The type rtc::scoped_refptr<T> is now part of api/. Please include it from
api/scoped_refptr.h.
More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o.
Bug: webrtc:9887, webrtc:8205
No-Try: True
Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf
Reviewed-on: https://webrtc-review.googlesource.com/c/119041
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26414}
Removes the deprecated video codec factories and the related flag and
helper classes.
Bug: webrtc:7925
Change-Id: I0a6d1666ece9ad074fefc79b626ba241765e1b98
Reviewed-on: https://webrtc-review.googlesource.com/c/113940
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26245}
- Remove visibility of encoder target.
- Remove unnecessary dependency on task_queue.
- Remove CreateRtcEventLogFactory() declaration from the rtc_event_log_api target
since the function is not defined in that target.
Bug: None
Change-Id: Id9edee86f358d08ea063d62bd96e9653c5b06d55
Reviewed-on: https://webrtc-review.googlesource.com/c/116060
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26215}
The macOS demo's Info.plist doesn't contains camera and microphone usage description, which will cause demo crash when starting call.
Bug: none
Change-Id: Ie85b0087e6aa6e768a8e6740fffe0b95891b20dd
Reviewed-on: https://webrtc-review.googlesource.com/c/116703
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26188}
These are part of AppRTCMobile and should use framework style imports.
Bug: webrtc:9627
Change-Id: Ieefb12b19edd8e680c69c3508b66bc02545fb49f
Reviewed-on: https://webrtc-review.googlesource.com/c/113920
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25966}
This is a propagation of upstream chromium change needed to
resume DEPS autorolls into WebRTC.
Original comment from upstream change:
> This change is made in preparation for an ErrorProne
> check to catch this at compile time. See bug for details.
Bug: chromium:771683
Change-Id: I56aed15f73a633dcadae7ece6c645cd3596f9257
Reviewed-on: https://webrtc-review.googlesource.com/c/113505
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25951}
Without this, the application can't find the WebRTC dynamic library
when started from the built app bundle (debugging in Xcode worked).
Bug: webrtc:10111
Change-Id: I1610948aae070fe9938e873ce073e05ba7255c7d
Reviewed-on: https://webrtc-review.googlesource.com/c/113805
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25949}
This is a follow-up to
https://webrtc-review.googlesource.com/c/src/+/106280.
This time the whole code base is covered.
Some files may have not been fixed though, whenever the IWYU tool
was breaking the build.
Bug: webrtc:8311
Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef
Reviewed-on: https://webrtc-review.googlesource.com/c/111965
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25830}
Windows UWP allows an application to be built that targets
across all Windows 10 based systems and the Windows store.
Change-Id: I69694bb7e83fb01ad6db2438b065b55738cf01fd
Bug: webrtc:10046
Reviewed-on: https://webrtc-review.googlesource.com/c/110570
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25814}
This test fails on slow runners when TCPChannelClient has not
yet finished communication, but the test thread times out checking
that mock methods are called.
Bug: webrtc:9955
Change-Id: Ia91ada6b01ca1bab48afa57fe76aedd08770a641
Reviewed-on: https://webrtc-review.googlesource.com/c/111383
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25751}
This CL moves webrtc::CreatePeerConnectionFactory definitions out of
pc:create_pc_factory and merges it with its declaration in the api/
directory.
In order to avoid circular dependencies a new build target is created:
* api:create_peerconnection_factory
Bug: webrtc:9862
Change-Id: Ie215c94460cba026f5bf7d11c9a5aa03792064af
Reviewed-on: https://webrtc-review.googlesource.com/c/111186
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25744}
This change makes it possible for ios apps to use the new standards-compliant PeerConnectionState.
Originally reviewed as https://webrtc-review.googlesource.com/c/110502, with an added check to prevent calling unimplemented optional method.
Bug: webrtc:9977
Change-Id: Iebac8ce58d435e38450add51b8915575d0ffd934
Reviewed-on: https://webrtc-review.googlesource.com/c/111084
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25727}
This reverts commit 586725dc9a.
Reason for revert: misses a check to see if the optional callback is implemented.
Original change's description:
> Add ios bindings for PeerConnectionState.
>
> This change makes it possible for ios apps to use the new standards-compliant PeerConnectionState.
>
> Bug: webrtc:9977
> Change-Id: Icf69bb1faa0383ae239cb7508f2a740a2d489697
> Reviewed-on: https://webrtc-review.googlesource.com/c/110502
> Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25651}
TBR=kthelgason@webrtc.org,jonasolsson@webrtc.org
Change-Id: Iff919e9876e6b8dddc6d8ab7df302081d0cfa917
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9977
Reviewed-on: https://webrtc-review.googlesource.com/c/111062
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25659}
This support is needed if there is a big delay between the creation of
frames and the time they are delivered to the WebRTC C++ layer in
AndroidVideoTrackSource. This is the case if e.g. some heavy video
processing is applied to the frames that takes a couple of hundred
milliseconds. Currently, timestamps coming from Android video sources
are aligned to rtc::TimeMicros() once they reach the WebRTC C++ layer in
AndroidVideoTrackSource. At this point, we "forget" any latency that
might occur before this point, and audio/video sync consequently
suffers.
Bug: webrtc:9991
Change-Id: I7b1aaca9a60a978b9195dd5e5eed4779a0055607
Reviewed-on: https://webrtc-review.googlesource.com/c/110783
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25654}
This change makes it possible for android apps to use the new standards-compliant PeerConnectionState.
Bug: webrtc:9977
Change-Id: Iad19c38e664a59e86879715ec7a04a59a9894bee
Reviewed-on: https://webrtc-review.googlesource.com/c/109883
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25652}
This change makes it possible for ios apps to use the new standards-compliant PeerConnectionState.
Bug: webrtc:9977
Change-Id: Icf69bb1faa0383ae239cb7508f2a740a2d489697
Reviewed-on: https://webrtc-review.googlesource.com/c/110502
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25651}
Implicitly retaining self pointer (assuming this is intended behavior) causes compiler warning `-Wimplicit-retain-self`. We should do it explicitly.
Bug: webrtc:9971
Change-Id: If77a67168d8a65ced78d5119b9a7332391d20bc9
Reviewed-on: https://webrtc-review.googlesource.com/c/109641
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25609}
Compared the original CL: https://webrtc-review.googlesource.com/c/src/+/94782
This new CL added backward compatible functions to WebRtcMediaEngineFactory so that internal projects will not be broken.
