- Loss based bwe has 3 states: increasing (increasing when loss limited), decreasing (decreasing when loss limited), or delay based bwe (the same as delay based estimate).
- When bandwidth is loss limited and decreasing, and probe result is available, GetLossBasedResult = min(estimate, probe result).
- When bandwidth is loss limited and increasing, and the estimate is bounded by acked bitrate * a factor.
- When bandwidth is loss limited and probe result is available, use probe bitrate as the current estimate, and reset probe bitrate.
Bug: webrtc:12707
Change-Id: I53cb82aa16397941c0cfaf1035116f775bdce72b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277400
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38382}
This CL:
- makes it easier to understand the (nontrivial) metric interpretation
- corrects the computation of BufferDelay to use 0 for absent delay
- deletes metric MaxSkewShiftCount, unused since https://webrtc-review.googlesource.com/c/src/+/119701
- updates the unit test to directly test metric reporting
Corresponding update to histograms.xml:
https://crrev.com/c/3944909
Previous revert:
https://webrtc-review.googlesource.com/c/src/+/279040
This CL is identical to the original, except:
- the test is updated to spam fewer EXPECT_EQ failures on failure (EXPECT_EQs moved out of inner loop)
- the test not resets metrics (metrics::Reset()) at the beginning, like other histogram tests
Bug: webrtc:8671, chromium:1349051
Change-Id: Ie802e1f9d03a22ff7018f522a63b19e0b6eec2e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279046
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38376}
This is a speculative fix for the DCHECK at the top of
ScreenCapturerX11::CaptureScreen(). Whenever |selected_monitor_rect_|
changes, |queue_| should be reset, so that new frames are allocated
with the correct size. This CL adds a reset to UpdateMonitors() which
modifies |selected_monitor_rect_| and is called whenever an X11
configuration-change event is received (for example, when a monitor is
resized).
Bug: chromium:1372579
Change-Id: I9cc84a8b6990802f9d7dde05966ee17a80ddd48e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279065
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Lambros Lambrou <lambroslambrou@chromium.org>
Auto-Submit: Lambros Lambrou <lambroslambrou@chromium.org>
Cr-Commit-Position: refs/heads/main@{#38374}
- Set the initial input volume to that forced by startup min volume
since the latter is removed in a follow-up CL
- Remove unwanted expectations
Bug: webrtc:7494
Change-Id: I2df28f5bfaf4e592dfeae5e03b157268473cc822
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278784
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38370}
If the network thread and worker thread is the same, this log will spam.
Bug: webrtc:14502
Change-Id: Icb283f38fe6fbbca06ce911b9c0793148d459eef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278790
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38363}
This CL:
- makes it easier to understand the (nontrivial) metric interpretation
- corrects the computation of BufferDelay to use 0 for absent delay
- deletes metric MaxSkewShiftCount, unused since https://webrtc-review.googlesource.com/c/src/+/119701
- updates the unit test to directly test metric reporting
Corresponding update to histograms.xml:
https://crrev.com/c/3944909
Bug: webrtc:8671, chromium:1349051
Change-Id: If73b6fca4de7343bff2c53f72cedda458d36c599
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278782
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38362}
This makes the implementation in line with the existing X11
implementation:
https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/desktop_capture/linux/x11/screen_capturer_x11.cc;l=240-243
The issue I am observing on slightly slower machines with 4k monitor
is that the frames tend to go back in time. I believe this happens
when the shared frame queue is full and has its frame shared. When
that happens, we still end up calling MoveToNextFrame and doing so
we will wrap around the queue and if the capturer captures a frame
again, it sees an older frame. This is causing screen glitches.
This CL normalizes the implementation with X11 (which is known to
work fine) and moves to next frame and always uses it. This helps
to keep the current_frame_ in sync for the caller / capturer and
the capturer will then always see the video moving forward.
On the same machine, these screencasts were taken:
Without this fix: https://youtu.be/7Toi8dL5eYw
With this fix: https://youtu.be/LOE8Si5iOuQ
Bug: chromium:1291247
Change-Id: I51d3d700d3417d31371b12a94f445fc7b530cf73
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278700
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Salman Malik <salmanmalik@chromium.org>
Cr-Commit-Position: refs/heads/main@{#38342}
Instead of re-using the sender task queue, a new task queue will
suffice.
Bug: webrtc:14445
Change-Id: Ia7395ace2f0bb66bf9e76e3783b208f2cd0385dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275771
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38332}
This CL adds #includes to header files in order to make them
self contained after the preprocessor pass.
Bug: b/251890128
Change-Id: I81c3ba38fb8ab8a2bbd151ba99aa871fae9f1b1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278422
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38327}
Add passing optional speech level and speech probability to Process().
