Commit Graph

117 Commits

Author SHA1 Message Date
5bd8cb74a6 Revert "Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5.""
This reverts commit a0adf3d4409036d095480e9bfa0fc06990362f84.

Reason for revert: Suspected of breaking chromium trybots, blocking
webrtc from rolling into chromium.

- First failed roll:
https://chromium-review.googlesource.com/c/chromium/src/+/1889997

- Second failed roll:
https://chromium-review.googlesource.com/c/chromium/src/+/1890837

Example failure:
https://ci.chromium.org/p/chromium/builders/try/linux-rel/230122
Log:
https://logs.chromium.org/logs/chromium/buildbucket/cr-buildbucket.appspot.com/8898155661969383856/+/steps/browser_tests__with_patch_/0/logs/Deterministic_failure:_DesktopCaptureApiTest.ChooseDesktopMedia__status_FAILURE_/0

Including lines like:
[12413:12413:1030/102514.183135:INFO:CONSOLE(0)] "[FAIL] screenShareWithAudioPermissionGetStream: NotReadableError: Could not start video source
Error", source: chrome-extension://knldjmfmopnpolahpmmgbagdohdnhkik/_generated_background_page.html (0)

Original change's description:
> Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5."
> 
> This is a reland of e7153012682ccd3d1eacc18f802cab7820e3bad3
> 
> Original change's description:
> > Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate entension version 1.5.
> >
> > Bug: chromium:396091
> > Change-Id: Ia1b36c771632c536bb8d15322461b479fabc409e
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148768
> > Commit-Queue: Sergey Ulanov <sergeyu@chromium.org>
> > Reviewed-by: Sergey Ulanov <sergeyu@chromium.org>
> > Cr-Commit-Position: refs/heads/master@{#29083}
> 
> Bug: chromium:396091
> Change-Id: I0d9171ae5f340e0489e4b45ce5d97bc52b0a4904
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156067
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29655}

TBR=zijiehe@chromium.org,tommi@webrtc.org,julien.isorce@chromium.org,sergeyu@chromium.org,trevor.axiom@gmail.com,jonringle@gmail.com

Change-Id: I2af6a0d5eaf74a0ee536d1c5440049a21d6f7dbf
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:396091
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158740
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29658}
2019-10-30 18:14:52 +00:00
a0adf3d440 Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5."
This is a reland of e7153012682ccd3d1eacc18f802cab7820e3bad3

Original change's description:
> Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate entension version 1.5.
>
> Bug: chromium:396091
> Change-Id: Ia1b36c771632c536bb8d15322461b479fabc409e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148768
> Commit-Queue: Sergey Ulanov <sergeyu@chromium.org>
> Reviewed-by: Sergey Ulanov <sergeyu@chromium.org>
> Cr-Commit-Position: refs/heads/master@{#29083}

Bug: chromium:396091
Change-Id: I0d9171ae5f340e0489e4b45ce5d97bc52b0a4904
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156067
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29655}
2019-10-30 12:20:20 +00:00
682dabd1c1 Add RTCStatisticsReport.h to WebRTC.framework.
Bug: webrtc:11041
Change-Id: I2ae5a7db9697f70426feaf0c31ee4e0b9b654cc6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157800
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29554}
2019-10-21 11:16:10 +00:00
36d171b73f Add Ramprakash Jelari to AUTHORS.
For contributions relating to CL:
https://webrtc-review.googlesource.com/c/src/+/68520

In discussion:
https://groups.google.com/d/msg/discuss-webrtc/Tqa-YzX5K4I/Nhxxsf9EBwAJ

Bug: None
Change-Id: Ic7f3cbfab63cacecdfd3e2c201a18a8e3971b58b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156942
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29476}
2019-10-15 08:07:26 +00:00
fa77ba6af1 SetStreams API of RtpSender wrapped for iOS and Android
Bug: webrtc:10129
Change-Id: I36ea0110de655bbffa2bd18a024abd15a2136838
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155983
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29405}
2019-10-08 13:51:19 +00:00
63173d5bef pipewire: handle deleting the capturer while a D-Bus call is in progress
If a D-Bus call is in progress when a BaseCapturerPipeWire is deleted, then
the user_data is invalid when the callback function is called. This results
in memory corruption.

To fix this, use a GCancellable. If it is canceled, the callback will be
called with a corresponding error. Detect this error and abort before
accessing the user_data.

