Since the ice_transport element of DtlsTransport is never changed
over the lifetime of the object, it's safe to return it on any thread.
Added const qualifier to make this obvious.
Bug: chromium:907849
Change-Id: Ib180682689a57df5106717b125c626d7ac9e7561
Reviewed-on: https://webrtc-review.googlesource.com/c/123781
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26800}
Replaces use of field trials in RtpSender and RtpVideoSender with injectable WebRtcKeyValueConfig.
Implementation still defaults to field trials.
BUG: webrtc:10335
Change-Id: I5fc6d327ee5d011ccc41385734b38344df172627
Reviewed-on: https://webrtc-review.googlesource.com/c/123447
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26795}
This change disables the ERLE estimation of onsets and instead assumes
minimum ERLE. This reduces the risk of echo leaks during onsets. The
estimated ERLE was sometimes incorrect due to:
- Not enough data to train on.
- Platform noise suppression can change the echo-path.
Bug: chromium:119942,webrtc:10341
Change-Id: I1dd1c0f160489e76eb784f07e99af02f44f387ec
Reviewed-on: https://webrtc-review.googlesource.com/c/123782
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26794}
It was previously possible to escape the sandbox by calling
rtc::Thread::SetAllowBlockingCalls(true).
This CL only removes the loophole on non-Android builds, because we
still have old Android code that relies on it. We expect that code to
go away soon-ish, though.
Bug: webrtc:9987
Change-Id: Ida96400d0abe430af4c2046284795d37d64f6613
Reviewed-on: https://webrtc-review.googlesource.com/c/123523
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26792}
This cl change RtpSender to feed the PacedSender with RTP packet size rather than payload size in experiment WebRTC-SendSideBwe-WithOverhead. Before this cl, the congestion controller was feed with packet size but not the pacer. That means that the pacer budget was updated with an estimate that includes the RTP headers, but the media budget only use the payloads.
BUG: webrtc:10325 webrtc:6762
Change-Id: I35c8350603a7881ea162debcd89ed901cbb50950
Reviewed-on: https://webrtc-review.googlesource.com/c/123444
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26788}
To support analyze of spatial layers we will continue sending them
into the network on encoder side, but will mark which should be then
discarded and which should be processed. On decoder side we will drop
layers, if they should be discarded and decode only parts, that
should be processed.
Bug: webrtc:10138
Change-Id: Ic8b8fe7787674c0ec49b879fcc29e54e8e3d787f
Reviewed-on: https://webrtc-review.googlesource.com/c/123185
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26784}
This reverts commit 5341aaccdb64e3336abf5875e8828222446adffa.
Reason for revert: Order of initialization of global static strings.
Original change's description:
> Reland of https://webrtc-review.googlesource.com/c/src/+/114883
>
> The difference to the original is new bitexactness strings AND
> global static file string constants. The reason for reland is breaking
> downstream projects.
>
> Original CL description:
>
> Tests for multi-stream Opus.
>
> This CL (mainly) adds bit-exactness tests for multi-stream Opus. The
> tests are in audio_coding_unittest.cc. Some refactoring of
> AcmSendTestOldApi, AcmSenderBitExactnessOldApi is done to make it
> possible. A few checks for "channels \in {1, 2}" are replaced with
> "channels \in {1, 2, 4, 6, 8}" in the WebRTC Opus codec wrapper. A few
> other changes are made to be able to write and read multi-channel WAV
> files.
>
> The SDP changes are NOT included; as of this CL there is no way to set
> up a multi-stream opus en/de-coder from SDP strings.
>
> Bug: webrtc:8649
> Change-Id: I9fd47c790c241c1876c4a731b0840bec30b4f1b2
> Reviewed-on: https://webrtc-review.googlesource.com/c/123387
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Alex Loiko <aleloi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26774}
TBR=aleloi@webrtc.org,ossu@webrtc.org
Change-Id: I88060f2050ccee83d6091b042a10f79b3c4edc47
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8649
Reviewed-on: https://webrtc-review.googlesource.com/c/123580
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26777}
It is possible to get event logs from both RTP and MediaTransport when
media transport is only used for data channel. This is undesirable. We
would rather not get any logs from media transport when it's not used
for media.
This change disables rtc event log when media transport is not used for
media.
Bug: webrtc:9719
Change-Id: Ibc660b37c5d98001144e5f68b32f0608fd6ede33
Reviewed-on: https://webrtc-review.googlesource.com/c/123260
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26776}
Added audio format field and set method to Builder. - WebRTCAudioRecord. Added audio format field, added to constructor. Default audio format value AudioFormat.ENCODING_PCM_16BIT. initRecord(), added how to calculate bytesPerFrame, depends on audioFormat.