This enables computing an override for the RMS error from
Agc::GetRmsErrorDb(). Currently no speech level or probability are
passed outside the tests and no override happens elsewhere.
Bug: webrtc:7494
Change-Id: I0a7b1204aa51bcde8588963a5af023410405e83d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277560
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38318}
Add a buffer class to store speech probabilities and to estimate speech
activity. Follows the implementation of speech activity computation in
LoudnessHistogram but uses floats for computations.
Bug: webrtc:7494
Change-Id: I6ee72ec52919904ea4e1fbe51d61993aa7813c9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277801
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38309}
The pacer can thus run on the Worker thread or an owned TQ depending on field trial string "WebRTC-SendPacketsOnWorkerThread"
Bug: webrtc:14502
Change-Id: Ic74b92b21371cc62c7b2f62f039bc800dcceef8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277622
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38301}
The class will be used in experiment aiming at reducing the number of
used threads. The experiment will remove the need for the Pacer TQ and
RTP module worker TQ.
The helper ensure calls are made on either the worker thread a TQ
depending on the field trial
"WebRTC-SendPacketsOnWorkerThread"
Bug: webrtc:14502
Change-Id: I47581e3e3203712a244f1cb76952cd94734cc3f1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277444
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38289}
Reasons:
1) It is not used by `PeerConnection` (only in tests)
2) We have no plans on using it
3) The code is functionally untouched since many years
Bug: b/249972434
Change-Id: I1d30edd34231f25d86e8495ff71f1786ba2b0a1c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277445
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38260}
This is to keep the deprecated VideoReceiver separate from the
implementation used by VideoReceiver2 before updating
VCMDecoderDatabase to have ownership of the registered decoders.
Fixing typo (DataBase->Database) in the name of the remaining class.
Bug: webrtc:14486, webrtc:14497
Change-Id: I5ee755921454b0831b3af6d0161f5b48c7c60540
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276781
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38247}
This updates VideoReceiveStream2::Stop() to symmetrically tear down
state that's built up in VideoReceiveStream2::Start().
Bug: webrtc:11993, webrtc:14486
Change-Id: I41f4feea5584e5baaeed2143432136f8b9761321
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272537
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38244}
When loss rate is above a certain threshold, set instant_limit = 500 - 1000 * average_loss_rate, which returns 200kbps at 30% loss rate, or 100kbps at 40% loss rate. When the loss rate is above 50%, use the min_bitrate from send_side_bandwidth_estimation.
The high_loss_rate_threshold is set to 1.0, so the change is not activated by default.
Tested the change with hamrit, when average loss rate is above 50%, bandwidth backed to 10kbps, and it took ~10s to ramp up to 1.5Mbps.
https://screenshot.googleplex.com/7dvPoWa2b5SgMSL
Bug: webrtc:12707
Change-Id: I5eea04ef709a183bdf696246094dbd4a204e48f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272061
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38243}
This cl move VideoEncoderConfig from api/ to video/config.
VideoStreamEncoderInterface and VideoStreamEncoderObserver
are moved as collateral.
brandt@ think that the reason these were in api/ in the
first place had to downstream project.
Functionality wise, this is a NOP, but it makes it easier
to modify the encoder (config).
Bug: webrtc:14451
Change-Id: I2610d815aeb186298498e7102cac773ecac8cd36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277002
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38242}
MaybeInitializeCapture may overwrite the render configuration of a concurrent render reinitialization, leading to a second render reinitialization on the next render processing call.
See bug description for details.
Tested: Verified bitexactness offline (single-threaded) on a large number of aecdumps.
Bug: webrtc:14495
Change-Id: I9b70b454ce1c27859c3414c9c9ec89b7bbe35559
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277380
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38241}
This change adds support to allow ChromeOS capturers to also pass a
WindowId with a source. This WindowID can be used to help allow plumbing
and passing an Id that the capturing process knows about, in case it
wants to use any in-process capturing logic.
Bug: chromium:1273189
Change-Id: Ibcf494a75aec06eb1c44e6ff5fbdd9e2952e9b7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267086
Auto-Submit: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#38238}
This cl/ is a NOP refactoring,
moving the EncoderStreamFactory from within webrtc_video_engine.cc
into own file in video/. simulcast.cc is collateral.
Bug: webrtc:14451
Change-Id: Ia69b9241d8cd8a12be6628d887701f2e244c07cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276861
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38224}
This CL wraps the |Dav1dPicture| data directly for |VideoFrame| using
instead of copy data out to new buffer.
Bug: None
Change-Id: I21ceffb5cac7dda4a44eafbd0ed221974b8d45ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276526
Commit-Queue: Zhaoliang Ma <zhaoliang.ma@intel.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38194}
