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}
The original rtc_event_log_api is refactored to a pure API target plus
multiple targets coupled with WebRTC implementations.
Bug: None
Change-Id: Iab9eee3f7bf4228c52d94a5f26fc39bb99b5033f
Reviewed-on: https://webrtc-review.googlesource.com/43247
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#21811}
This method is an anti-pattern. Removes usage of the method from
CameraCapturer and deletes it.
Bug: webrtc:8456
Change-Id: I8a70ce968af412fa6e6b9308a9e05d6a8a1ba05d
Reviewed-on: https://webrtc-review.googlesource.com/46140
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21808}
When using VideoFrames, expect_encode_from_texture is true even for
ByteBuffer frames. This causes the encoder to sometimes initialize
itself in surface mode even when egl_context_ is not available.
This leads to a crash.
Bug: webrtc:8776
Change-Id: I8cac36514725b8f430d7bf456d481a4b0c6fcd42
Reviewed-on: https://webrtc-review.googlesource.com/43861
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21789}
In crrev.com/531028, the JNI generator starts to add heap profiler
events to JNI generated functions.
This will cause a ~80KiB regression and at the moment it is breaking
the Chromium Roll into WebRTC.
This CL defines a void macro to re-enable the Chromium Roll avoiding
the size regression.
Bug: chromium:801260
Change-Id: I9543299199c4e14b6b9b235c5cb98c0d53cf29ea
Reviewed-on: https://webrtc-review.googlesource.com/43021
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21730}
The custom template seems to be broken. The necessary flag
"use_unprocessed_jars" has been added to dist_jar template and the
custom template is not needed anymore.
Bug: None
Change-Id: I0ca7a91ee47c8de659bcaffa5661bff55af50375
Reviewed-on: https://webrtc-review.googlesource.com/42680
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21721}
This makes it possible to only inject 1 or 0 video codec factories when
consuming the API using the PeerConnectionFactory+Native header.
Bug: webrtc:7925
Change-Id: I671d8dcdbdf2198a31f3890ff6b416441bd32d48
Reviewed-on: https://webrtc-review.googlesource.com/42661
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21715}
WebRTC is still targeting C++11, this C++14 only feature sneaked in
because the Android toolchain used to build WebRTC on the trybots
uses C++14 features.
Bug: None
Change-Id: I095fb76134dff729c72b7660cdb3d6abc4de2e0c
Reviewed-on: https://webrtc-review.googlesource.com/42501
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21711}
This allows a user to only injecting the decoder or encoder factory.
This behavior also matches how it is implemented for Android.
Bug: webrtc:7925
Change-Id: I3dfca6ea2eaeea437b5b78da2373bd6f7cedc8fa
Reviewed-on: https://webrtc-review.googlesource.com/40860
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21695}
- Move files from voice_engine/ to audio/.
- Rename voice_engine/utility.* to remix_resample.* since there are no other
utilities in those files.
- Move test/mock_voe_channel_proxy.h to audio/.
- Removed voe_channel_id from Audio[Receive|Send]Stream::Config.
- Remove VoiceEngine* from AudioState::Config.
- Fix a few cpplint complaints which showed when moving files.
NOPRESUBMIT=true
Bug: webrtc:4690
Change-Id: Id266c822d956625c358fa5e193e6f4837164aef8
Reviewed-on: https://webrtc-review.googlesource.com/39268
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21657}
This patch exposes the network_thread so that
a custom PortAllocator can use it instead of e.g
creating own thread.
Bug: webrtc:8640
Change-Id: I705629e4f1a4d0a4fed7d53a774ba9564ba076fe
Reviewed-on: https://webrtc-review.googlesource.com/39925
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21643}
Instead of keeping the umbrella header in sync manually and needing
ifdefs to make it include the correct headers depending on platform,
generate it based on the headers we include in the framework target.
Can also be used to only include internal software codec headers when
compiling with support for them.
