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}
Using fully qualified paths to include libyuv headers allows WebRTC to
avoid to rely on the //third_party/libyuv:libyuv_config target to
set the -I compiler flag.
Today some WebRTC targets depend on //third_party/libyuv only to
include //third_party/libyuv:libyuv_config but with fully qualified
paths this should not be needed anymore.
A follow-up CL will remove //third_party/libyuv from some targets that
don't need it because they are not including libyuv headers.
Bug: webrtc:8605
Change-Id: Icec707ca761aaf2ea8088e7f7a05ddde0de2619a
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/28220
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21209}
This moves all WebRTC internal code from using
SessionDescriptionInterface::type() which returns a string and
from using CreateSessionDescription with a string type parameter.
Bug: webrtc:8613
Change-Id: I1cdd93dc4b26dec157e22476fdac569d5da2810a
Reviewed-on: https://webrtc-review.googlesource.com/29500
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21147}
When audio_only is on for the webrtc unity plugin, there is a bug that
the audio from hologram cannot be heard at the remote side.
Actually we found the audio is transmitted to the remote side, but the
remote side wants video data also to playout everything. So without
video data, the remote side will drop all the audio data.
Thus, on the hologram (using webrtc unity plugin) side, we should not
hook up a dummy camera, but instead we should use media constraint to
request the remote side to send video data.
This CL fixes the bug.
Bug: webrtc:8555
Change-Id: I21ddda65185b645088aa4ac15f47b3f8ffad1873
Reviewed-on: https://webrtc-review.googlesource.com/24680
Commit-Queue: Qiang Chen <qiangchen@chromium.org>
Reviewed-by: George Zhou <gyzhou@chromium.org>
Cr-Commit-Position: refs/heads/master@{#21094}
Previously, wrapped native codec instances would leak the native object
if it was never used. This change fixes it by changing getNative method
to createNative.
Also fixes "Video codec hardware acceleration" setting in AppRTCMobile.
Bug: webrtc:7925
Change-Id: I53f6dc1dd5e37dea8d14278423122dede17719c5
Reviewed-on: https://webrtc-review.googlesource.com/24881
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20859}
This is a debug feature and should be disabled by default. Any client
that needs this functionality should call setEnableInternalTracer.
Bug: webrtc:8553
Change-Id: I78d718ebb95fc5cb8c464327b5b36e385ccfa9c0
Reviewed-on: https://webrtc-review.googlesource.com/24540
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20814}
Update the output pixel format to the client supplied format when
starting camera capture.
Also add a new API method to get the preferred output pixel format
according to the
AVCaptureVideoDataOutput#availableVideoCVPixelFormatTypes method and
use it in AppRTCMobile.
Bug: webrtc:8505
Change-Id: Ia24eaf91d70d0703a34d38b06bb6eea28fb922b8
Reviewed-on: https://webrtc-review.googlesource.com/22680
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Zeke Chin <tkchin@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20697}
This means we can properly declare the dependency between
libjingle_peerconnection_api and video_frame_api. i420
pulls in system_wrappers, which can't be a dependency of
the public API.
Plan:
1) Land this CL + send out PSA
2) Make all direct users of i420_buffer depend on the
new video_frame_api_i420 target
3) Move i420_buffer.cc to the new target
4) Make libjingle_peerconnection_api depend on
video_frame_api, since it no longer contains i420 code
Bug: webrtc:7504
Change-Id: I30d90f2ac7af53748859bbde8aed92386d5501f9
Reviewed-on: https://webrtc-review.googlesource.com/9382
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20656}
This is similar to https://webrtc-review.googlesource.com/c/src/+/3620
for iOS.
Using the new WebRtcMediaEngineFactory::Create API, the built-in
software video codecs are no longer appended to the injected codecs.
To be able to use the software codecs, they are exposed as Java
classes through SoftwareVideoEncoderFactory etc.
There is also a new DefaultVideoEncoderFactory used by AppRTCMobile.
This factory tries to use hardware implementations where available,
but falls back to using the injected software codecs.
The HardwareVideoEncoderFactory is temporarily also falling back on
the software codecs in its default configuration in order to
maintain backwards compatibility.
Bug: webrtc:7925
Change-Id: I3e8c5ed492ccd160aca968986ad217d7978a951c
Reviewed-on: https://webrtc-review.googlesource.com/17480
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20647}
To achieve this, the CL does the following
- Adds sample mp4 video
- Refactors the existing RTCFileVideoCapturer to achieve continious
capture and adds tests.
Bug: webrtc:8406
Change-Id: Ibc0891176c58ec9053b42e340d2113036e7199ec
Reviewed-on: https://webrtc-review.googlesource.com/12180
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Daniela Jovanoska Petrenko <denicija@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20598}
In https://chromium-review.googlesource.com/c/chromium/src/+/750645
Chromium started to use an ErrorProne plugin to discourage synchronized
public methods (an encourage the usage of synchronized blocks).
In order to unblock the Chromium Roll we can suppress these warnings
and decide if we want to align with Chromium on this check or ask
them to make it optional.
More details in the bug.
TBR=magjed@webrtc.org
Bug: webrtc:8491
Change-Id: Ie77a324e54aab44a4f59853959549f1d21f884a0
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/20060
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20569}
This is a reland of 30915a742d86df55ac5c04501c0e8104675a612e
Original change's description:
> Simple Default ObjC video codec factories.
>
> Move the simple video encoder/decoder factory from AppRTCMobile into the
> public API so users who don't have special requirements for video codecs
> can easily get started.
>
> Also clean up the API a little.
>
> This CL replaces the more flexible default factories in
> https://webrtc-review.googlesource.com/c/src/+/7741 and clients that
> want to implement their own codecs will have to supply their own
> encoder/decoder factories as well. The benefits of the approach in
> this CL are a simpler API and less effects on the rest of the code.