Because of that, now we can revert all the changes to SDK and PeerConnection and do it in following CLs. This makes this CL cleaner.
One temporary disadvantage of this is the media engine now need to take a dependency onto builtin video bitrate factory, but practically it just moved code around and should not result in a large binary size change. We can remove this dependency later if needed.
Bug: webrtc:9513
Change-Id: I38708762ff365e4ca05974b99fac71edc739a756
Reviewed-on: https://webrtc-review.googlesource.com/c/109040
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25574}
This reverts commit 61c6e5643e.
Reason for revert: downstream projects prepared for this change
Original change's description:
> Revert "Isolating APM API build target: making :api an actual target."
>
> This reverts commit a7f77a7c05.
>
> Reason for revert: breaking downstream
>
> Original change's description:
> > Isolating APM API build target: making :api an actual target.
> >
> > This CL is part of a refactoring work to unblock other CLs
> > that would generate a circular dependency when including
> > modules/audio_processing. It will also allow to easily move
> > the APM interface part under //api.
> >
> > More in detail, this change moves the APM interface files from
> > the build target modules/audio_processing to
> > modules/audio_processing:api. It also adds :api as dependency
> > where needed.
> >
> > Bug: webrtc:9535
> > Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd
> > Reviewed-on: https://webrtc-review.googlesource.com/c/109501
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#25539}
>
> TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org
>
> Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9535
> Reviewed-on: https://webrtc-review.googlesource.com/c/109820
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25540}
TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org
Change-Id: Ic8ed4cc3baf43d639ce13cae256c007728c3ad92
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/c/109884
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25547}
This reverts commit a7f77a7c05.
Reason for revert: breaking downstream
Original change's description:
> Isolating APM API build target: making :api an actual target.
>
> This CL is part of a refactoring work to unblock other CLs
> that would generate a circular dependency when including
> modules/audio_processing. It will also allow to easily move
> the APM interface part under //api.
>
> More in detail, this change moves the APM interface files from
> the build target modules/audio_processing to
> modules/audio_processing:api. It also adds :api as dependency
> where needed.
>
> Bug: webrtc:9535
> Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd
> Reviewed-on: https://webrtc-review.googlesource.com/c/109501
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25539}
TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org
Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/c/109820
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25540}
This CL is part of a refactoring work to unblock other CLs
that would generate a circular dependency when including
modules/audio_processing. It will also allow to easily move
the APM interface part under //api.
More in detail, this change moves the APM interface files from
the build target modules/audio_processing to
modules/audio_processing:api. It also adds :api as dependency
where needed.
Bug: webrtc:9535
Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd
Reviewed-on: https://webrtc-review.googlesource.com/c/109501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25539}
After [1], the Chromium Roll into WebRTC fails with the following error:
FAILED: gen/examples/ \
AppRTCMobile_stubbed_video_io_test_apk__apk_manifest/AndroidManifest.xml
uses-sdk:minSdkVersion 13 cannot be smaller than version 14 declared in
library [...]/android_arch_lifecycle_runtime_java/AndroidManifest.xml
as the library might be using APIs not available in 13
Suggestion: use a compatible library with a minSdk of at most 13,
or increase this project's minSdk version to at least 14,
or use tools:overrideLibrary="android.arch.lifecycle" to force
usage (may lead to runtime failures)
[1] - https://chromium-review.googlesource.com/c/chromium/src/+/1298342
Bug: None
Change-Id: I839dd9dbb346d8f40c25f6a6b93b5d5fc1c26ae9
Reviewed-on: https://webrtc-review.googlesource.com/c/108080
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25381}
This is a reland of ff292f30d9
I'm leaving empty .py files in place in order to not break downstream client builds.
Original change's description:
> Remove deprecated barcode scanning functionality
>
> This code is not used anymore, but it's not possible to land this CL
> until issue webrtc:9665 is fixed.
>
> Bug: webrtc:9642,webrtc:9665
> Change-Id: Idb68e9bdf51b4239788cd6869dcb44dae87d7c56
> Reviewed-on: https://webrtc-review.googlesource.com/c/95951
> Reviewed-by: Paulina Hensman <phensman@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25289}
TBR=phensman@webrtc.org,phoglund@webrtc.org
Bug: webrtc:9642, webrtc:9665
Change-Id: I248f8656b14c89b0b92e777f4408ee6a6dad41f9
Reviewed-on: https://webrtc-review.googlesource.com/c/107360
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25296}
This code is not used anymore, but it's not possible to land this CL
until issue webrtc:9665 is fixed.
Bug: webrtc:9642,webrtc:9665
Change-Id: Idb68e9bdf51b4239788cd6869dcb44dae87d7c56
Reviewed-on: https://webrtc-review.googlesource.com/c/95951
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25289}
This is a reland of 5ccdc1331f
Original change's description:
> Prefix flag macros with WEBRTC_.
>
> Macros defined in rtc_base/flags.h are intended to be used to define
> flags in WebRTC's binaries (e.g. tests).
>
> They are currently not prefixed and this could cause problems with
> downstream clients since these names are quite common.
>
> This CL adds the 'WEBRTC_' prefix to them.
>
> Generated with:
>
> for x in DECLARE DEFINE; do
> for y in bool int float string FLAG; do
> git grep -l "\b$x\_$y\b" | \
> xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
> done
> done
> git cl format
>
> Bug: webrtc:9884
> Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
> Reviewed-on: https://webrtc-review.googlesource.com/c/106682
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25270}
TBR=kwiberg@webrtc.org
Bug: webrtc:9884
Change-Id: I5ba5368a231a334d135ed5e6fd7a279629ced8a3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/107161
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25277}
This reverts commit 5ccdc1331f.
Reason for revert: Breaks downstream project.
Original change's description:
> Prefix flag macros with WEBRTC_.
>
> Macros defined in rtc_base/flags.h are intended to be used to define
> flags in WebRTC's binaries (e.g. tests).
>
> They are currently not prefixed and this could cause problems with
> downstream clients since these names are quite common.
>
> This CL adds the 'WEBRTC_' prefix to them.