Note: The first argument is the 'source_object'. For g_dbus_proxy_call()
this is the proxy object not the connection. This was not a problem before,
because it was not used.

Bug: None
Change-Id: I8d5e3fb5c49fcc9afd61cdb8e8249f78b9434faf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149817
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29326}
2019-09-26 18:58:56 +00:00
ba5f8e94c8 Revert "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate entension version 1.5."
This reverts commit e7153012682ccd3d1eacc18f802cab7820e3bad3.

Reason for revert: Suspect of breaking build on linux_rel Chromium bot, preventing rolls into Chromium.

Sample log message:

FAILED: deb_chrome.deps 
python ../../chrome/installer/linux/debian/calculate_package_deps.py chrome ../../build/linux/debian_sid_amd64-sysroot x64 deb_chrome.deps --distro-check
Dependency libxrandr2 (>= 2:1.5.0) not satisfiable on distro Debian 8 (Jessie) caused by binary chrome
[71411/71430] LINK ./browser_tests
[71412/71430] LINK ./unit_tests
ninja: build stopped: subcommand failed.

Sample failed roll: https://chromium-review.googlesource.com/c/chromium/src/+/1788844

Original change's description:
> Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate entension version 1.5.
> 
> Bug: chromium:396091
> Change-Id: Ia1b36c771632c536bb8d15322461b479fabc409e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148768
> Commit-Queue: Sergey Ulanov <sergeyu@chromium.org>
> Reviewed-by: Sergey Ulanov <sergeyu@chromium.org>
> Cr-Commit-Position: refs/heads/master@{#29083}

TBR=zijiehe@chromium.org,tommi@webrtc.org,julien.isorce@chromium.org,sergeyu@chromium.org,trevor.axiom@gmail.com,jonringle@gmail.com

Change-Id: I4a07213951a14eae524427c0a92d0cad72ade84c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:396091
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151761
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29088}
2019-09-06 05:36:23 +00:00
e715301268 Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate entension version 1.5.
Bug: chromium:396091
Change-Id: Ia1b36c771632c536bb8d15322461b479fabc409e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148768
Commit-Queue: Sergey Ulanov <sergeyu@chromium.org>
Reviewed-by: Sergey Ulanov <sergeyu@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29083}
2019-09-05 21:19:41 +00:00
1e00dbccc2 Stun server should return XOR-MAPPED-ADDRESS/MAPPED-ADDRESS correctly
* Return xor mapped address for RFC5389 compatible client
* fix a typo in function name
* update stunserver unitest case
* update author

Bug: webrtc:10764
Change-Id: I466799744a343508233c18b7c477d2212680392a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143841
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28421}
2019-06-28 19:12:14 +00:00
08f6a6c672 Import proto_library.gni when rtc_enable_protobuf is true
Import proto_library.gni when rtc_enable_protobuf is true instead of when
build_with_mozilla is false.

Makes it maybe easier to reason about the intention (e.g. intention is to not
compile any protobuf in, hence flag rtc_enable_protobuf)

The build file could not work if build_with_mozilla = true but
rtc_enable_protobuf = true.

Bug: webrtc:10338
Change-Id: I26e5983bd1519aa46c308b11796d518de5ef7597
Reviewed-on: https://webrtc-review.googlesource.com/c/123763
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26868}
2019-02-27 09:56:42 +00:00
ce27875b83 [AndroidAudioRecord] Added audio format parameter to configure AudioRecord - JavaAudioDeviceModule
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}
2019-02-20 15:08:53 +00:00
8905d04109 Add ',' between elements in RTCStatsReport::ToJson
There was no ',' between array-elements the stats-report, yielding
invalid JSON. Let's bring back the ",", which was lost during recent
refactoring.

Bug: webrtc:10173
Change-Id: Ib58025d56c4895c6af33b9777cb2ebdb94a678ea
Reviewed-on: https://webrtc-review.googlesource.com/c/116483
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26161}
2019-01-08 16:05:23 +00:00
74cdf7874d add cstring include need for strncmp
Propose resolution of Issue 10011 : (GCC) build fails desktop_capturer.cc:66:66: error: ‘strncmp’ was not declared in this scope

Bug: webrtc:10011
Change-Id: I4afdfd96f8bbc8e39380a365138ab79e237568e3
Reviewed-on: https://webrtc-review.googlesource.com/c/111885
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Reviewed-by: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25790}
2018-11-26 20:49:36 +00:00
83aa5ace99 Adding Microsoft Corporation (*@microsoft.com) to WebRTC AUTHORS
Bug: None
Change-Id: I7c425b7ca48580d87757db7a70db30fcbe259adb
Reviewed-on: https://webrtc-review.googlesource.com/c/110360
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25642}
2018-11-14 18:09:23 +00:00
318da51f99 Reland "Add support for screen sharing with PipeWire on Wayland"
The content_unittests failure was caused by wrong path in the cfi
blacklist (when the files from x11 folder were moved to the linux
folder by this change).