>
> Bug: None
> Change-Id: I4ed94090d778b4fc38b49864de1d4de4ff125d6a
> Reviewed-on: https://webrtc-review.googlesource.com/15141
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20441}
Bug: None
Change-Id: If0910cc540dc835dfec4eeb5bea527d88482d110
Reviewed-on: https://webrtc-review.googlesource.com/16780
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20476}
The ObjC API (the files in sdk/objc/Framework/Headers/WebRTC/) needs to
be pure ObjC. The changes that are reverted here introduced C++ which
turns it into ObjC++.
We don't have a test protectcing this right now, but it's probably
something we should add to catch changes like this in the future.
TBR=alexnarest@webrtc.org,deadbeef@webrtc.org
Bug: webrtc:8243
Change-Id: Idea688f4014cd44c27cf2cb2a5ec8a9ea7da3c00
Reviewed-on: https://webrtc-review.googlesource.com/16429
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20463}
This is done in preparation to make all javac warnings into errors for
WebRTC targets.
Bug: webrtc:6597
Change-Id: I402043157bd75943adf0de52111e5a1bb179c6d1
Reviewed-on: https://webrtc-review.googlesource.com/15104
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20450}
Apparently WebSocketObserver gets garbage collected if it is not stored
by us. This caused some external tests to break.
Bug: None
Change-Id: If62786e84f84a5a63172d67962bb4de8ae3e8479
Reviewed-on: https://webrtc-review.googlesource.com/16100
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20449}
Summary:
Adds AudioTrackStartErrorCode to separate different types of error
codes in combination with StartPlayout.
Harmonizes WebRtcAudioRecord and WebRtcAudioTrack implementations
to ensure that init/start/stop is performed identically.
Adds thread checking in WebRtcAudio track.
Bug: webrtc:8453
Change-Id: Ic913e888ff9493c9cc748a7b4dae43eb6b37fa85
Reviewed-on: https://webrtc-review.googlesource.com/15481
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20448}
This reverts commit 30915a742d86df55ac5c04501c0e8104675a612e.
Reason for revert: Breaks downstream.
Original change's description:
> Simple Default ObjC video codec factories.
>
> Move the simple video encoder/decoder factory from AppRTCMobile into the
> public API so users who don't have special requirements for video codecs
> can easily get started.
>
> Also clean up the API a little.
>
> This CL replaces the more flexible default factories in
> https://webrtc-review.googlesource.com/c/src/+/7741 and clients that
> want to implement their own codecs will have to supply their own
> encoder/decoder factories as well. The benefits of the approach in
> this CL are a simpler API and less effects on the rest of the code.
>
> Bug: None
> Change-Id: I4ed94090d778b4fc38b49864de1d4de4ff125d6a
> Reviewed-on: https://webrtc-review.googlesource.com/15141
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20441}
TBR=magjed@webrtc.org,andersc@webrtc.org,kthelgason@webrtc.org
Change-Id: I3d4395cc9667e6c6cdb33a3b0f5c5fb5bfde9028
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/15182
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20446}
Move the simple video encoder/decoder factory from AppRTCMobile into the
public API so users who don't have special requirements for video codecs
can easily get started.
Also clean up the API a little.
This CL replaces the more flexible default factories in
https://webrtc-review.googlesource.com/c/src/+/7741 and clients that
want to implement their own codecs will have to supply their own
encoder/decoder factories as well. The benefits of the approach in
this CL are a simpler API and less effects on the rest of the code.
Bug: None
Change-Id: I4ed94090d778b4fc38b49864de1d4de4ff125d6a
Reviewed-on: https://webrtc-review.googlesource.com/15141
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20441}
peerConnectionParameters.videoCodec can be null in some cases.
Previously, this would cause a crash in AppRTCMobile.
Bug: b/67938523
Change-Id: I30ebf0f91fad23a3cf34946736b9f4e6c266277f
Reviewed-on: https://webrtc-review.googlesource.com/14200
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20372}
Always use gn.py in depot_tools instead of just gn.
The https://cs.chromium.org/chromium/src/build/find_depot_tools.py
is looking up the DEPS-pinned copy in third_party/depot_tools
and adds it to the path when add_depot_tools_to_path() is called.
Similar use:
https: //cs.chromium.org/search/?q=%22find_depot_tools.add_depot_tools_to_path()%22&sq=package:chromium&type=cs
Bug: webrtc:8393
Change-Id: I3cfa3d96b4d0f60e8099e556876bc94340b1bbb5
Reviewed-on: https://webrtc-review.googlesource.com/12540
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@google.com>
Commit-Queue: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20333}
This reverts commit 54d1da13a584680ae80a1f229291e5bb7e76e6e1.
Reason for revert: Breaking tests
Original change's description:
> BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic.
>
> This CL implements the main logic and IOS appRTC integration.
>
> Unit tests and Android appRTC will be in separate CL.
>
> Bug: webrtc:8243
> Change-Id: If8e5195294046a47316e9fade1b0dfec211155e1
> Reviewed-on: https://webrtc-review.googlesource.com/4860
> Commit-Queue: Alex Narest <alexnarest@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20329}
TBR=deadbeef@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,alexnarest@webrtc.org
Change-Id: I5be1da78f360f72be66f9d56dd6b88c1cc13e963
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8243
Reviewed-on: https://webrtc-review.googlesource.com/12560
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20330}
This CL implements the main logic and IOS appRTC integration.
Unit tests and Android appRTC will be in separate CL.
Bug: webrtc:8243
Change-Id: If8e5195294046a47316e9fade1b0dfec211155e1
Reviewed-on: https://webrtc-review.googlesource.com/4860
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20329}