First commit and contribution, updated AUTHORS file
Bug: None
Change-Id: I16f660d42350ec9ce2e329b239bd7f6324e76dfe
Reviewed-on: https://webrtc-review.googlesource.com/c/122302
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26775}
The difference to the original is new bitexactness strings AND
global static file string constants. The reason for reland is breaking
downstream projects.
Original CL description:
Tests for multi-stream Opus.
This CL (mainly) adds bit-exactness tests for multi-stream Opus. The
tests are in audio_coding_unittest.cc. Some refactoring of
AcmSendTestOldApi, AcmSenderBitExactnessOldApi is done to make it
possible. A few checks for "channels \in {1, 2}" are replaced with
"channels \in {1, 2, 4, 6, 8}" in the WebRTC Opus codec wrapper. A few
other changes are made to be able to write and read multi-channel WAV
files.
The SDP changes are NOT included; as of this CL there is no way to set
up a multi-stream opus en/de-coder from SDP strings.
Bug: webrtc:8649
Change-Id: I9fd47c790c241c1876c4a731b0840bec30b4f1b2
Reviewed-on: https://webrtc-review.googlesource.com/c/123387
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26774}
The timestamps doesn't always match properly, currently causing
flakiness and crashes. Pending a better solution we'll assume that
no frames are lost.
Bug: webrtc:9510
Change-Id: I1b0a5025ac9a45c71b611bcddbbad7a8cf385e01
Reviewed-on: https://webrtc-review.googlesource.com/c/123483
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26772}
The discardability flag denotes whether the frame may be dropped by
the decoder with no effect on the decodability of subsequent frames.
Bug: webrtc:10214
Change-Id: I3654951d8863b50effe9670b8d1d7eb051240039
Reviewed-on: https://webrtc-review.googlesource.com/c/122241
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26763}
The FFT output buffers sizes in SpectralFeaturesExtractor have been reduced
from N to N/2+1, where N is the audio frame size. This is required since
ComputeBandEnergies() currently calls ComputeBandCoefficients() indicating
a higher value for max_freq_bin_index, hence polluting the higher bands with
unwanted energy (coming from the symmetric conjugate copy of the Fourier
coefficients).
Bug: webrtc:10332
Change-Id: Ie080050c4f357fa95e256cf2a6bf572222e8ca44
Reviewed-on: https://webrtc-review.googlesource.com/c/123239
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Pablo Barrera González <barrerap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26761}
Intended to make copy construction and assignment of EncodedImage
cheaper, but otherwise not have any effect on users of the class.
Bug: webrtc:9378, chromium:931692
Change-Id: I22cf8c05f6ef7b7b5cf7ef08fd0dfc5c61211196
Reviewed-on: https://webrtc-review.googlesource.com/c/123442
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26760}
Most of the implementation in rtp_sender.cc is a copy paste for both
Audio & Video RTP senders. This change moves all the common behavior
into the base RtpSenderInternal class.
Template method pattern is used to accomodate for the very slight differences
between audio and video senders.
Bug: None
Change-Id: I6d4e93cd32fbb0fb361fd0e1883791019bde9a92
Reviewed-on: https://webrtc-review.googlesource.com/c/123411
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26758}
Added a layer in RtpSender that bridges the gap between the layers
that the user sees and the layer that the media engine sees.
Media engine still maintains the invariant that the number of layers
cannot be changed, while RtpSender adds and removes layers between
the user GetParameters and SetParameters calls and the media engine.
Bug: webrtc:10251
Change-Id: I33839c1f9a9052cb6130253e5a582606f2cbe54a
Reviewed-on: https://webrtc-review.googlesource.com/c/122641
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26756}
On Windows 10, the hidden taskbar won't be totally hidden, but having a
2 pixel margin on the screen. While a maximized app window will use up
the full screen, there will be overlapping between a hidden taskbar and
a maximized app window, which will impact window capture to that
maximized window. If the target window doesn't support GDI methods well,
the capture may be black (i.e. Chrome) or still (i.e. Word).
Because there is no solid way to identify a hidden taskbar window, we
have to make an exemption to the overlapping to a maximized window is
2-pixel X screen-width/height, which is thin enough to be noticed in
the cropping result.
Bug: chromium:838062
Change-Id: I9e0fbdf43b4445ca9fbbf5ed43bb266ae726a5b8
Reviewed-on: https://webrtc-review.googlesource.com/c/123261
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#26755}