Bug: chromium:682122
Change-Id: I4f7f6c5a73a981feeac18494749f85935e812981
Reviewed-on: https://webrtc-review.googlesource.com/c/110461
Commit-Queue: Tomáš Popela <tomas.popela@gmail.com>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25621}
2018-11-13 15:05:05 +00:00
3ddaf3c408 Revert "Add support for screen sharing with PipeWire on Wayland"
This reverts commit dd20c9c1e3f681f6c33d1879c76f588bd4b095bd.

Reason for revert: Speculative revert; looks like it causes crashes on official builders. See crbug.com/901319.

Original change's description:
> Add support for screen sharing with PipeWire on Wayland
>
> Currently, when users want to use the screen sharing and are using the
> Wayland display server (the default on Fedora distribution), then it
> doesn't work, because the WebRTC only includes the X11 implementation.
> This change adds the support by using the PipeWire multimedia server.
>
> The PipeWire implementation in WebRTC stays in
> screen-capturer-pipewire.c and is guarded by the rtc_use_pipewire build
> flag that is automatically enabled on Linux.
>
> More information are included in the relevant commit messages.
>
> Tested on the current Chromium master and Firefox.
>
> The sysroot changes are requested in:
> https://chromium-review.googlesource.com/c/chromium/src/+/1258174
>
> Co-authored-by: Jan Grulich <grulja@gmail.com>
> Co-authored-by: Eike Rathke <erathke@redhat.com>
> Change-Id: I212074a4bc437b99a77bf383266026c5bfae7c4a
>
> BUG=chromium:682122
>
> Change-Id: I212074a4bc437b99a77bf383266026c5bfae7c4a
> Reviewed-on: https://webrtc-review.googlesource.com/c/103504
> Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Brave Yao <braveyao@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25461}

TBR=phoglund@webrtc.org,jamiewalch@chromium.org,niklas.enbom@webrtc.org,braveyao@webrtc.org,tomas.popela@gmail.com

# Not skipping CQ checks because original CL landed > 1 day ago.

NOPRESUBMIT=true

Bug: chromium:682122, chromium:901319
Change-Id: I4ca5da77daea73cae1232953a0d633900a85a93d
Reviewed-on: https://webrtc-review.googlesource.com/c/109584
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25522}
2018-11-06 14:17:34 +00:00
bf0d0c1b30 Add IPv6 configuration parameters to iOS API
Adds |disableIPV6| and |disableIPV6OnWiFi| properties to
RTCConfiguration

Bug: None
Change-Id: Id59fb2002afadd7817f7caeaa62231bf90ecb274
Reviewed-on: https://webrtc-review.googlesource.com/c/109280
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25496}
2018-11-05 10:56:10 +00:00
dd20c9c1e3 Add support for screen sharing with PipeWire on Wayland
Currently, when users want to use the screen sharing and are using the
Wayland display server (the default on Fedora distribution), then it
doesn't work, because the WebRTC only includes the X11 implementation.
This change adds the support by using the PipeWire multimedia server.

The PipeWire implementation in WebRTC stays in
screen-capturer-pipewire.c and is guarded by the rtc_use_pipewire build
flag that is automatically enabled on Linux.

More information are included in the relevant commit messages.

Tested on the current Chromium master and Firefox.

The sysroot changes are requested in:
https://chromium-review.googlesource.com/c/chromium/src/+/1258174

Co-authored-by: Jan Grulich <grulja@gmail.com>
Co-authored-by: Eike Rathke <erathke@redhat.com>
Change-Id: I212074a4bc437b99a77bf383266026c5bfae7c4a

BUG=chromium:682122

Change-Id: I212074a4bc437b99a77bf383266026c5bfae7c4a
Reviewed-on: https://webrtc-review.googlesource.com/c/103504
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25461}
2018-11-01 08:46:38 +00:00
0eb7d3ff35 Always call ConvertToI420 with positive crop_height
Source height may be negative, causing libyuv to invert the image.
However the height of the destination buffer specified by crop_height
should be positive. Remaining calls in common_video_unittests are valid.