Bug: webrtc:7925
Change-Id: I63f97af1efc8710cfd62d527fcb343fed05daae2
Reviewed-on: https://webrtc-review.googlesource.com/38702
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@{#21613}
New OwnedPeerConnection takes ownership of the observer. This is done
to allow NativePeerConnectionFactory to return a capsulated object.
Bug: webrtc:8662
Change-Id: Ie876f7b9a1a17ebcfbe51537f712a32ab1a7cbfb
Reviewed-on: https://webrtc-review.googlesource.com/35300
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21610}
This is a reland of 046f78cae6
Original change's description:
> Make freeNativePeerConnectionObserver generic.
>
> Previously, it was only possible to free PeerConnectionObserverJni
> objects using this method. Now it is generic and can free any
> PeerConnectionObserver.
>
> Bug: webrtc:8662
> Change-Id: I619ca5ed88a0c2553fa6d19ce41e510947d5bd44
> Reviewed-on: https://webrtc-review.googlesource.com/35222
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21411}
Bug: webrtc:8662
Change-Id: Iba64d613f7b434260a0d7b762ca67d49b295a84f
Reviewed-on: https://webrtc-review.googlesource.com/38901
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21587}
This removes the DefaultAudioProcessingFactory, PostProcessingFactory and DefaultAudioProcessingFactoryTest classes and leaves the interface AudioProcessingFactory without any default implementation (as the default APM is already created by the PeerConnectionFactory JNI).
Bug: webrtc:8701
Change-Id: I259108afbc5b24cab5161485f45af4236f775c18
Reviewed-on: https://webrtc-review.googlesource.com/37220
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21577}
This reverts commit c73e1f4378.
Reason for revert:
The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660
Original change's description:
> Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""
>
> This reverts commit 588c548657.
>
> Reason for revert:
>
> Breaks Chrome FYI:
>
> /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
> -> returned 1
> ERROR at //build/split_static_library.gni:12:5: Dependency not allowed.
> static_library(target_name) {
> ^----------------------------
> The item //content/renderer:renderer
> can not depend on //third_party/webrtc/media:rtc_internal_video_codecs
> because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [
> //third_party/webrtc/*
> //third_party/webrtc_overrides/*
> ]
>
> https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout
>
> Original change's description:
> > GN rtc_* templates: Set default visibility to webrtc_root + "/*"
> >
> > This means that by default, targets are visible to everything under
> > the WebRTC root, but not visible to anything else.
> >
> > API targets are manually tagged with visibility "*", so that targets
> > outside the WebRTC tree can see them.
> >
> > BUG=webrtc:8254
> >
> > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
> > Reviewed-on: https://webrtc-review.googlesource.com/24140
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21548}
>
> TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
>
> Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8254
> Reviewed-on: https://webrtc-review.googlesource.com/38760
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21555}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org
Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8254
Reviewed-on: https://webrtc-review.googlesource.com/38860
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21558}
On Meizu devices (and maybe on other devices too) when camera is
disallowed in settings `android.hardware.Camera.open` returns
non-null Camera instance, but when any method is invoked on `Camera`
instance the `RuntimeException` is thrown. It claims that Camera
instance is used after `release()` was invoked on that instance.
`Camera1Session.open` didn't handle that case and crashed whole
application when returned Camera instance was already released or
more likely was not even properly initialized during `Camera.open`.
Bug: webrtc:8685
Change-Id: I5cda397a599e87971bf9a4bd3faf6fc4a0d198f3
Reviewed-on: https://webrtc-review.googlesource.com/36300
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21556}
This reverts commit 588c548657.
Reason for revert:
Breaks Chrome FYI:
/b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
-> returned 1
ERROR at //build/split_static_library.gni:12:5: Dependency not allowed.
static_library(target_name) {
^----------------------------
The item //content/renderer:renderer
can not depend on //third_party/webrtc/media:rtc_internal_video_codecs
because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [
//third_party/webrtc/*
//third_party/webrtc_overrides/*
]
https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout
Original change's description:
> GN rtc_* templates: Set default visibility to webrtc_root + "/*"
>
> This means that by default, targets are visible to everything under
> the WebRTC root, but not visible to anything else.