>
> Generated with:
>
> for x in DECLARE DEFINE; do
> for y in bool int float string FLAG; do
> git grep -l "\b$x\_$y\b" | \
> xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
> done
> done
> git cl format
>
> Bug: webrtc:9884
> Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
> Reviewed-on: https://webrtc-review.googlesource.com/c/106682
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25270}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
Change-Id: Ia79cd6066ecfd1511c34f1b30fd423e560ed6854
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9884
Reviewed-on: https://webrtc-review.googlesource.com/c/107160
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25276}
Macros defined in rtc_base/flags.h are intended to be used to define
flags in WebRTC's binaries (e.g. tests).
They are currently not prefixed and this could cause problems with
downstream clients since these names are quite common.
This CL adds the 'WEBRTC_' prefix to them.
Generated with:
for x in DECLARE DEFINE; do
for y in bool int float string FLAG; do
git grep -l "\b$x\_$y\b" | \
xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
done
done
git cl format
Bug: webrtc:9884
Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
Reviewed-on: https://webrtc-review.googlesource.com/c/106682
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25270}
This change does three things:
- Move rtc_json into rtc_base/strings/, a non-API directory more fitting to
its purpose.
- Make a target for the currently unused json_unittest.
- Make the code available for use in non-test code again.
Bug: webrtc:9802
Change-Id: Id964a8a4b47b732a962a364894a4dbd3e7f4650f
Reviewed-on: https://webrtc-review.googlesource.com/103126
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24932}
This CL enables tests that were previously disabled and fixes the issues
that made them flaky.
Bug: webrtc:6889, webrtc:7888
Change-Id: I914b59200d7bf2973e8993b04de867cc3355b8a8
Reviewed-on: https://webrtc-review.googlesource.com/98381
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24930}
This features is not needed anymore, with this CL it is also possible
to address two issues:
- The need to pick a default implementation.
- The need to use -Wno-global-constructors.
Bug: webrtc:9631, webrtc:9693
Change-Id: Id3daf34179fbc8db26969fc701ccbfa7182c6a9b
Reviewed-on: https://webrtc-review.googlesource.com/102543
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24904}
In the effort of enabling -Wglobal-constructors and
-Wexit-time-destructors, WebRTC has to remove the Winsock global
initializer.
This will also remove it from Chromium (since it was unused).
After this CL, applications will have to explicitly initialize Winsock
before using WebRTC, this can be done by using the class
rtc::WinsockInitializer provided in rtc_base/win32socketinit.h.
Bug: webrtc:9693, webrtc:9754
Change-Id: I4aae12ff43671ef2713a6fc4592e20759dc6b495
Reviewed-on: https://webrtc-review.googlesource.com/99660
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24903}
This CL removes some deprecated build targets (and their headers)
from system_wrappers:
- field_trial_api
- field_trial_default
- metrics_api
- metrics_default
It also refreshes all the dependencies on field_trial.h and metrics.h.
A nice side effect is that it is finally possible to remove 'nogncheck'
from the following files (when it was used with field_trial_default
and metrics_default):
- sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h
- sdk/android/src/jni/pc/peerconnectionfactory.cc
- sdk/objc/api/peerconnection/RTCFieldTrials.mm
Bug: webrtc:9631
Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/100524
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24878}
This is a reland of d8ff3f29ce.
See https://webrtc-review.googlesource.com/c/src/+/100681/1..4 for
the fix. Error "Failed to open video file for emulated camera" should
be addressed by that change.
Original change's description:
> Compile frame analyzer for the host machine on perf tests.
>
> Bug: webrtc:9665
> Change-Id: I05c01ee4bef0995556b1a679498b3d9132de7c26
> Reviewed-on: https://webrtc-review.googlesource.com/100360
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24756}
TBR=phoglund@webrtc.org, oprypin@webrtc.org
Bug: webrtc:9665
Change-Id: If6a4f2259dabf50718abf47c9cf303d143a1895a
Reviewed-on: https://webrtc-review.googlesource.com/100681
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24762}
Removes redundant field initializers such as null, 0 and false.
Bug: webrtc:9742
Change-Id: I1e54f6c6000885cf95f7af8e2701875a78445497
Reviewed-on: https://webrtc-review.googlesource.com/99481
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24676}
This CL replaces std::o?stringstream with rtc::StringBuilder where that's possible to do without changing any of the surrounding code. It also updates includes and build files as appropriate.
The CL was generated by running 'git grep -l -P std::o?stringstream | xargs perl -pi -e "s/std::o?stringstream/rtc::StringBuilder/g"'. Then I've manually updated the #includes and BUILD files, run 'git cl format' and unstaged any file that would need more complex fixes.
Bug: webrtc:8982
Change-Id: Ibc32153f4a3fd177e260b6ad05ce393972549357
Reviewed-on: https://webrtc-review.googlesource.com/98460
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24605}
This CL separates the files under sdk/objc into logical directories, replacing
the previous file layout under Framework/.
A long term goal is to have some system set up to generate the files under
sdk/objc/api (the PeerConnection API wrappers) from the C++ code. In the shorter
term the goal is to abstract out shared concepts from these classes in order to
make them as uniform as possible.
The separation into base/, components/, and helpers/ are to differentiate between
the base layer's common protocols, various utilities and the actual platform
specific components.
The old directory layout that resembled a framework's internal layout is not
necessary, since it is generated by the framework target when building it.
Bug: webrtc:9627
Change-Id: Ib084fd83f050ae980649ca99e841f4fb0580bd8f
Reviewed-on: https://webrtc-review.googlesource.com/94142
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24493}
This is a reland of d65e143801
The binary for frame_analyzer.cpp is precompiled and stored in the cloud, so it
won't automatically pick up change to the source file. Therefore, restore all
old code to be backwards compatible.
Original change's description:
> Update video_quality_analysis to align videos instead of using barcodes
>
> This CL is a follow-up to the previous CL
> https://webrtc-review.googlesource.com/c/src/+/94773 that added generic
> logic for aligning videos. This will allow us to easily extend
> video_quality_analysis with new sophisticated video quality metrics.
> Also, we can use any kind of video that does not necessarily need to
> contain bar codes. Removing the need to decode barcodes also leads to a
> big speedup for the tests.
>
> Bug: webrtc:9642
> Change-Id: I74b0d630b3e1ed44781ad024115ded3143e28f50
> Reviewed-on: https://webrtc-review.googlesource.com/94845
> Reviewed-by: Paulina Hensman <phensman@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24423}
TBR=phensman@webrtc.org,phoglund@webrtc.org
Bug: webrtc:9642
Change-Id: Id8d129ce103284504c67690f8363c03eaae3eee7
Reviewed-on: https://webrtc-review.googlesource.com/96000
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24429}
This reverts commit d65e143801.