Bug: webrtc:9447
Change-Id: I6d398909ae80a99d228ccbbd8c1d7ae804e5bf8d
Reviewed-on: https://webrtc-review.googlesource.com/c/86540
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25427}
2018-10-30 12:02:32 +00:00
6fcf6ca710 Modified PressEnterToContinue() to actualy check if Enter is pressed
Modified PressEnterToContinue() to run the Windows message loop in the
context of the SingleThreadedTaskQueueForTesting thread. The previous
PressEnterToContinue() was running the message loop in the context of
the main thread, but the "Local Preview" and "Loopback Video #0" are
created in the context of the SingleThreadedTaskQueueForTesting thread
and the message loop must be executed in the context of the thread that
created these windows in order for these windows to respond to any
event.

BUG=webrtc:9123

Change-Id: I2ec19f2569a940a510d3b2bd3881a89032d70332
Reviewed-on: https://webrtc-review.googlesource.com/c/67520
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25408}
2018-10-29 10:21:24 +00:00
d2fb1bfeba Generate module.modulemap file when building Mac Framework
Without this file, the Framework can't be used by Swift projects.

Bug: webrtc:9142
Change-Id: I8803ec8b194dc116e133257e205f4620bb34a692
Reviewed-on: https://webrtc-review.googlesource.com/c/103340
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25047}
2018-10-08 12:32:45 +00:00
289e980708 Remove unused var in device info bits from video capture module for Linux
Bug: None
Change-Id: Icea40fe58e7f65cd1eb311c456ce3cdc802f88a8
Reviewed-on: https://webrtc-review.googlesource.com/97421
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24700}
2018-09-12 10:36:33 +00:00
e0c8b230e7 Frame marking RTP header extension (PART 1: implement extension)
Bug: webrtc:7765
Change-Id: I23896d121afd6be4bce5ff4deaf736149efebcdb
Reviewed-on: https://webrtc-review.googlesource.com/85200
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24695}
2018-09-11 22:35:30 +00:00
8cec4fb6c2 Use default RTCConfiguration on iOS
With "aggressive" preset the default bundlePolicy is set to "maxBundle" when it shoud be "balanced" according to spec.

Bug: webrtc:9458
Change-Id: Ifbdd76be3a6d9968574cba857f178d5f859dcb87
Reviewed-on: https://webrtc-review.googlesource.com/88567
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24650}
2018-09-10 12:16:53 +00:00
ccee56beee Add certificate generate/set functionality to bring iOS closer to JS API
The JS API supports two operations which have never been implemented in
the iOS counterpart:
 - generate a new certificate
 - use this certificate when creating a new PeerConnection

Both functions are illustrated in the generateCertificate example code:
 - https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/generateCertificate

Currently, on iOS, a new certificate is automatically generated for
every PeerConnection with no programmatic way to set a specific
certificate.

Work sponsored by |pipe|

Bug: webrtc:9498
Change-Id: Ic1936c3de8b8bd18aef67c784727b72f90e7157c
Reviewed-on: https://webrtc-review.googlesource.com/87303
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24276}
2018-08-13 22:25:15 +00:00
25cc8ad198 Fixed issue with BGRA RTCCVPixelBuffer scale and crop
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}
2018-07-24 08:23:26 +00:00
e250645ea4 Call callback in IDLE state
In current state, if you want to do something with the capturer (eg. switch to next camera again) it fails with an exception that camera switch is already in progress.

Change-Id: I908eb590b54fdf3346441097b39f1f2a2eb56ce8

Bug: webrtc:9527
Change-Id: I908eb590b54fdf3346441097b39f1f2a2eb56ce8
Reviewed-on: https://webrtc-review.googlesource.com/88700
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23995}
2018-07-17 07:27:37 +00:00
43800f95bf Generalize SimulcastEncoderAdapter, use for H264 & VP8.
* Move SimulcastEncoderAdapter out under modules/video_coding
* Move SimulcastRateAllocator back out to modules/video_coding/utility
* Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility
* Move any VP8 specific code - such as temporal layer bitrate budgeting -
  under codec type dependent conditionals.
* Plumb the simulcast index for H264 in the codec specific and RTP format data structures.

