See https://webrtc-review.googlesource.com/c/src/+/121764 for the
overall vision.
This CL adds a multistream Opus decoder. It's a new code-path to not
interfere with the standard Opus decoder. We introduce new SDP syntax,
which uses terminology of RFC 7845. We also set up the decoder side to
parse it. The encoder part will come in a later CL.
E.g. this is the new SDP syntax for 6.1 surround sound:
"multiopus/48000/6 channel_mapping=0,4,1,2,3,5 num_streams=4 coupled_streams=2"
Bug: webrtc:8649
Change-Id: Ifbc584cbb6d07aed373f223512a20d6d72cec5ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129768
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27493}
When a LossNotification RTCP message is received, the sequence numbers
it refers to must be converted to timestamps before passing the message
down to the encoder. This CL gives VideoSendStreamImpl access to that
information via VideoSendStreamImpl::rtp_video_sender_.
TBR=sprang@webrtc.org
Bug: webrtc:10501
Change-Id: If207f0b6d2fb344da35b525cc104e8ba5cc614ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131323
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27489}
Any new capture implementation for windows should use VideoSourceInterface,
like current capturers for android, mac and ios.
Bug: None
Change-Id: I0dffa9fde5ba069aab6273bfef826d17f03877d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131337
Commit-Queue: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27483}
A reasonable amount of incoming packets could generate feedback
for millions of packets.
Bug: chromium:949020
Change-Id: I7f3e6b75b683af5b2732c472cc92c6788540486b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131333
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27481}
This change makes it possible to disable AEC3's render delay
controller and delay estimator, and instead rely on an external
delay estimator. The delay is communicated via SetAudioBufferDelay.
When the feature is enabled, no echo removal will be performed
until the first delay is provided.
The delay is
Bug: b/130016532
Change-Id: I16643109d78d770ff1d2713cf247b0b9cce1bc1c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131327
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27467}
After https://webrtc-review.googlesource.com/c/src/+/131141 there are some minor
changes to the encoding performance, hence the updated values.
Bug: none
Change-Id: Ifa661eea15a0d52f4760f4aac9294074faab757f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131382
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27463}
Extracting the work that's thread dependent from the work that will
also be done when using task queue.
Bug: webrtc:10365
Change-Id: I648796fe016c966c731c9b7f85d2a871c1f2a349
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131241
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27454}
Switch to explicit channel mappings (RFC 7845) when creating
multi-stream Opus en/de-coders. The responsibility of setting up the
channel mappings will shift from WebRTC to the WebRTC user.
See https://webrtc-review.googlesource.com/c/src/+/121764 for the
current vision. See also the first child CL
https://webrtc-review.googlesource.com/c/src/+/129768
that sets up the Decoder to use this code.
Bug: webrtc:8649
Change-Id: I55959a293d54bb4c982eff68ec107c5ef8666c5c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129767
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27452}
The video encoder should be informed of incoming LossNotification
RTCP messages. Since it is unaware of RTP sequence numbers,
or anything else RTP-related, the sequence numbers mentioned
in the RTCP message must first be mapped to timestamps,
since those are meaningful to the encoder.
This CL introduces RtpSequenceNumberMap, which maps RTP sequence
numbers to timestamps, while providing:
1. Capping the number of entries.
2. Wrap-around handling.
RtpSequenceNumberMap also remembers which packets were first
and/or last in the frame.
Later CLs will wire this up.
Bug: webrtc:10501
Change-Id: Ie0662cdb5706a3bcf63aa2934816a9df88439357
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130497
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27448}
This CL extends, and partly corrects, the benchmarking
code in audioproc_f to provide statistics for the API
call durations in audioproc_f
Bug: chromium:939791
Change-Id: I4c26c4bb3782335f13dd3e21e6f861842539ea62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129260
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27443}
When setting display scale to 200%, the mouse was shared only
for the top left quarter.
Regressed since https://chromium-review.googlesource.com/641075.
Indeed frame->rect() takes into account scale_factor while the
frame is constructed with a size that does not take this scale
factor into account.
Also make sure to do a float disivison in DesktopFrame::scale_factor()
so that it returns 1.5 instead of 1 when dpi is 144 (i.e. 150%).