Reason for revert: Breaks perf bots. frame_analyzer is a prebuilt binary, so it won't automatically pick up changes in the .cc file.
Original change's description:
> Update video_quality_analysis to align videos instead of using barcodes
>
> This CL is a follow-up to the previous CL
> https://webrtc-review.googlesource.com/c/src/+/94773 that added generic
> logic for aligning videos. This will allow us to easily extend
> video_quality_analysis with new sophisticated video quality metrics.
> Also, we can use any kind of video that does not necessarily need to
> contain bar codes. Removing the need to decode barcodes also leads to a
> big speedup for the tests.
>
> Bug: webrtc:9642
> Change-Id: I74b0d630b3e1ed44781ad024115ded3143e28f50
> Reviewed-on: https://webrtc-review.googlesource.com/94845
> Reviewed-by: Paulina Hensman <phensman@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24423}
TBR=phoglund@webrtc.org,magjed@webrtc.org,phensman@webrtc.org
Change-Id: Ia590b465687b861fe37ed1b14756d4607ca90da1
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9642
Reviewed-on: https://webrtc-review.googlesource.com/95946
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24428}
This CL is a follow-up to the previous CL
https://webrtc-review.googlesource.com/c/src/+/94773 that added generic
logic for aligning videos. This will allow us to easily extend
video_quality_analysis with new sophisticated video quality metrics.
Also, we can use any kind of video that does not necessarily need to
contain bar codes. Removing the need to decode barcodes also leads to a
big speedup for the tests.
Bug: webrtc:9642
Change-Id: I74b0d630b3e1ed44781ad024115ded3143e28f50
Reviewed-on: https://webrtc-review.googlesource.com/94845
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24423}
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}
This matches Chromium pattern of naming instrumentation test apks with
a name ending in _test_apk. The old naming confuses generate_gradle.py.
Renames:
- AppRTCMobileTest
-> AppRTCMobile_test_apk
- AppRTCMobileTestStubbedVideoIO
-> AppRTCMobile_stubbed_video_io_test_apk
- libjingle_peerconnection_android_unittest
-> android_instrumentation_test_apk
Bug: webrtc:9588
TBR: phoglund
Change-Id: Idb82dc4bd089bc7c90e9373f7c3d572f9fd2d95a
Reviewed-on: https://webrtc-review.googlesource.com/92380
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24184}
This was left by a mistake in a previous refactoring.
R=magjed
Bug: None
Change-Id: Ia2b469e730844780fa3b9ce5540d4bdd4d10b556
Reviewed-on: https://webrtc-review.googlesource.com/91480
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24169}
This should only be landed after clients have been given time to
upgrade to the new interface.
Bug: webrtc:9496, webrtc:9181
Change-Id: Ideb37637d9f0b9a3a9748811879c263c64f81d11
Reviewed-on: https://webrtc-review.googlesource.com/87308
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24080}
Android rules contain `assert is_android`.
This didn't cause any problems only because GN doesn't touch files if they are not referenced from the root BUILD.gn file.
Skipping presubmit because this CL triggers a warning even though it's just adding indentation.
No-Presubmit: True
Bug: None
Change-Id: Ifcb8f0e1d47784ff800507f9d560c68e8f78c717
Reviewed-on: https://webrtc-review.googlesource.com/90040
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24061}
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.
Bug: webrtc:9251, webrtc:163
Change-Id: I9f3c3ce78aa82cbf68f34999b0a7fa9507fe5154
Reviewed-on: https://webrtc-review.googlesource.com/89741
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24059}
Abseil uses -isystem and -imsvc because of some warnings, these two
flags are not used by "gn check", this introduced some regressions.
CL https://chromium-review.googlesource.com/c/chromium/src/+/1124478
will try to switch back absl to -I.
Bug: None
Change-Id: I52e857ef1d11831393c35a1bee09479b83827bad
Reviewed-on: https://webrtc-review.googlesource.com/88121
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23923}
Prepare for building without built-in software codecs. When passing
null, inject the new type of factories but wrap them in the built-in
software codecs outside the videoengine.
Bug: webrtc:7925
Change-Id: I7408e6e46e6b9efdf346852954bf51a97e023b5c
Reviewed-on: https://webrtc-review.googlesource.com/83729
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23897}
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}
This reverts commit 3f1d15b352.
Reason for revert: Removing this breaks a debugging tool that people relied on. I will update that tool to use the new capturer before relanding this.
Original change's description:
> Remove deprecated mac capture code.
>
> Bug: webrtc:6898, webrtc:6333, webrtc:7861
> Change-Id: Ie33eaa47585012f98b59ccffc0c849c1d9da54da
> Reviewed-on: https://webrtc-review.googlesource.com/79920
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23454}
TBR=henrika@webrtc.org,andersc@webrtc.org,kthelgason@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:6898, webrtc:6333, webrtc:7861
Change-Id: Ifc367eecfe92a2b2e4a826a820dc9c3c970ea01e
Reviewed-on: https://webrtc-review.googlesource.com/84380
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23681}
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}
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}
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}
This reverts commit fc4a9c9333.
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}
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 CL removes the use of the @JNINamespace annotation and instead
sets the correct JNI namespace in the build file.
Bug: webrtc:8278
Change-Id: Ia4490399e45a97d56b02c260fd80df4edfa092bf
Reviewed-on: https://webrtc-review.googlesource.com/76440
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23299}
For simplicity, camera with index 0 is used. User also has to manually
give the permission to use the camera for the app.
Bug: webrtc:8769
Change-Id: I371f26f94d629411fd299671b4f3202e84556b80
Reviewed-on: https://webrtc-review.googlesource.com/76982
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23284}
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}
Bigger buttons, fewer taps makes it less tedious to test loopback calls
locally. See webrtc:9240 for details.
Bug: webrtc:9240
Change-Id: I0dfcbc6020f27f284eae25903b2bdc1f272221b6
Reviewed-on: https://webrtc-review.googlesource.com/74583
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23137}
To prepare for making the software codecs optional and injectable, these
codec factories provide a way to pass in identical factories as were the
default old behaviour.