TBR=sprang@webrtc.org,stefan@webrtc.org,titovartem@webrtc.org

Bug: webrtc:5840
Change-Id: I2d3b130622dd7ceec5528f3ab6c46f109e6bafb8
Reviewed-on: https://webrtc-review.googlesource.com/84743
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23715}
2018-06-21 15:57:43 +00:00
6f440ed5b5 Revert "Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8."
This reverts commit 07efe436c9002e139845f62486e3ee4e29f0d85b.

Reason for revert: Breaks downstream project.

cricket::GetSimulcastConfig method signature has been updated.
I think you can get away with a default value for temporal_layers_supported (and then you can remove it after a few days when projects will be updated).


Original change's description:
> Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8.
> 
> * Move SimulcastEncoderAdapter out under modules/video_coding
> * Move SimulcastRateAllocator back out to modules/video_coding/utility
> * Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility
> * Move any VP8 specific code - such as temporal layer bitrate budgeting -
>   under codec type dependent conditionals.
> * Plumb the simulcast index for H264 in the codec specific and RTP format data structures.
> 
> Bug: webrtc:5840
> Change-Id: Ieced8a00e38f273c1a6cfd0f5431a87d07b8f44e
> Reviewed-on: https://webrtc-review.googlesource.com/64100
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23705}

TBR=sprang@webrtc.org,stefan@webrtc.org,mflodman@webrtc.org,hta@webrtc.org,sergio.garcia.murillo@gmail.com,titovartem@webrtc.org,agouaillard@gmail.com

Change-Id: Ic9d3b1eeaf195bb5ec2063954421f5e77866d663
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:5840
Reviewed-on: https://webrtc-review.googlesource.com/84760
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23710}
2018-06-21 13:41:14 +00:00
07efe436c9 Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8.
* Move SimulcastEncoderAdapter out under modules/video_coding
* Move SimulcastRateAllocator back out to modules/video_coding/utility
* Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility
* Move any VP8 specific code - such as temporal layer bitrate budgeting -
  under codec type dependent conditionals.
* Plumb the simulcast index for H264 in the codec specific and RTP format data structures.

Bug: webrtc:5840
Change-Id: Ieced8a00e38f273c1a6cfd0f5431a87d07b8f44e
Reviewed-on: https://webrtc-review.googlesource.com/64100
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23705}
2018-06-21 12:23:03 +00:00
b90e63c620 Fix: NetEq PacketBuffer logs discarded packet with wrong codec level when new packet replaces the lower level packet
Bug: webrtc:9370
Change-Id: I59606ef6ea9bbf26de844a2fd3f597856271a86a
Reviewed-on: https://webrtc-review.googlesource.com/81700
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23555}
2018-06-08 14:58:18 +00:00
e7e0602a0d ObjC: Notify local video track
The macOS demo add camera preview in didReceiveLocalVideoTrack callback, but this callback is never called.

Bug: webrtc:9276
Change-Id: I60b9cc69672f3654d4e36de0e8140e0bbb957540
Reviewed-on: https://webrtc-review.googlesource.com/77950
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23458}
2018-05-30 22:36:14 +00:00
f8d8d6d00c Use range-based-for instead of std::for_each and std::mem_fun
std::mem_fun is deprecated in C++11, and removed in C++17. Using C++17
option for building libwebrtc causes build failure. This is found during
upgrading WebKit tree from C++14 to C++17.
This patch replaces std::for_each and std::mem_fun with range-based-for.
We also merge loops for streams_ into one.

Bug: webrtc:9277
Change-Id: I44a7e44ea21fc33ffa9a586ddfea570f97dfacb6
Reviewed-on: https://webrtc-review.googlesource.com/77280
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23285}
2018-05-17 13:51:02 +00:00
ebd9abc1a2 Use IFA_LOCAL instead of IFA_ADDRESS over IPv4 network on ANDROID
IFA_ADDRESS gives DESTINATION address in case of point-to-point
connection, which is not able to create ports for candidate gathering.
Use IFA_LOCAL to avoid this problem.

