavior implementation should be responsible
for handling any configuration.)
Bug: webrtc:9510
Change-Id: I218d70c0359774d9891178fbd8b1bbc729cbad92
Reviewed-on: https://webrtc-review.googlesource.com/c/120346
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26450}
Support varies by codec, especially in the simulcast case, but using
the EncoderSimulcastProxy codec should fix this.
Bug: webrtc:10069
Change-Id: Idb6a5f400ffda1cdb139004f540961a9cf85d224
Reviewed-on: https://webrtc-review.googlesource.com/c/119400
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26449}
FrameConfig is not specific to temporal layers. Anything that
can control referenced/updated buffers could potentially use it.
Bug: webrtc:10259
Change-Id: I04ed177ee884693798c3b69e35fd4255ce1e9062
Reviewed-on: https://webrtc-review.googlesource.com/c/120355
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26448}
Prior to this CL, RtpPayloadParams had code that assumed
dependency patterns in VP8, in order to write that information
into the [Generic Frame Descriptor] RTP extension.
This CL starts moving that code out of RtpPayloadParams.
Upcoming CLs will migrate additional encoder-wrappers to
the new scheme, then remove the deprecated code.
Bug: webrtc:10249
Change-Id: I5fc84aedf8e11f79d52b989ff8b7ce9568b6cf32
Reviewed-on: https://webrtc-review.googlesource.com/c/119958
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26438}
Bug: webrtc:9719
Change-Id: I90bd1d9858c259d7339420c574ad83d6fb18299c
Reviewed-on: https://webrtc-review.googlesource.com/c/118946
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26426}
Currently it's possible that no-op DTLS is created if media transport is only used for data channels.
Changing it so that no-op DTLS is only created when both media & data will flow through media transport.
Bug: webrtc:9719
Change-Id: I87f27fc778ea21b12f2904bad1452d893f66b541
Reviewed-on: https://webrtc-review.googlesource.com/c/119909
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26416}
The type rtc::scoped_refptr<T> is now part of api/. Please include it from
api/scoped_refptr.h.
More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o.
Bug: webrtc:9887, webrtc:8205
No-Try: True
Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf
Reviewed-on: https://webrtc-review.googlesource.com/c/119041
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26414}
This CL removes the legacy, no longer used, render buffering code. It
also removes four unused parameters from the AEC3 config. The change
is tested for bit-exactness.
Bug: webrtc:8671
Change-Id: I2bb6cb7a1097863f228767d757d551c00593bb00
Reviewed-on: https://webrtc-review.googlesource.com/c/119701
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26399}
Some codecs don't support directly creating simulcast layers with
non-optimal parameters. This proxy will detect this and create
multiple encoders then, one for each layer as a fallback.
Bug: webrtc:10069
Change-Id: I4bcafcfdd68d9ed466e2fafe564db849de6ed4f6
Reviewed-on: https://webrtc-review.googlesource.com/c/119264
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26373}