Bug: webrtc:7925
Change-Id: I0c70fa3c56c999e9d1af6e172eff2fbba849e921
Reviewed-on: https://webrtc-review.googlesource.com/71162
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23096}
with call to RuntimeEnvironment.application which provides the same instance.
Bug: None
Change-Id: I4e318955086dff990cb572a09c116d28a1023a34
Reviewed-on: https://webrtc-review.googlesource.com/73244
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23064}
This CL removes internal support for anything else than Android frames
that are wrapped Java VideoFrames. This allows for a big internal
cleanup and we can remove the internal class AndroidTextureBuffer and
all logic related to that. Also, the C++ AndroidVideoTrackSource no
longer needs to hold on to a C++ SurfaceTextureHelper and we can
remove all JNI code related to SurfaceTextureHelper. Also, when these
methods are removed, it's possible to let VideoSource implement the
CapturerObserver interface directly and there is no longer any need for
AndroidVideoTrackSourceObserver. Clients can then initialize
VideoCapturers themselves outside the PeerConnectionFactory, and a new
method is added in the PeerConnectionFactory to allow clients to create
standalone VideoSources that can be connected to a VideoCapturer outside
the factory.
Bug: webrtc:9181
Change-Id: Ie292ea9214f382d44dce9120725c62602a646ed8
Reviewed-on: https://webrtc-review.googlesource.com/71666
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23004}
This CL updates the WebRTC code to stop using the old VideoRenderer and
VideoRenderer.I420Frame classes and instead use the new VideoSink and
VideoFrame classes.
This CL is the first step and the old classes are still left in the code
for now to keep backwards compatibility.
Bug: webrtc:9181
Change-Id: Ib0caa18cbaa2758b7859e850ddcaba003cfb06d6
Reviewed-on: https://webrtc-review.googlesource.com/71662
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22989}
Splits out audio_java into audio_api_java and
java_audio_device_module_java.
Makes depending on java_audio_device_module_jni optional for clients
that do not use it. It is only necessary to depend on this target if
depending on java_audio_device_module_java.
Also some cleanup.
Bug: webrtc:7452
Change-Id: Ic6c4dbe11db3ed8330802a8e90203acb8ef18e72
Reviewed-on: https://webrtc-review.googlesource.com/70220
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22981}
PeerConnectionFactory.initialize() should be the first call before
any other call to the Android WebRTC API. The reason this is important
is mainly because PeerConnectionFactory.initialize() loads the native
C++ code, so all other WebRTC calls that rely on native calls will fail
before this has been done.
Bug: webrtc:7474, webrtc:9153
Change-Id: Id0cb78eaf18ea036f39d616d00ac6e32696266bb
Reviewed-on: https://webrtc-review.googlesource.com/70428
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22954}
Fixes a mismatch between "useHardware" and "disableBuiltIn" when
creating JavaAudioDeviceModule.
Bug: webrtc:7452
Change-Id: Ia5572822dc4514ff9a06811af1bdbb8362a2c71c
Reviewed-on: https://webrtc-review.googlesource.com/69987
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22908}
This CL introduces sdk/android/api/org/webrtc/audio/AudioDeviceModule.java,
which is the new interface for audio device modules on Android.
This CL also refactors the main AudioDeviceModule implementation, which
is sdk/android/api/org/webrtc/audio/JavaAudioDeviceModule.java and makes
it conform to the new interface. The old code used global static methods
to configure the audio device code. This CL gets rid of all that and uses
a builder pattern in JavaAudioDeviceModule instead. The only two dynamic
methods left in the interface are setSpeakerMute() and setMicrophoneMute().
Removing the global static methods allowed a significant cleanup, and e.g.
the file sdk/android/src/jni/audio_device/audio_manager.cc has been
completely removed.
The PeerConnectionFactory interface is also updated to allow passing in
an external AudioDeviceModule. The current built-in ADM is encapsulated
under LegacyAudioDeviceModule.java, which is the default for now to
ensure backwards compatibility.
Bug: webrtc:7452
Change-Id: I64d5f4dba9a004da001f1acb2bd0c1b1f2b64f21
Reviewed-on: https://webrtc-review.googlesource.com/65360
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22765}
This CL splits out the audio device module Java code into a separate
target, and also splits up the audio device module implementations into
three different build targets, one for OpenSLES, AAudio, and the Java
based implementation.
Bug: webrtc:7452, webrtc:9048
Change-Id: I8ec09c73580b468837223ddd420fb29ca61fdea5
Reviewed-on: https://webrtc-review.googlesource.com/66461
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22727}
The class called AudioDeviceModule today is an implementation of a
future interface. We want to reserve the name AudioDeviceModule for
the actual interface. The implementation class has been renamed to
JavaAudioDeviceModule. 'Java' here refers to the fact that the
implementation is using android.media.AudioRecord as input and
android.media.AudioTrack as output, and this is opposed to native
AudioDeviceModule implementations such as OpenSLES and AAudio.
Bug: webrtc:7452
Change-Id: Ifc243c2e169b12a50128ee3252f06d574aa7b358
Reviewed-on: https://webrtc-review.googlesource.com/65400
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22673}
jsr305 is necessary dependency for Nullable annotations.
Also adds a flag to release_aar.py to specify the build directory
manually. This makes it easier to test the script without full
recompilation.
Bug: webrtc:8881
Change-Id: Ib4b8cd4592ced9c92ca2810928bcbb6173d2164e
Reviewed-on: https://webrtc-review.googlesource.com/65081
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22671}
This CL refactors the way RecordedAudioToFileController is connected to
AudioRecord. Instead of allowing to dynamically set and update the
AudioSamplesCallback, it's set once at start time and then stopping is
implemented in RecordedAudioToFileController by simply ignoring calls to
onWebRtcAudioRecordSamplesReady.
The reason for this CL is to reduce the amount of methods we need to
add to the future AudioDeviceModule interface. The more functionality
we can move to creation time in the ctor, the less methods we need to
have in the interface.
Bug: webrtc:7452
Change-Id: I462df275d8579c848e1d2c86cbd8e881da89cbf3
Reviewed-on: https://webrtc-review.googlesource.com/64988
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22653}
To facilitate testing both the old and new AudioDeviceModule path, a
setting is added to AppRTC. Enable "Use legacy audio device" to use
the old path.