Bug: webrtc:9189
Change-Id: Ifcb1955b1b4011dc69c93d99b4e223b370dc16eb
Reviewed-on: https://webrtc-review.googlesource.com/69620
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23055}
2018-04-27 14:16:01 +00:00
a72b7fc30a ObjC: Add missing _lastDrawnFrame assignments
Currently there are several checks against _lastDrawnFrame in RTCEAGLVideoView.mm but this variable is not assigned anywhere. Seems like it was missed in 13941912b1 during work on injecting custom shaders.

Bug: webrtc:9133
Change-Id: Ie979a63de343e7253e4b4e70e3b98ffb0880af04
Reviewed-on: https://webrtc-review.googlesource.com/68720
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22819}
2018-04-11 12:51:06 +00:00
3cfe9e167e Fixed video capturing on Mac.
On specific Macbooks (no exact pattern, unfortunately),
video from an integrated camera is not captured.
Changed AVCaptureVideoDataOutput pixel format configuration
as in Chromium which solved the problem.
https://chromium.googlesource.com/chromium/src/media/+/master/capture/video/mac/video_capture_device_avfoundation_mac.mm
FourCharCode best_fourcc = kCVPixelFormatType_422YpCbCr8;

Tested with external cameras as well.

Bug: webrtc:8958
Change-Id: Ib99382b38d1914e2963761a33df310024524c9a4
Reviewed-on: https://webrtc-review.googlesource.com/58880
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22709}
2018-04-03 16:23:01 +00:00
2870b0a57e Expose a link-local network interfaces enumeration option
The bug 8432 is caused by trying to connect through a
"link-local" interface (IP address 169.254.0.x/16),
which is listed among the iPhone network interfaces.
The bug is not happening if the link-local network interfaces
are skipped in the ICE candidate gethering process.

To control this behaviour an option - disable_link_local_networks -
is added inside the RTCConfiguration.
It is used to set the new BasicPortAllocatorSession flag -
PORTALLOCATOR_DISABLE_LINK_LOCAL_NETWORKS.
The port allocator flag is added if the configuration option is set.

IPIsLinkLocal IPAddress function and its friends (IPIsLoopback, IPIsPrivate)
are refactored to work on both IPv4 and IPv6.
Unit test IPIsLinkLocal.

Bonus: fix a bug in IPIsLinkLocalV6:
take into account just 10 network mask bits instead of 16.

Bug: webrtc:8432
Change-Id: Ibe8f677a36098057b7fcad5c798380727b23359b
Reviewed-on: https://webrtc-review.googlesource.com/36380
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21922}
2018-02-06 19:12:04 +00:00
5e4833cc90 Add missing stdio.h header in files using scanf/sscanf function.
Various files in webrtc codebase use scanf/sscanf function without
including stdio.h header file which is supposed to define it. This
somehow works when using glibc, but fails with uClibc.

Bug: webrtc:8641
Change-Id: Ie4ae17af32b32ed8cea567166b6b0e5193966995
Reviewed-on: https://webrtc-review.googlesource.com/32261
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21775}
2018-01-26 13:15:52 +00:00
b8874356f6 RemoteBitrateEstimatorAbsSendTime: check clock is a valid ref
Bug: webrtc:8607
Change-Id: Idc3b6c0b3896381f0140584d8c2952ee26db1646
Reviewed-on: https://webrtc-review.googlesource.com/31320
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21623}
2018-01-16 01:27:11 +00:00
d3c642bc1f Fix typo in the include path of ooura_fft.h
Bug: None
Change-Id: Iaac4a80f75dcd81ab0d2665cb20f27f0342cb17d
Reviewed-on: https://webrtc-review.googlesource.com/38441
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21565}
2018-01-11 07:57:40 +00:00
6728003bcf Skip H246 scaling lists in SPS packets
This code is originally written by marc@frankensteinmotorworks.com

Bug: webrtc:8275
Change-Id: I35e6d21b12e71199e0209ff91740d95c9df3bd10
Reviewed-on: https://webrtc-review.googlesource.com/36520
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21528}
2018-01-09 10:22:30 +00:00
5a7508ab24 Fixed NPE inside org.webrtc.Camera1Session.create
On some devices `android.hardware.Camera.open` returns null
instead of raising exception. It causes `NPE` inside
`Camera1Session.create` when method `setPreviewTexture` is
invoked on local variable `camera`, which is `null`.

