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}
This is the first CL that tries generating Java -> C++ JNI code.
Bug: webrtc:8278
Change-Id: I1ff0df10db0f80a3c84a61a4e82be8f713690a69
Reviewed-on: https://webrtc-review.googlesource.com/20873
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20592}
The first example CL for generating JNI code
(https://webrtc-review.googlesource.com/c/src/+/4500) seems to stick, so
this CL updates the rest of the VideoEncoder. The JNI code for
Java -> C++ is still done manually.
This CL puts the necessary helper Java methods in a class called
VideoEncoderWrapper.
Bug: webrtc:8278
Change-Id: Ic3a6defe59c094f67ffd8ea86d6c272c676980ae
Reviewed-on: https://webrtc-review.googlesource.com/20871
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20587}
This CL adds a replacement for JNIEnv::FindClass that works from any
thread, i.e. from native C++ threads as well. This function will be used
from the generated JNI code. Long term, we should stop using
classreferenceholder that relies on a hardcoded list of WebRTC classes.
Bug: webrtc:8278
Change-Id: I4f40c744325ac02b73bd8fa479ab50b684429dc2
Reviewed-on: https://webrtc-review.googlesource.com/20223
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20583}
Previously input buffers would be filled incorrectly for sparsely
packed buffers where stride is not equal to the plane width.
Bug: webrtc:8478
Change-Id: I080fa3c354a27982bb996be8c1e41b103384e4bc
Reviewed-on: https://webrtc-review.googlesource.com/17321
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20550}
This is a reland of ba78b5a905bffa05933a135673996df02328f2a4
Original change's description:
> Android: Generate JNI code for VideoSink and VideoEncoder
>
> This is the first CL to start generating JNI code. It has updated two of
> the most recent classes to use JNI code generation.
>
> Bug: webrtc:8278
> Change-Id: I1b19ee78c273346ceeaa0401dbdf8696803f16c7
> Reviewed-on: https://webrtc-review.googlesource.com/3820
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#19994}
Bug: webrtc:8278
Change-Id: Id3e6513736eb87d7c234be3b0d13c5d30435201c
Reviewed-on: https://webrtc-review.googlesource.com/4500
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20548}
Second attempt to land https://webrtc-review.googlesource.com/c/src/+/16180
Now removes voice_engine dependency from peerconnection and fixes a minor
const issue in NullAudioPoller.
TBR=solenberg
Bug: webrtc:7313
Change-Id: Ibfddbdc76118581e4a4dc64575203f84c1659e5c
Reviewed-on: https://webrtc-review.googlesource.com/17784
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20526}
This reverts commit 90bace095806a635411edd40fb8490a144e59e63.
Reason for revert: The original problem of this CL has been fixed in https://webrtc-review.googlesource.com/17540 but sounds like it is also adding voice_engine as a dependency of pc:peerconnection. We should investigate this because probably we can avoid it.
Original change's description:
> Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API
>
> (this CL is based on the work by Taylor and Steve in https://webrtc-review.googlesource.com/c/src/+/10201)
>
> This SetAudioPlayout method lets applications disable audio playout while
> still processing incoming audio data and generating statistics on the
> received audio.
>
> This may be useful if the application wants to set up media flows as
> soon as possible, but isn't ready to play audio yet. Currently, native
> applications don't have any API point to control this, unless they
> completely implement their own AudioDeviceModule.
>
> The SetAudioRecording works in a similar fashion but for the recorded
> audio. One difference is that calling SetAudioRecording(false) does not
> keep any audio processing alive.
>
> TBR=solenberg
>
> Bug: webrtc:7313
> Change-Id: I0aa075f6bfef9818f1080f85a8ff7842fb0750aa
> Reviewed-on: https://webrtc-review.googlesource.com/16180
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20499}
TBR=solenberg@webrtc.org,henrika@webrtc.org,kwiberg@webrtc.org
Change-Id: I8431227e21dbffcfed3dd0e6bd7ce26c4ce09394
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7313
Reviewed-on: https://webrtc-review.googlesource.com/17701
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20512}
(this CL is based on the work by Taylor and Steve in https://webrtc-review.googlesource.com/c/src/+/10201)
This SetAudioPlayout method lets applications disable audio playout while
still processing incoming audio data and generating statistics on the
received audio.