Bug: webrtc:7452
Change-Id: I221378ac7bb0fa4e543c3fd081c7a322621621a0
Reviewed-on: https://webrtc-review.googlesource.com/64760
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22609}
This CL contains some follow-up fixes for
https://webrtc-review.googlesource.com/c/src/+/60541. It removes all use
of the old voiceengine implementation from AppRTCMobile.
Bug: webrtc:7452
Change-Id: Iea21a4b3be1f3cbb5062831164fffb2c8051d858
Reviewed-on: https://webrtc-review.googlesource.com/63480
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22530}
This CL adds a stand-alone Android AudioDeviceModule in the
sdk/android folder. It's forked from modules/audio_device/android/
and then simplified for the Android case. The stand-alone Android
ADM is available both in the native_api and also under a field trial
in the Java API.
Bug: webrtc:7452
Change-Id: If6e558026bd0ccb52f56d78ac833339a5789d300
Reviewed-on: https://webrtc-review.googlesource.com/60541
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22517}
Demonstrates how to use the iOS native API to wrap components into
C++ classes.
This CL also introduces a native API wrapper for the capturer.
The C++ code is forked from the corresponding CL for Android at
https://webrtc-review.googlesource.com/c/src/+/60540
Bug: webrtc:8832
Change-Id: I12d9f30e701c0222628e329218f6d5bfca26e6e0
Reviewed-on: https://webrtc-review.googlesource.com/61422
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22484}
Downstreams have been updated, and this now updates all uses of label()
to id() within WebRTC code. This change also makes id() pure virtual and
removes label().
Bug: webrtc:8977
Change-Id: Ib045ea4fabba6f14447c64875c7aeba87dc2be24
Reviewed-on: https://webrtc-review.googlesource.com/60382
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22431}
Add a flag to Android perf tests, so we can specify the number of
retries.
Bug: chromium:755660
Change-Id: Ic498373421b7e0fdf779a4659a0c79d47a59fbde
Reviewed-on: https://webrtc-review.googlesource.com/61103
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22390}
The app is a simple loopback demo demonstrating the usage of Android
native API. This is an initial version and I will add support for
HW codecs etc. in the future.
Bug: webrtc:8769
Change-Id: Ifb6209769dabeb8ca3185b969a1ef8afd6d84390
Reviewed-on: https://webrtc-review.googlesource.com/60540
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22385}
1, Let targets libwebrtc_unity and webrtc_unity_plugin built with Ninja -C out/***.
2, Fixed compile issue of libwebrtc_unity.
3, Built libwebrtc_unity classes into Java 7 instead of Java 8 for android.
4, Added an interface to enable peerconnectionFactory for android in Unity.
Bug: webrtc:8986
Change-Id: I2a206a77ab38895ec9ac845ce89507d61076d396
Reviewed-on: https://webrtc-review.googlesource.com/59000
Reviewed-by: Qiang Chen <qiangchen@chromium.org>
Commit-Queue: George Zhou <gyzhou@chromium.org>
Cr-Commit-Position: refs/heads/master@{#22373}
Add native api conversions for video frames and video renderer. This
also requires some changes to sdk/BUILD to avoid cyclic dependencies.
Bug: webrtc:8832
Change-Id: Ibf21e63bdcae195dcb61d63f9262e6a8dc4fa790
Reviewed-on: https://webrtc-review.googlesource.com/57142
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22340}
This updates AppRTC to use addTrack instead of addStream, and removes
the use of onAddStream, because we no longer have to wait for this to be
fired to set the remote track's video renderers.
Bug: webrtc:8869
Change-Id: I1ecae684a9bc4b30512e8c5d717e72b52c589831
Reviewed-on: https://webrtc-review.googlesource.com/57840
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22318}
The naming convention according to the spec is stream id, not stream
labels.Changing things now to be spec compliant, before it is widely
used. This also includes changes to objective C wrapper code to be in
sync with the change.
Bug: webrtc:7932
Change-Id: I5705e6d8a647aaeed860316466a7320132f24b00
Reviewed-on: https://webrtc-review.googlesource.com/59301
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22316}
This removes the routing for the deprecated audio control setting
Bug: none
Change-Id: If7a134ee487b80a653ba982768ba74ce2d539e0a
Reviewed-on: https://webrtc-review.googlesource.com/58941
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22288}
This removes the routing for the deprecated audio control setting
Change-Id: Id83ff548625279d5b34c9e3cadc097c25a00ef05
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/58900
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22279}
This also changes AppRTC to use addTrack instead of addStream and
"early media" using the RtpTransceiver API.
Bug: webrtc:8870
Change-Id: Ie2848a87c71a95adb785367d822c61e1f753d8c6
Reviewed-on: https://webrtc-review.googlesource.com/56440
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22255}
We want to evaluate the impact on battery consumption from using the
fullscreen HW scaling.
Bug: None
Change-Id: If5becf02c6eaf77f9d0877827db39773ae17fc05
Reviewed-on: https://webrtc-review.googlesource.com/59101
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22243}
Uses new WebRtcAudioRecordSamplesReadyCallback which was added recently in
https://webrtc-review.googlesource.com/c/src/+/49981.
This CL:
- Serves as a test of new WebRtcAudioRecordSamplesReadyCallback.
- Useful for debugging purposes since it records the most native raw audio.
Bug: None
Change-Id: I57375cbf237c171e045b0bdb05f7ae1401930fbc
Reviewed-on: https://webrtc-review.googlesource.com/53120
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22128}
It turns out that some headers were not owned by any targets.
These were:
RTCVideoCodec.h
RTCVideoCodecFactory.h
RTCVideoCodecH264.h
RTCVideoEncoderVP8.h
RTCVideoDecoderVP8.h
RTCVideoEncoderVP9.h
RTCVideoDecoderVP9.h
And some were owned by multiple targets, namely:
RTCPeerConnectionFactory+Native.h
RTCPeerConnectionFactory+Private.h
RTCVideoFrameBuffer.h
These have all been moved to their appropriate homes.
This CL also fixes a lot of cyclic interdependencies in the iOS sdk build files.
Bug: webrtc:8855
Change-Id: I1b7ddb6c2a93868d1510ccf0a64bd3dd169ec3e7
Reviewed-on: https://webrtc-review.googlesource.com/49060
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22052}
We want api/peerconnectioninterface.h (and corresponding build target)
to not depend on call.h, and generally we treat Call as an internal,
non-api, class. But we need CallFactoryInterface in the api in order to
enable use of PeerConnection with or without support for media.