Bug: webrtc:8658
Change-Id: Ic65b4aef2c0b8b65735a9db02433b536bfe92ddd
Reviewed-on: https://webrtc-review.googlesource.com/33620
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21352}
2017-12-19 10:01:20 +00:00
1c62ffa530 Normalize main(..) routines for WinUWP
In order to support WinUWP platform, all main(..) routines must be normalized to the formal int main(int argc, char* argv[]) form. A platform wrapper main is auto-created linking against the default main(...). This can only work if the linkage is exactly matching the proper formal definition and not a loosely defined main(...) alternative.

Bug: webrtc:8608
Change-Id: I606663aaea7df1792c7c5636279617b8926fa5cc
Reviewed-on: https://webrtc-review.googlesource.com/28721
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21229}
2017-12-12 14:32:56 +00:00
12e555b715 Delete wrapper API ConvertToI420 for YUV conversion to I420
Directly use the libyuv API for YUV conversion to I420

Bug: None
Change-Id: Iea6e8fa8f7179c800ea850305170002398cb00dc
Reviewed-on: https://webrtc-review.googlesource.com/17260
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20681}
2017-11-15 11:10:20 +00:00
149533abd4 Move rendering code in SurfaceViewRenderer to a separate class.
The new SurfaceEglRenderer helper class extends EglRenderer and
implements rendering on a SurfaceView.

Bug: webrtc:8242
Change-Id: Ic532fe487755d3b54c6bd03f239d714e1ecb10ad
Reviewed-on: https://webrtc-review.googlesource.com/2940
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20562}
2017-11-06 13:52:26 +00:00
e21be1db4c Reland of Fix the video buffer size should take rtt into consideration (patchset #2 id:160001 of https://codereview.chromium.org/3002033002/ )
Reason for revert:
Fixes has landed.

Original issue's description:
> Revert of Fix the video buffer size should take rtt into consideration (patchset #3 id:40001 of https://codereview.chromium.org/2980413002/ )
>
> Reason for revert:
> We are not certain this is the behavior we want.
>
> Original issue's description:
> > Fix the video buffer size should take rtt into consideration
> >
> > BUG=webrtc:8010
> >
> > Review-Url: https://codereview.webrtc.org/2980413002
> > Cr-Commit-Position: refs/heads/master@{#19285}
> > Committed: f1e08d0b58
>
> TBR=sprang@webrtc.org,gustavogb@gmail.com
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:8010
>
> Review-Url: https://codereview.webrtc.org/3002033002
> Cr-Commit-Position: refs/heads/master@{#19442}
> Committed: bdbc8895f3

TBR=sprang@webrtc.org,gustavogb@gmail.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:8010

Review-Url: https://codereview.webrtc.org/3016633002
Cr-Commit-Position: refs/heads/master@{#19944}
2017-09-25 13:37:12 +00:00
bdbc8895f3 Revert of Fix the video buffer size should take rtt into consideration (patchset #3 id:40001 of https://codereview.chromium.org/2980413002/ )
Reason for revert:
We are not certain this is the behavior we want.

Original issue's description:
> Fix the video buffer size should take rtt into consideration
>
> BUG=webrtc:8010
>
> Review-Url: https://codereview.webrtc.org/2980413002
> Cr-Commit-Position: refs/heads/master@{#19285}
> Committed: f1e08d0b58

TBR=sprang@webrtc.org,gustavogb@gmail.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:8010

Review-Url: https://codereview.webrtc.org/3002033002
Cr-Commit-Position: refs/heads/master@{#19442}
2017-08-22 09:08:51 +00:00
f1e08d0b58 Fix the video buffer size should take rtt into consideration
BUG=webrtc:8010

Review-Url: https://codereview.webrtc.org/2980413002
Cr-Commit-Position: refs/heads/master@{#19285}
2017-08-09 12:43:08 +00:00
f3a48ab6dc Delete unused field from AndroidVideoTrackSource
BUG=None

Review-Url: https://codereview.webrtc.org/2974713002
Cr-Commit-Position: refs/heads/master@{#19117}
2017-07-24 08:06:39 +00:00
ff7acb19a1 Reset isFirstFrameRendered on init of SurfaceViewRenderer
If a SurfaceViewRenderer is reinitialized, the onFirstFrameRendered
callback is not fired.

Ensure that we reset the flag when the SurfaceViewRenderer is
initialized.

BUG=webrtc:7985

Review-Url: https://codereview.webrtc.org/2981793002
Cr-Commit-Position: refs/heads/master@{#19016}
2017-07-14 09:35:53 +00:00