This may be useful if the application wants to set up media flows as
soon as possible, but isn't ready to play audio yet. Currently, native
applications don't have any API point to control this, unless they
completely implement their own AudioDeviceModule.
The SetAudioRecording works in a similar fashion but for the recorded
audio. One difference is that calling SetAudioRecording(false) does not
keep any audio processing alive.
TBR=solenberg
Bug: webrtc:7313
Change-Id: I0aa075f6bfef9818f1080f85a8ff7842fb0750aa
Reviewed-on: https://webrtc-review.googlesource.com/16180
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20499}
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}
Old CreateVideoDecoder interface is deprecated. This allows
VideoDecoderFactoryWrapper to create codecs for types that WebRTC
doesn't know about.
Bug: webrtc:8140
Change-Id: I69aa1a0164642b4e4377daa1abeb9039c04fd884
Reviewed-on: https://webrtc-review.googlesource.com/15401
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20438}
This allows injection of a user-defined post processing module from
the Android layer.
Bug: webrtc:8163
Change-Id: If3a6b4726c34c5f82d186b8cf95373c283cbd3f6
Reviewed-on: https://webrtc-review.googlesource.com/7610
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20367}
AddRef() now returns void, and Release() returns an enum
RefCountReleaseStatus, to indicate whether or not this Release
call implied deletion.
Bug: webrtc:8270
Change-Id: If2fb77f26118b61751b51c856af187c72112c630
Reviewed-on: https://webrtc-review.googlesource.com/3320
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20366}
Previous strict threading requirements we not always followed and this
causes crashes in some scenarios.
Bug: webrtc:7760
Change-Id: I4808edfff19c63a9c4a9585834d46f9c73bb1c4b
Reviewed-on: https://webrtc-review.googlesource.com/13660
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20353}
AndroidNetworkMonitor::BindSocketToNetwork incorrectly checks if the
socket_fd is NETWORK_UNSPECIFIED, when it should instead be checking
if the network_handle is that. This causes the PhysicalSocket to think
that it has succesfully bound to the port, when it really has not.
This stops IceCandidates from being created for these addresses
(although a socket is listening).
The net result of this is that webrtc connections don't work when both
peers are using Wifi-Direct to communicate, although we didn't notice
this previously since we were using Wifi-Direct only on one side.
TBR=pthatcher@webrtc.org
Bug: webrtc:8403
Change-Id: Ibe8f5424654ac9db1f79927ba7ac241fd40d9cba
Reviewed-on: https://webrtc-review.googlesource.com/12442
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20331}
This fixes a crash caused by access to already freed memory returned
by VideoDecoderWrapper::ImplementationName method.
Bug: webrtc:7760
Change-Id: Ia4b020d1dd861e6a45637abde35f12951b7c43ea
Reviewed-on: https://webrtc-review.googlesource.com/9420
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20290}
This patch adds an interface that allows modification of stun messages
sent by TurnPort. A user can inject a TurnCustomizer on the RTCConfig
and the TurnCustomizer will be invoked by TurnPort before sending
message. This allows user to e.g add custom attributes as described
in rtf5389.
BUG=webrtc:8313
Change-Id: I6f4333e9f8ff7fd20f32677be19285f15e1180b6
Reviewed-on: https://webrtc-review.googlesource.com/7618
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20233}
This will handle the scenario where, for example, the initial
offer/answer only negotiates audio, and video is added later (to the
same stream). Previously, there was absolutely no way to get a handle to
the new track without hacking the SDP. Now, the stream will be updated
after setRemoteDescription finishes.
Bug: webrtc:5677
Change-Id: Iea31bb7744da6b82afdaf44c8f74d721298a9474
Reviewed-on: https://webrtc-review.googlesource.com/6261
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20228}
This reverts commit b23ed7f1af467a228cbdc63e839cac8856e9df8d.
