Current way with updates on each frame caused a bogus jitter estimate
and lots of dropped frames in unfiltered KSVC stream.
Bug: chromium:912122
Change-Id: I4a1af71a242af3f9b5f5a411b194331b2df24f68
Reviewed-on: https://webrtc-review.googlesource.com/c/117566
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26322}
If we do, the bitrate allocator will assume there can be a lot a FEC
and other things and bumps the max probing bitrate by 2x.
This caused a bunch of perf tests to change in a non-obvious way.
This is a follow-up to
https://webrtc-review.googlesource.com/c/src/+/115410
Bug: webrtc:10155, chromium:922396
Change-Id: I51d3611cb21d98a8fab1bfab2d8f167ed859696d
Reviewed-on: https://webrtc-review.googlesource.com/c/118043
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26319}
PlatfromThread is used by several TaskQueue implementations that plan
to take it's name as string_view and thus wouldn't guarantee it is 0-terminated.
Bug: webrtc:10191
Change-Id: I7dadb917192257f0185f9cb4f0ec207d7156d3c3
Reviewed-on: https://webrtc-review.googlesource.com/c/118140
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26317}
This flag only needs to be set in kOn interlayer prediction mode, because
in all others, if new layer is enabled - a keyframe is generated.
Also, use external reference control in that case, because libvpx creates
rtp-incompatible references in that case.
Bug: webrtc:10180
Change-Id: I0fad188fa8cd424f831bac219769dbad3a788b1d
Reviewed-on: https://webrtc-review.googlesource.com/c/118041
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26316}
This CL adds a single class to manage the use case of having a task
that repeats itself by a fixed or variable interval. It replaces the
repeating task previously locally defined for rtp transport controller
send as well as the cancelable periodic task. Furthermore, it is
introduced where one off repeating tasks were created before.
It provides the currently used functionality of the cancelable periodic
task, but not some of the unused features, such as allowing cancellation
of tasks before they are started and cancellation of a task after the
owning task queue has been destroyed.
Bug: webrtc:9883
Change-Id: Ifa7edee836c2a64fce16a7d0f682eb09c879eaca
Reviewed-on: https://webrtc-review.googlesource.com/c/116182
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26313}
The WebRtcKeyValueConfig interface allows providing custom key value
configurations that changes per instance of GoogCcNetworkController.
Bug: webrtc:10009
Change-Id: I520fff030d1c3c755455ec8f67896fe8a6b4d970
Reviewed-on: https://webrtc-review.googlesource.com/c/116989
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26312}
When Flex and RTX are both specified, Flex will not be used because
RTX will introduce a new SSRC making the total SSRC count > 1.
Flex can only protect a single stream, so if the total SSRC count
is > 1, it is not used.
The fix is simple, to check the number of "primary SSRCs" before
redundancy streams are added when determining if Flex should be used.
Bug: None
Change-Id: I98df1b807d306bdcce1a76dfb163aa14e60d0052
Reviewed-on: https://webrtc-review.googlesource.com/c/118220
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26308}
This reverts commit 83ed89a45f4578ca07efef48e772b9aafb263163.
Reason for revert: breaks downstream project
Original change's description:
> Opus multistream.
>
> This is a backwards-compatible change. It makes WebRTC use the Opus
> multistream decoder for all Opus packets. Single-stream packets are a
> special case of multistream ones (with stream=1).
>
> The tricky parts are 'WebRtcOpus_GetMaxPlaybackRate' and
> 'WebRtcOpus_GetSurroundParameters'. GetMaxPlaybackRate is supposed to
> do what opus_encoder_ctl(encoder, OPUS_GET_MAX_BANDWIDTH(&bandwidth))
> did when we had single-stream encoders. Now there may be several
> independent encoders with possibly different BANDWIDTH. The new
> GetMaxPlaybackRate queries all of them, and returns a playback rate if
> all the encoder's rates are equal.
>
> WebRtcOpus_GetSurroundParameters is a configuration convention. It
> maps the number of channels to a multi-stream encoder/decoder
> configuration. As described in RFC 7845
> https://tools.ietf.org/html/rfc7845#section-5.1.1, a multi-stream
> encoder/decoder needs a number of streams, number of coupled streams
> and a 255-byte mapping array. The function GetSurroundParameters
> computes all of these from the number of channels. [1, 2, 4, 6, 8]
> channels are supported.
