BGRA RTCCVPixelBuffers were cropped and scaled incorrectly. Libyuv’s
`ARGBScale` method is used in RTCCVPixelBuffer to scale and crop the
pixel buffer. To crop by `cropX` and `cropY` pixels, pointer
arithmetic is used to offset the src pointer of the original pixel
buffer bytes. There is a bug in how this offset is calculated.
The offset is done by `src += srcStride * _cropY + _cropX`. Libyuv
expects that the src pointer will point to the start of a new pixel.
However, if _cropX is a not a multiple of 4 (4 bytes for BGRA), the src
pointer will point to a byte in the middle of a pixel and thus libyuv
will incorrectly treat the data as the start of pixel (incorrectly
treating the first byte as red when it is actually green, etc...). To
fix this, the src pointer needs to be offset to always point to the
start of a new pixel.
Before this change:
Original Test Gradient image with a cropX of 2:
https://i.imgur.com/gSIgwGV.jpg
Scaled image (notice the colors are incorrect):
https://i.imgur.com/oPxbTEK.jpg
After this change:
Scaled image (notice the colors are correct):
https://i.imgur.com/dqBsmsH.jpg
A new unit test which tests scaling with cropX and cropY values has been
added. The test fails without this change and now passes with the
correct src pointer offsetting.
Bug: webrtc:9555
Change-Id: I87cbd7b91bc139d51fb4e11cc50ccb014cfa8051
Reviewed-on: https://webrtc-review.googlesource.com/89220
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24076}
This changeset adds the ability for API users to enable or disable GCM Cipher
suites from objective-c.
Bug: chromium:713701
Change-Id: I0ac7b60f55dd56bebbcfb315a542ef4843099802
Reviewed-on: https://webrtc-review.googlesource.com/89263
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24028}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script passing top level directories except rtc_base and api
find $@ -type f \( -name \*.h -o -name \*.cc -o -name \*.mm \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I9465c172e65ba6e6ed4e4fdc35b0b265038d6f71
Reviewed-on: https://webrtc-review.googlesource.com/84584
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23697}
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
Add a new flag to RtcConfiguration. By setting that flag to true, the
SRTP parameters will be reset whenever the DTLS transports are reset
after every offer/answer negotiation.
The flag is added to Android and Objc wrapper as well.
This should only be used as a workaround for the linked bug, if the
application knows that the other party is affected (for instance,
using a version number).
TBR=sakal@webrtc.org, denicija@webrtc.org
Bug: chromium:835958
Change-Id: I6db025e1c69bf83e1b1908f7df4627430db9920c
Reviewed-on: https://webrtc-review.googlesource.com/83101
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23587}
This provides an environment for testing out using WebRTC from an iOS
extension. It implements a ReplayKit broadcast extension for live
streaming games and screensharing.
The extension is only supported on iOS 11+ and is guarded by a build
flag.
Bug: webrtc:9335
Change-Id: Id218d6c73ef7599f5953c5a1e0e62e5d0dc4f10b
Reviewed-on: https://webrtc-review.googlesource.com/80000
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23504}
Expose this functionality in the Obj-C SDK to make it nicer to use for
Obj-C clients.
Bug: None
Change-Id: I5cb511af8799ac0fda15153d16f2550b848b93b2
Reviewed-on: https://webrtc-review.googlesource.com/80481
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23488}
This reverts commit fc4a9c933326cac2eb048eb507e63021c75e705e.
Reason for revert: Remote video is not showing in a video call.
Original change's description:
> Metal rendering should account for cropping.
>
> Also:
> - added a rotation override to allow ignoring frame rotation
> - fixed a couple of minor issues
> - made it possible to run the MTKView without the DisplayLink
>
> Bug: webrtc:9301
> Change-Id: Ia83c152d9b6d45d56ceb80d287b5d3eacfaebddd
> Reviewed-on: https://webrtc-review.googlesource.com/78282
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23452}
TBR=andersc@webrtc.org,kthelgason@webrtc.org,peterhanspers@webrtc.org
Change-Id: Iddf7793368531d2d7268c1ec138bb3a9874a4ab7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9301
Reviewed-on: https://webrtc-review.googlesource.com/80020
Reviewed-by: JT Teh <jtteh@webrtc.org>
Commit-Queue: JT Teh <jtteh@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23455}
Also:
- added a rotation override to allow ignoring frame rotation
- fixed a couple of minor issues
- made it possible to run the MTKView without the DisplayLink
Bug: webrtc:9301
Change-Id: Ia83c152d9b6d45d56ceb80d287b5d3eacfaebddd
Reviewed-on: https://webrtc-review.googlesource.com/78282
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23452}
Made some minor changes to resolve the issue. Only affects Debug builds.
NOTRY=TRUE
Bug: webrtc:9310
Change-Id: Ieeeb57d24b559282b2eefd4d8785f7cfe4f44e40
Reviewed-on: https://webrtc-review.googlesource.com/79624
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23434}
This removes usage of the old OnFailure methods on CreateSessionDescriptionObserver
and SetSessionDescriptionObserver, so that WebRTC will continue to compile
once all the default implementations are removed.
Bug: chromium:589455
Change-Id: Id67295b3ad0c30d24d79589c2041acdd507a19f3
Reviewed-on: https://webrtc-review.googlesource.com/78480
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23427}
This is a reland of 5faf36ef3c582350fba5ef97a3549e440d81a283
The issue in Chrome has been fixed and this should be safe to reland.
TBR=deadbeef
Original change's description:
> Implement RtpParameters.transaction_id for PC RtpSenderInterface
>
> The transaction_id field should be refreshed for every getParameters()
> call and checked at each setParameters() call.
> This also checks that getParameters() was ever called to return a proper
> error code.
>
> Bug: webrtc:7580
> Change-Id: I6c6fe289542e486fc422cdc61577982b0529d4c1
> Reviewed-on: https://webrtc-review.googlesource.com/70820
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23120}
Bug: webrtc:7580
Change-Id: Iabd41fb21afdf452c039d5513824ae334f8d1d3f
Reviewed-on: https://webrtc-review.googlesource.com/76980
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23247}
This change forks the existing iOS audio device module and audio device
from modules/audio_device/ into sdk/objc/Framework. It also updates
RTCPeerConnectionFactory to use the forked implementation.
The unit tests are re-implemented as XCTests.
(was: https://webrtc-review.googlesource.com/c/src/+/67300)
Bug: webrtc:9120
Change-Id: I07340505137b16c2dd487569ad0112f984557bba
Reviewed-on: https://webrtc-review.googlesource.com/75125
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23208}
Build targets involving files under api/video/ are moved into this
file, from api/BUILD.gn. In addition, drop "_api" part of target
names, and move the header file api/videosinkinterface.h to
api/video/video_sink_interface.h.
Bug: webrtc:9253
Change-Id: I2896d3f063db8dff902bc29738578395b2fcc155
Reviewed-on: https://webrtc-review.googlesource.com/75500
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23207}
This change forks the existing iOS audio device module and audio device
from modules/audio_device/ into sdk/objc/Framework. It also updates
RTCPeerConnectionFactory to use the forked implementation.
The unit tests are re-implemented as XCTests.
Bug: webrtc:9120
Change-Id: Ie60cafae796efbd7966d21ff6877c92cbe850fb7
Reviewed-on: https://webrtc-review.googlesource.com/67300
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23163}