>
> API targets are manually tagged with visibility "*", so that targets
> outside the WebRTC tree can see them.
>
> BUG=webrtc:8254
>
> Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
> Reviewed-on: https://webrtc-review.googlesource.com/24140
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21548}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8254
Reviewed-on: https://webrtc-review.googlesource.com/38760
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21555}
This means that by default, targets are visible to everything under
the WebRTC root, but not visible to anything else.
API targets are manually tagged with visibility "*", so that targets
outside the WebRTC tree can see them.
BUG=webrtc:8254
Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
Reviewed-on: https://webrtc-review.googlesource.com/24140
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21548}
The AudioProcessingBuilder was recently introduced in https://webrtc-review.googlesource.com/c/src/+/34651 to make it easier to create APM instances. This CL replaces all calls to the old Create methods with the new AudioProcessingBuilder.
Bug: webrtc:8668
Change-Id: Ibb5f0fc0dbcc85fcf3355b01bec916f20fe0eb67
Reviewed-on: https://webrtc-review.googlesource.com/36082
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21534}
This creates a new target for pure defines and interfaces. I think
that makes sense (though include/ makes it harder to see when .cc and
.h files should live together).
Bug: webrtc:7620
Change-Id: Ifb0f50faf99166202836c0446feed3443eb52c6e
Reviewed-on: https://webrtc-review.googlesource.com/34657
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21516}
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}
Starting from Chromium Roll [1], WebRTC should start to use NDK r16
for Android builds. The roll cannot be completed because of three
compilation errors:
../../sdk/android/src/jni/pc/androidnetworkmonitor.cc:15:9: error: 'RTLD_NOLOAD' macro redefined [-Werror,-Wmacro-redefined]
^
../../third_party/android_tools/ndk/sysroot/usr/include/dlfcn.h:62:9: note: previous definition is here
../../modules/audio_device/android/audio_record_jni.cc:251:41: error: format specifies type 'long long' but the argument has type 'jlong' (aka 'long') [-Werror,-Wformat]
ALOGD("direct buffer capacity: %lld", capacity);
../../modules/audio_device/android/audio_track_jni.cc:229:41: error: format specifies type 'long long' but the argument has type 'jlong' (aka 'long') [-Werror,-Wformat]
ALOGD("direct buffer capacity: %lld", capacity);
This CL forward fixes these errors in order to fix the Chromium Roll
into WebRTC.
[1] - https://webrtc-review.googlesource.com/c/src/+/37540
Bug: webrtc:8710
Change-Id: I5bc64e73919eee7c9e965a442a386b5e1897b56a
Reviewed-on: https://webrtc-review.googlesource.com/37640
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21510}
Expose RTCDtmfSender API for ObcC SDK via exising RTCRtpSender
to provide ability to use DTMF tones in ObjC apps which uses WebRTC.
Android SDK has already exposed DTMF API via Java's DtmfSender
object, there changes provide similar functionaly to ObjC SDK.
Bug: webrtc:8713
Change-Id: Id68fddbbc362211dc8032fa31b38812d1cff8ed9
Reviewed-on: https://webrtc-review.googlesource.com/35800
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21505}
Exposed setOptions API for iOS SDK via RTCPeerConnectionFactory method
to provide ability to disable encryption and control which network
adapters are ignored.
Only subset of webrtc::PeerConnectionFactoryInterface::Options options
are exposed via iOS SDK, additional options can be exposed as requested.
Android SDK has already exposed setOption API via Java's PeerConnection
constructor, there changes provide similar functionaly to iOS SDK.
Bug: webrtc:8712
Change-Id: Ia2de38cf382afc1bad9bbec6c6eac21ad29aee89
Reviewed-on: https://webrtc-review.googlesource.com/34900
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21504}