>
> Bug: webrtc:8649
> Change-Id: I271de8e387d738254d6aa53af7fcf8644a53edb5
> Reviewed-on: https://webrtc-review.googlesource.com/c/111750
> Commit-Queue: Alex Loiko <aleloi@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26293}
TBR=aleloi@webrtc.org,minyue@webrtc.org
Change-Id: I1002e3273b44d3cccacdba84b8c363eefd537c4b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8649
Reviewed-on: https://webrtc-review.googlesource.com/c/118201
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26306}
VideoFrameWriter is designed to accept webrtc::VideoFrame as input and
write it with Y4mFrameWriterImpl to the output file, transforming
webrtc::VideoFrame to the uint8_t* frame_buffer. VideoFrameWriter will
be used to write webrtc::VideoFrames during dumping input and output
video in peer connection level test framework and will be injected
in webrtc::test::FrameGenerator and rtc::VideoSinkInterface<VideoFrame>.
Bug: webrtc:10138
Change-Id: Iadec7d3ad66f226836acbebe070cf88ceb242f62
Reviewed-on: https://webrtc-review.googlesource.com/c/117200
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26305}
This reverts commit 74ba99062c48b278675cfe52643719202296fddc.
Reason for revert: Breaks downstream project.
Original change's description:
> AEC3: Lockless transfer of render data to the capture thread
>
> This CL implements a lockless queue that replaces SwapQueue
> in the RenderWriter. This avoid stalls when the render and
> capture threads are accessing the queue at the same time.
>
> Bug: webrtc:10205
> Change-Id: Ie7d6fcf9c80fad957e2a90537658fb730ca2ed72
> Reviewed-on: https://webrtc-review.googlesource.com/c/117643
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26298}
TBR=gustaf@webrtc.org,peah@webrtc.org
Change-Id: Ie76ee8835da4e44982d181a152c9ffa19ff33e23
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10205
Reviewed-on: https://webrtc-review.googlesource.com/c/118142
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26304}
This analyzer will be used in implementatino of peer connection level
test framework before main analyzer will be implemented.
Bug: webrtc:10138
Change-Id: Ibb7c5cd94b0f07c6fc5a2415f04b0f0ae7ae75e2
Reviewed-on: https://webrtc-review.googlesource.com/c/117221
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26301}
This CL add new data to the VideoEncoder::EncoderInfo struct, indicating
how the encoder intends to allocate frames across spatial and temporal
layers.
This metadata will be used in upcoming CLs to control how the encoder's
rate controller performs.
Bug: webrtc:10155
Change-Id: Id56fae04bae5f230d1a985171097d7ca83a3be8a
Reviewed-on: https://webrtc-review.googlesource.com/c/117900
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26300}
kMaxSimulcastStreams, kMaxSpatialLayers and kMaxTemporalStreams don't
really beling on VideoBitrateAllocation.
common_types.h is going away and it feels dubious to requrie include
of the full VideoEncoder api to use them. Therefore moving them into a
seprate file/target.
Also includes some remaining cleanup of includes.
Bug: webrtc:9271
Change-Id: I7ded3d97a9a835ac756159700774445a2b93a697
Reviewed-on: https://webrtc-review.googlesource.com/c/117305
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26299}
This CL implements a lockless queue that replaces SwapQueue
in the RenderWriter. This avoid stalls when the render and
capture threads are accessing the queue at the same time.
Bug: webrtc:10205
Change-Id: Ie7d6fcf9c80fad957e2a90537658fb730ca2ed72
Reviewed-on: https://webrtc-review.googlesource.com/c/117643
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26298}
It only tests the videoadaptation in the test class FakeVideoCapturer.
Bug: webrtc:6353
Change-Id: I4766eebc5cfa7412fde9fd6e5173f1b2381a5d87
Reviewed-on: https://webrtc-review.googlesource.com/c/118042
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26297}
This is a backwards-compatible change. It makes WebRTC use the Opus
multistream decoder for all Opus packets. Single-stream packets are a
special case of multistream ones (with stream=1).
The tricky parts are 'WebRtcOpus_GetMaxPlaybackRate' and
'WebRtcOpus_GetSurroundParameters'. GetMaxPlaybackRate is supposed to
do what opus_encoder_ctl(encoder, OPUS_GET_MAX_BANDWIDTH(&bandwidth))
did when we had single-stream encoders. Now there may be several
independent encoders with possibly different BANDWIDTH. The new
GetMaxPlaybackRate queries all of them, and returns a playback rate if
all the encoder's rates are equal.