Reason for revert: Breaks Chromium FYI build
Sample error log:
../../remoting/test/fake_port_allocator.cc:52:7: error: no matching constructor for initialization of 'cricket::BasicPortAllocator'
: BasicPortAllocator(network_manager, socket_factory),
^ ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
../../third_party/webrtc/p2p/client/basicportallocator.h:32:12: note: candidate constructor not viable: requires single argument 'network_manager', but 2 arguments were provided
explicit BasicPortAllocator(rtc::NetworkManager* network_manager);
^
../../third_party/webrtc/p2p/client/basicportallocator.h:27:7: note: candidate constructor (the implicit copy constructor) not viable: requires 1 argument, but 2 were provided
class BasicPortAllocator : public PortAllocator {
^
../../third_party/webrtc/p2p/client/basicportallocator.h:29:3: note: candidate constructor not viable: requires 3 arguments, but 2 were provided
BasicPortAllocator(rtc::NetworkManager* network_manager,
^
../../third_party/webrtc/p2p/client/basicportallocator.h:33:3: note: candidate constructor not viable: requires 3 arguments, but 2 were provided
BasicPortAllocator(rtc::NetworkManager* network_manager,
^
../../third_party/webrtc/p2p/client/basicportallocator.h:36:3: note: candidate constructor not viable: requires 5 arguments, but 2 were provided
BasicPortAllocator(rtc::NetworkManager* network_manager,
Original change's description:
> TurnCustomizer - an interface for modifying stun messages sent by TurnPort
>
> This patch adds an interface that allows modification of stun messages
> sent by TurnPort. A user can inject a TurnCustomizer on the RTCConfig
> and the TurnCustomizer will be invoked by TurnPort before sending
> message. This allows user to e.g add custom attributes as described
> in rtf5389.
>
> BUG=webrtc:8313
>
> Change-Id: Ibf5cc10af84c57288f1eb4c578ca064611a769f1
> Reviewed-on: https://webrtc-review.googlesource.com/4781
> Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20197}
TBR=deadbeef@webrtc.org,sakal@webrtc.org,jonaso@webrtc.org
Change-Id: I624efb22f6e3ceac1b2ff8af1ec47e4cfdde9140
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8313
Reviewed-on: https://webrtc-review.googlesource.com/7680
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20199}
This patch adds an interface that allows modification of stun messages
sent by TurnPort. A user can inject a TurnCustomizer on the RTCConfig
and the TurnCustomizer will be invoked by TurnPort before sending
message. This allows user to e.g add custom attributes as described
in rtf5389.
BUG=webrtc:8313
Change-Id: Ibf5cc10af84c57288f1eb4c578ca064611a769f1
Reviewed-on: https://webrtc-review.googlesource.com/4781
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20197}
This simplifies the code and ensures we don't starve the decoder if
there are multiple output buffers queued.
Bug: webrtc:7760
Change-Id: I42c31f5045fca96847001260b8796d6756900d0f
Reviewed-on: https://webrtc-review.googlesource.com/5522
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20161}
This renames I420BufferImpl to JavaI420Buffer and moves it as part of
the API.
Bug: webrtc:7749
Change-Id: I70726f248ba4601b4922996712bdfdafbfa4a1e1
Reviewed-on: https://webrtc-review.googlesource.com/5381
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20145}
Get the encoder queue in init instead of the constructor. The
constructor is not always called on the same thread as init. The
encoder may also be reinitialized on a different thread.
Bug: webrtc:7760
Change-Id: I32a025a8bdf652ab019ac4c2ffc6be1533008925
Reviewed-on: https://webrtc-review.googlesource.com/5480
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20074}
This should help users of the library to more easily debug issues.
Bug: None
Change-Id: I85d8101d3b26ccbc34c8beded069461252e61293
Reviewed-on: https://webrtc-review.googlesource.com/4663
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20073}
Do not post releasing the texture output buffer.
Bug: webrtc:7760
Change-Id: Ie4d7165a24c791a406be75688c814e2b9d9cde8f
Reviewed-on: https://webrtc-review.googlesource.com/5440
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20072}
I420Buffer implementations (I420BufferImpl and WrappedNativeI420Buffer) rely on
the 'position' of the underlying ByteBuffers to indicate the start of Y, U, and
V channels. Returning slices prevents callers from altering the state of the
I420Buffer by changing the position.
ByteBuffers are especially prone to accidentally moving the position: relative
read operations (such as get()) increment the position by the size of data read.
BUG=webrtc:8303
Change-Id: I52edce8a3bf46a6c41980ff5110a9480f021f22f
Reviewed-on: https://webrtc-review.googlesource.com/4521
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20050}
Previously VideoFrame.Buffers would be converted to I420 if
apply_rotation() is true. With this change the operation is skipped if
the rotation is 0.