Bug: chromium:948362
Change-Id: Ic10f44946c9f1b53181244a44a5b45109c259f9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130371
Reviewed-by: Brave Yao <braveyao@webrtc.org>
Commit-Queue: Julien Isorce <julien.isorce@chromium.org>
Cr-Commit-Position: refs/heads/master@{#27424}
This prepares for running WebRTC in simulated time where event::Wait
based timing doesn't work.
Bug: webrtc:10365
Change-Id: Ia0f9b1cc8e3c8c27a38e45b40487050a4699d8cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129962
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27422}
Fixed todo by replacing SequenceNumberUnwrapper with updated class
SeqNumUnwrapper that correctly handles reordering of early packets.
Bug: webrtc:10263
Change-Id: Iffd93db924fee132d35752996b8d29acbb315d24
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130498
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27417}
This change removes the (unused) ability of EchoRemover overriding
the delay of the RenderDelayController. The change is tested for
bit-exactness.
Bug: webrtc:8671
Change-Id: I188ef740f1437de64ffe236d07a7dcd4128192c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130518
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27414}
The existing windows dshow capture code was ignoring the requested
capture format when enumerating preferred formats. This CL changes
the behavior to move the requested format to the start of the
preferred list. This should fix an issue causing MJPEG to be not
selected on certain devices because other formats were preferred,
resulting in reduced framerates for HD capture (USB 2.0 throughput
isn't high enough to support capturing 30fps 720p I420 data, and many
cameras, even internal, are on the USB 2.0 bus).
Note that Chromium has yet a different implementation, which is if
MJPEG is selected, it claims to *only* support MJPEG and no other
formats. This solution seems more correct; if, for some reason,
MJPEG was listed as supported but isn't asked of the pin, this
code will still produce some output, just possibly at a lower framerate
or resolution.
Bug: None
Change-Id: Id86f345936f6f32e08beec02925e41056c87a544
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130843
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27412}
This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
to mark WebRTC symbols as visible from a shared library, this doesn't
mean these symbols are part of the public API (please continue to refer
to [1] for info about what is considered public WebRTC API).
[1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md
Bug: webrtc:9419
Change-Id: Ib2c29054b2ae008f5291bd3b762a504b18534326
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130513
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27410}
This fixes a regression introduces way back in August 2018:
https://webrtc-review.googlesource.com/c/src/+/91863/
For bonus points, also fixing an auxiliary test issue.
Bug: webrtc:10479, webrtc:10260
Change-Id: I4e99fe6e070446d10357d9d1a9d1ffc9dedcf419
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129926
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27409}
All application with empty title were not listed, for example Photos.
Fallback to owner name in that case while making sure to keep ignoring
the ghost window.
Most of the ghost windows can be filtered with IsWindowOnScreen
or IsWindowFullScreen except a few. For the remaining ghost we check
if there is no other window with the same pid.
Bug: chromium:516230
Test: Hangouts or Rumpus
Change-Id: Ibb9f98887e5aedf822fc0611836b1938b5056d43
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130360
Commit-Queue: Julien Isorce <julien.isorce@chromium.org>
Reviewed-by: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27401}
The has some benefits:
* We no longer need locks to protect the emulated network node state.
* We only process when there are packets in flight.
* It makes Scenario more similar to network emulation manager.
Bug: webrtc:10365
Change-Id: I8bd1ad1edfb54b047e8109dabd9846ae451cef17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127548
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27393}
And delete corresponding dependencies on :webrtc_common. After this
change, common_types.h is included directly only from code in the
following directories:
api/
api/video/
api/video_codecs/
common_video/libyuv/include/
media/base/
modules/remote_bitrate_estimator/
modules/rtp_rtcp/source/
modules/video_coding/codecs/vp9/
There remains plenty of indirect dependencies on the types declared in
common_types.h, but the fewer direct dependencies should make it
easier to find the proper place for each type.
Bug: webrtc:5876
Change-Id: I93e8f214025ecb613c19fdec2015bd3f96c59aae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27376}
In this CL:
- Created static helper function GenericFrameInfo::DecodeTargetInfo to
convert DTI symbols to a list of GenericFrameInfo::OperatingPointIndication.
- Added per frame DTI information for the different stream structures.
Bug: webrtc:10342
Change-Id: I62ff2e9fc9b380fe1d0447ff071e86b6b35ab249
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129923
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27350}