Making CallConfig a top-level class makes it possible to forward declare
it, together with Call, for use in callfactoryinterface.h and
peerconnectioninterface.h.
Delete the peerconnection_and_implicit_call_api target, replaced by
new target callfactory_api, to link between Call and Peerconnection.
Bug: webrtc:7504
Change-Id: I5e3978ef89bcd6705e94536f8676bcf89fc82fe1
Reviewed-on: https://webrtc-review.googlesource.com/46201
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22020}
Enable diagnostic packet and event recording as in the "webrtc-internal"
setting in Chromium.
Bug: webrtc:8859
Change-Id: I1d4a19e0dd60133cdd0d4e18a55780623b65653c
Reviewed-on: https://webrtc-review.googlesource.com/49541
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21987}
cpuMonitor is actually null at the time of the call, but it works
because isSupported doesn't touch 'this' (being a static call).
Bug: None
Change-Id: I177807ee04075d16356878ec72262546d0547aa1
Reviewed-on: https://webrtc-review.googlesource.com/51861
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21983}
When executed on swarming, the script is run from //out/<android build dir>,
so it's better to keep that convention.
Given that all paths are given, cwd doesn't seem to be needed.
TBR=phoglund@webrtc.org
Bug: chromium:755660
Change-Id: Iabf6603983ff88b1281e8113da1aad3320967b72
Reviewed-on: https://webrtc-review.googlesource.com/46142
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21843}
This CL adds #error to spot where rtc_base/win32.h is unconditionally
included and fixes all the places where it happens.
Bug: webrtc:8814
Change-Id: I3c005acf2cdb58a51f1bcaa4acaeebd272c56660
Reviewed-on: https://webrtc-review.googlesource.com/46060
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21840}
This CL fixes an issue where the aecdump file handle gets garbage
collected and closed early in the call.
Bug: webrtc:8822
Change-Id: I959908da164b0ec61ccd976fc52f3d919da11b52
Reviewed-on: https://webrtc-review.googlesource.com/46103
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21839}
This CL adds a GN build flag to include builtin software codecs
(enabled by default).
When setting the flag to false, libvpx can also be excluded. The
benefit is that the resulting binary is smaller.
Replaces https://webrtc-review.googlesource.com/c/src/+/29203
Bug: webrtc:7925
Change-Id: Id330ea8a43169e449ee139eca18e4557cc932e10
Reviewed-on: https://webrtc-review.googlesource.com/36340
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21818}
This is the time to wait after creating the server to ensure it's
listening before trying to connect to it. The previous value of 10 was
not enough; tests occasionally failed.
Bug: webrtc:8711
Change-Id: I67d592fdb9a863d574f2a33096b7050935693f4e
Reviewed-on: https://webrtc-review.googlesource.com/44521
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21793}
Implements the dynamic permission model required by the newer SDK and
changes the theme.
Bug: webrtc:8803
Change-Id: I3ea23a25b27f196fcffd018c7cdd2ff6255b62d9
Reviewed-on: https://webrtc-review.googlesource.com/44400
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21788}
This is a reland of 1175ecd095.
Original change's description:
> Make it possible to run video_quality_loopback_test in swarming.
>
> I made two changes to video_quality_loopback_test to make it possible
> to run it on swarming.
>
> 1. We need to know the path to frame_analyzer when we're generating
> the build files, and it must be already present.
> I made frame_analyzer a resource, so it's downloaded to a known path
> before generating the build files.
> 2. The .zip files for apprtc and golang are downloaded and isolated.
> The script now extracts them and installs AppRTC.
>
> Passing task:
> https://chromium-swarm.appspot.com/task?id=3b230bcc04128210
>
> Bug: chromium:755660
> Change-Id: I34090897402421d5b7e29f21fbed354551197f92
> Reviewed-on: https://webrtc-review.googlesource.com/40920
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21716}
Bug: chromium:755660
Change-Id: Ie3ca62d64b4fe856485287b7d9d3d9e3f75dc091
Reviewed-on: https://webrtc-review.googlesource.com/42860
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21733}
This reverts commit 1175ecd095.
Reason for revert: Breaks the bots.
Original change's description:
> Make it possible to run video_quality_loopback_test in swarming.
>
> I made two changes to video_quality_loopback_test to make it possible
> to run it on swarming.
>
> 1. We need to know the path to frame_analyzer when we're generating
> the build files, and it must be already present.
> I made frame_analyzer a resource, so it's downloaded to a known path
> before generating the build files.
> 2. The .zip files for apprtc and golang are downloaded and isolated.
> The script now extracts them and installs AppRTC.
>
> Passing task:
> https://chromium-swarm.appspot.com/task?id=3b230bcc04128210
>
> Bug: chromium:755660
> Change-Id: I34090897402421d5b7e29f21fbed354551197f92
> Reviewed-on: https://webrtc-review.googlesource.com/40920
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21716}
TBR=phoglund@webrtc.org,ehmaldonado@webrtc.org
Change-Id: Id25d26adc547ff6f9ab178601e37527459c8b5ef
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:755660
Reviewed-on: https://webrtc-review.googlesource.com/42800
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21723}
I made two changes to video_quality_loopback_test to make it possible
to run it on swarming.
1. We need to know the path to frame_analyzer when we're generating
the build files, and it must be already present.
I made frame_analyzer a resource, so it's downloaded to a known path
before generating the build files.
2. The .zip files for apprtc and golang are downloaded and isolated.
The script now extracts them and installs AppRTC.
Passing task:
https://chromium-swarm.appspot.com/task?id=3b230bcc04128210
Bug: chromium:755660
Change-Id: I34090897402421d5b7e29f21fbed354551197f92
Reviewed-on: https://webrtc-review.googlesource.com/40920
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21716}
This is a reland of 727b7d0470
Original change's description:
> Reland "Reland "Put internal video codec factories into separate target""
>
> This is a reland of 0efd1e8b7e
> Original change's description:
> > Reland "Put internal video codec factories into separate target"
> >
> > This is a reland of 51698aefd4
> > Original change's description:
> > > Put internal video codec factories into separate target
> > >
> > > The purpose is to start splitting out the dependencies to the built-in
> > > SW video codecs, so that clients can decide to not depend on them and
> > > get a reduction in binary size.