WebRtcOpus_GetSurroundParameters is a configuration convention. It
maps the number of channels to a multi-stream encoder/decoder
configuration. As described in RFC 7845
https://tools.ietf.org/html/rfc7845#section-5.1.1, a multi-stream
encoder/decoder needs a number of streams, number of coupled streams
and a 255-byte mapping array. The function GetSurroundParameters
computes all of these from the number of channels. [1, 2, 4, 6, 8]
channels are supported.
Bug: webrtc:8649
Change-Id: I271de8e387d738254d6aa53af7fcf8644a53edb5
Reviewed-on: https://webrtc-review.googlesource.com/c/111750
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26293}
This is in preparation for letting Chrome extract DTLSTransport
information after SLD/SRD instead of doing it on-demand.
Bug: chromium:907849
Change-Id: Iac6b174c98d3d14136e1fd25bce4a9292f6c8b41
Reviewed-on: https://webrtc-review.googlesource.com/c/116984
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26289}
We'd like to disable RTP code path when media transport is used. In particular, we don't want occasional RTP/RTCP packets sent from the RTP code path when media transport is used.
Long term we will remove this new NoOp DTLS transport, when we stop creating rtp transport.
Bug: webrtc:9719
Change-Id: I27f121edef394465ddc8fe8003e6f4428b10c022
Reviewed-on: https://webrtc-review.googlesource.com/c/117700
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26286}
In this reland, I disabled high bitrate webrtc perf test on Android32.
This is a reland of 15df2774f4e85cf8900768c1793edcf17d651dcd
Original change's description:
> This CL adds a fake codec factory in WebRTC that can be used in tests to
> produce target bitrate output.
> We also add a high bitrate test that makes use of fake codec. This test assumes
> ideal network conditions with target bandwidth being available and exercises
> WebRTC calls with a high target bitrate(100 Mbps) end-to-end.
TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,emircan@webrtc.org,kron@webrtc.org
Bug: chromium:879723
Change-Id: I31a4b48d986bef9ca003ae71afeb567ae3e562c9
Reviewed-on: https://webrtc-review.googlesource.com/c/117980
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26285}
This makes it possible to save log outputs from scenario tests to
either files or memory.
Bug: webrtc:9510
Change-Id: I883bd8240ab712d31d54118adf979041bd83481a
Reviewed-on: https://webrtc-review.googlesource.com/c/116321
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26284}
So far, base channel was only notifying about 'first audio packet' when
RTP was used, and it never notified about it when media_transport
interface was used. This change adds a sigslot to notify about a new
media packet to the media transport interface.
Bug: webrtc:9719
Change-Id: Ie9230c407f35b1aaa71ba71008ac34ba8869e2d4
Reviewed-on: https://webrtc-review.googlesource.com/c/117249
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26282}
This adds calculation and reporting of harmonic frame rate (HFR) metric.
HFR is calculated as call_duration_secs / sum(frame_duration_secs ^ 2).
It penalizes long freezes and could better represent user experience
related to smoothness of playback.
Bug: none
Change-Id: I4d2d46deaa44bb4221b53969a1c0a334e0c1bde9
Reviewed-on: https://webrtc-review.googlesource.com/c/117661
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26279}
Test wiring to DegradationPreference passed to
VideoSendStream::SetSource, but not the adaptation implemented in the
test class FakeVideoCapturer.
Bug: webrtc:6353
Change-Id: Iec2ae89283fb856822ea2829db17eaa02337b467
Reviewed-on: https://webrtc-review.googlesource.com/c/117641
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26278}
NetEq currently only passes `jitterBufferDelay` to `getStats()`. We need its paired `jitterBufferEmittedCount` denominator stat for the calculations to be accurate.
Bug: webrtc:10192
Change-Id: I655aea629026ce9101409c2e0f18c2fa57a1c3ab
Reviewed-on: https://webrtc-review.googlesource.com/c/117320
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#26276}
Use size() accessor function. Also replace most nearby uses of _buffer
with data().
Bug: webrtc:9378
Change-Id: I1ac3459612f7c6151bd057d05448da1c4e1c6e3d
Reviewed-on: https://webrtc-review.googlesource.com/c/116783
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26273}