Bug: webrtc:7749
Change-Id: I24a1a8801e41d8f415b33fe57fec953b74df7459
Reviewed-on: https://webrtc-review.googlesource.com/4665
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20038}
All previous initialize methods are deprecated and a new initialize
that uses a builder pattern is added. This gives us full control over
the order of initialization.
Bug: webrtc:7474
Change-Id: I006190e50f2e75c5015f0be75b86d367676db2cc
Reviewed-on: https://webrtc-review.googlesource.com/4160
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20037}
This reverts commit 0cbaf1a6f6ad13a25993f6ea3be931894a196834.
Reason for revert: Makes a test flaky:
https://build.chromium.org/p/client.webrtc/builders/Android32%20%28M%20Nexus5X%29/builds/4603
Original change's description:
> Use injectable hardware video decoder/encoder in AppRTCMobile.
>
> Also include a small fix for getting the encoder queue.
>
> Bug: webrtc:7760
> Change-Id: I96dc8ffb363b90382276d88148f81d5f89dca5f2
> Reviewed-on: https://webrtc-review.googlesource.com/2683
> Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20022}
TBR=magjed@webrtc.org,sakal@webrtc.org
Change-Id: I6cb9a10eadb0eff2b85d5028d684746dc69bccfb
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7760
Reviewed-on: https://webrtc-review.googlesource.com/4480
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20024}
Also include a small fix for getting the encoder queue.
Bug: webrtc:7760
Change-Id: I96dc8ffb363b90382276d88148f81d5f89dca5f2
Reviewed-on: https://webrtc-review.googlesource.com/2683
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20022}
If this is not done, the RTC_DCHECK_CALLED_SEQUENTIALLY might fire
if the encoder is used on a new VideoStreamEncoder. This happens
after VideoSendStream recreations due to changes in the SDP.
BUG=b/66590444
Change-Id: I086370526afbbe2ba629805f97f89e512ba3f472
Reviewed-on: https://webrtc-review.googlesource.com/4360
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20020}
This is a reland of 7a2bfd22e69f14e2af989b9e30ddd834f585caa9
Original change's description:
> Improve unit testing for HardwareVideoEncoder and fix bugs.
>
> Improves the unit testing for HardwareVideoEncoder and fixes bugs in it.
> The main added feature is support for dynamically switching between
> texture and byte buffer modes.
>
> Bug: webrtc:7760
> Change-Id: Iaffe6b7700047c7d0f9a7b89a6118f6ff932cd9b
> Reviewed-on: https://webrtc-review.googlesource.com/2682
> Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#19963}
Bug: webrtc:7760
Change-Id: I605647da456525de8e535cc66cab9d0b3f14240b
Reviewed-on: https://webrtc-review.googlesource.com/3641
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20013}
The video codec factories should be owned by the video engine instead
of by the PeerConnectionFactory.
Bug: None
Change-Id: If63d47cef565138d51377af3fc9ea973950c9390
Reviewed-on: https://webrtc-review.googlesource.com/1601
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20002}
This reverts commit ba78b5a905bffa05933a135673996df02328f2a4.
Reason for revert: Breaks external projects.
Original change's description:
> Android: Generate JNI code for VideoSink and VideoEncoder
>
> This is the first CL to start generating JNI code. It has updated two of
> the most recent classes to use JNI code generation.
>
> Bug: webrtc:8278
> Change-Id: I1b19ee78c273346ceeaa0401dbdf8696803f16c7
> Reviewed-on: https://webrtc-review.googlesource.com/3820
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#19994}
TBR=magjed@webrtc.org,sakal@webrtc.org
Change-Id: I48e079f3ab9661ae4171a3ae5cca571a75d14810
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8278
Reviewed-on: https://webrtc-review.googlesource.com/4100
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19997}
This is the first CL to start generating JNI code. It has updated two of
the most recent classes to use JNI code generation.
Bug: webrtc:8278
Change-Id: I1b19ee78c273346ceeaa0401dbdf8696803f16c7
Reviewed-on: https://webrtc-review.googlesource.com/3820
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19994}
We are doing some unconventional stuff in jni_generator_helper.h in
order to integrate the Chromium script with WebRTC. Long term, we will
improve this and remove the lint suppressions.
Bug: webrtc:8278
Change-Id: I5d6f0017c4deab4586844647f7cd657641fecbab
Reviewed-on: https://webrtc-review.googlesource.com/3780
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19989}