> > >
> > > Replaces https://webrtc-review.googlesource.com/c/src/+/29101
> > >
> > > Bug: webrtc:7925
> > > Change-Id: I46b95aaf42ead70ba78776de60600b8a66a1fe0c
> > > Reviewed-on: https://webrtc-review.googlesource.com/33420
> > > Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#21381}
> >
> > Bug: webrtc:7925
> > Change-Id: I105287fd41ec3ee5bd964b94efcc9c7b3ecdb842
> > Reviewed-on: https://webrtc-review.googlesource.com/35261
> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21389}
>
> Bug: webrtc:7925
> Change-Id: Id1c7f270676e9e4ca57ca8aa1305cf5554290754
> Reviewed-on: https://webrtc-review.googlesource.com/35501
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21464}
Bug: webrtc:7925
Change-Id: I0b3b5e03d29dadbcbe13cb7ce5369299bb6c0454
Reviewed-on: https://webrtc-review.googlesource.com/37000
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21513}
I updated some dependency enforcement rules to allow examples and pc
to depend on common_video. I reckoned depending on common_video is
not controversial when they already dependend on media/base, which
is a lower-level abstraction.
Bug: webrtc:6828
Change-Id: I77dbeb10187b4e70dda1d873a29994fa76070758
Reviewed-on: https://webrtc-review.googlesource.com/34187
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21495}
The attribute android_manifest for android_library targets has been
removed in [1]. This CL renames it to android_manifest_for_lint (to
avoid lint errors) in all the rtc_android_library targets.
[1] - https://chromium-review.googlesource.com/c/chromium/src/+/848079
Bug: webrtc:8707
Change-Id: Ifa127790937fa49ed52d6aab0c7ce5ab03e1177b
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/37440
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21493}
This reverts commit 727b7d0470.
Reason for revert: Breaks build
Original change's description:
> Reland "Reland "Put internal video codec factories into separate target""
>
> This is a reland of 0efd1e8b7e
> Original change's description:
> > Reland "Put internal video codec factories into separate target"
> >
> > This is a reland of 51698aefd4
> > Original change's description:
> > > Put internal video codec factories into separate target
> > >
> > > The purpose is to start splitting out the dependencies to the built-in
> > > SW video codecs, so that clients can decide to not depend on them and
> > > get a reduction in binary size.
> > >
> > > Replaces https://webrtc-review.googlesource.com/c/src/+/29101
> > >
> > > Bug: webrtc:7925
> > > Change-Id: I46b95aaf42ead70ba78776de60600b8a66a1fe0c
> > > Reviewed-on: https://webrtc-review.googlesource.com/33420
> > > Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#21381}
> >
> > Bug: webrtc:7925
> > Change-Id: I105287fd41ec3ee5bd964b94efcc9c7b3ecdb842
> > Reviewed-on: https://webrtc-review.googlesource.com/35261
> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21389}
>
> Bug: webrtc:7925
> Change-Id: Id1c7f270676e9e4ca57ca8aa1305cf5554290754
> Reviewed-on: https://webrtc-review.googlesource.com/35501
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21464}
TBR=magjed@webrtc.org,andersc@webrtc.org
Change-Id: I8a0621eb91f9ce4835f012e74b6a1da9bf740963
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7925
Reviewed-on: https://webrtc-review.googlesource.com/36940
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21465}
This is a reland of 0efd1e8b7e
Original change's description:
> Reland "Put internal video codec factories into separate target"
>
> This is a reland of 51698aefd4
> Original change's description:
> > Put internal video codec factories into separate target
> >
> > The purpose is to start splitting out the dependencies to the built-in
> > SW video codecs, so that clients can decide to not depend on them and
> > get a reduction in binary size.
> >
> > Replaces https://webrtc-review.googlesource.com/c/src/+/29101
> >
> > Bug: webrtc:7925
> > Change-Id: I46b95aaf42ead70ba78776de60600b8a66a1fe0c
> > Reviewed-on: https://webrtc-review.googlesource.com/33420
> > Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21381}
>
> Bug: webrtc:7925
> Change-Id: I105287fd41ec3ee5bd964b94efcc9c7b3ecdb842
> Reviewed-on: https://webrtc-review.googlesource.com/35261
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21389}
Bug: webrtc:7925
Change-Id: Id1c7f270676e9e4ca57ca8aa1305cf5554290754
Reviewed-on: https://webrtc-review.googlesource.com/35501
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21464}
C++ API allows passing all configuration through RTCConfiguration
object. This adds all values previously passed through PC constraints
to Java RTCConfiguration object and deprecates API that takes PC
contraints.
Using the deprecated API overrides the values in RTCConfigration
object.
Bug: webrtc:8663, webrtc:8662
Change-Id: I128432c3caba74403513fb1347ff58830c643885
Reviewed-on: https://webrtc-review.googlesource.com/33460
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21357}
One reason for the circular deps is that common_types.h is a
historical dumping ground for various structs and defines that
are believed to be generally useful. I tried moving things out
that did not appear to be used downstream (StreamCounters,
RtpCounters etc) and moved the things that seemed used
(RtpHeader + supporting structs) to a new file api/rtp_headers.h.
This makes their place in the api more clear while moving out
the things that don't belong in the API in the first place.
I had to extract out typedefs.h from webrtc_common to resolve
another circular dependency. I believe checks includes typedefs,
but common depends on checks.
Bug: webrtc:7745
Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b
Reviewed-on: https://webrtc-review.googlesource.com/33001
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21295}
This splits things out of rtc_base and makes dependencies explicit.
Bug: webrtc:6828
Change-Id: Id521896c3c43595349021c857bec216e429a0c8d
Reviewed-on: https://webrtc-review.googlesource.com/32780
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21264}
This splits things out of rtc_base and makes dependencies explicit.
Bug: webrtc:6828
Change-Id: Ib813c7bd9e4de7ab015acb917bc09ee7204ba7bd
Reviewed-on: https://webrtc-review.googlesource.com/31940
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21245}
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 30915a742d
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}
Second attempt to land https://webrtc-review.googlesource.com/c/src/+/15481.
This time with an extra (dummy) interface to ensure that we don't
break downstream clients.
Improves native Android audio implementations.
Bug: webrtc:8453
Change-Id: I659a3013ae523a2588e4c41ca44b7d0d2d65efb7
Reviewed-on: https://webrtc-review.googlesource.com/16425
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20462}
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 30915a742d.
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}
