This reverts commit 64534fd917.
Reason for revert: no longer necessary.
Original change's description:
> Make HardwareVideo*Factory ctors that take fallbackToSoftware public.
>
> https://webrtc-review.googlesource.com/c/src/+/17480 changed the default behavior of HardwareVideoEncoderFactory and HardwareVideoDecoderFactory without providing any way to maintain the previous behavior. This breaks current users.
>
> Making these constructors public at least gives existing users a way to maintain the previous behavior.
>
> Bug: webrtc:7925
> Change-Id: Id8f0ec25026592f5e9096ac5f39fdda22993ff09
> Reviewed-on: https://webrtc-review.googlesource.com/22763
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20675}
TBR=magjed@webrtc.org,sakal@webrtc.org,andersc@webrtc.org,mellem@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:7925
Change-Id: Ia467337a3916371a619e1d238e1dd03aa0e19261
Reviewed-on: https://webrtc-review.googlesource.com/23380
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20695}
This CL also merged native_handle_impl.cc and videoframe_jni.cc to keep
all JNI code for the same Java class in the same file, and also renames
this file to jni/videoframe.cc.
The classes AndroidVideoBufferFactory and JavaVideoFrameFactory are
now unnecessary since we cache everything and can be simplified to
global static functions instead.
Bug: webrtc:8278
Change-Id: I03d7b0bbde64cfb407cd6210478ddf9d5599cd8c
Reviewed-on: https://webrtc-review.googlesource.com/22923
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20677}
https://webrtc-review.googlesource.com/c/src/+/17480 changed the default behavior of HardwareVideoEncoderFactory and HardwareVideoDecoderFactory without providing any way to maintain the previous behavior. This breaks current users.
Making these constructors public at least gives existing users a way to maintain the previous behavior.
Bug: webrtc:7925
Change-Id: Id8f0ec25026592f5e9096ac5f39fdda22993ff09
Reviewed-on: https://webrtc-review.googlesource.com/22763
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20675}
In the SoftwareVideoCodecFactories, don't try to create VP9 encoder or
decoder if WebRTC was built without support for it.
Bug: None
Change-Id: I09b87fdcf798c763310af4998dbea8011843010d
Reviewed-on: https://webrtc-review.googlesource.com/22924
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20672}
This fixes clients with custom JNI_OnLoad code. This is basically a
reland of https://webrtc-review.googlesource.com/21281, but called from
LoadGlobalClassReferenceHolder instead of InitGlobalJniVariables. The
problem it solves is that some clients call InitGlobalJniVariables
from a strange thread and without using the rest of our Java API.
TBR=lliuu@@webrtc.org,sakal@webrtc.org
Bug: webrtc:8278
Change-Id: I664e50920bcd7e087ec2d8bb5be13569e8ff3088
Reviewed-on: https://webrtc-review.googlesource.com/22801
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20663}
This reverts commit b3a8c980f7.
Reason for revert: Breaking internal projects
Original change's description:
> Move InitClassLoader from JNI_OnLoad to InitGlobalJniVariables.
>
> This fixes clients with custom JNI_OnLoad code.
>
> Bug: webrtc:8278
> Change-Id: I67b45c2ca07b13b0bf968c5ac17ec6f1bcc58a86
> Reviewed-on: https://webrtc-review.googlesource.com/21281
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20629}
TBR=magjed@webrtc.org,sakal@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:8278
Change-Id: I6add8182172501cb93bed998f68ce848c7280a98
Reviewed-on: https://webrtc-review.googlesource.com/22780
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20660}
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}
The crash occured if removeFrameListener was called after releasing
the EglRenderer.
Bug: b/69040588
Change-Id: I90acc3b280d2009e5f13bb8836a288eb20c7d1d0
Reviewed-on: https://webrtc-review.googlesource.com/21380
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20620}
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}
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}
The new SurfaceEglRenderer helper class extends EglRenderer and
implements rendering on a SurfaceView.
Bug: webrtc:8242
Change-Id: Ic532fe487755d3b54c6bd03f239d714e1ecb10ad
Reviewed-on: https://webrtc-review.googlesource.com/2940
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20562}
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 ba78b5a905
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 90bace0958.
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 affects the new injectable codecs.
Bug: webrtc:8459
Change-Id: I484a3ae4c29fd8bae8b13308315758b3689bdd4d
Reviewed-on: https://webrtc-review.googlesource.com/16861
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20478}
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}
YuvConverter was left package protected by mistake in the previous
change.
Bug: None
Change-Id: I2235f745c2f36f9b49199a3dad09da78f63c33e3
Reviewed-on: https://webrtc-review.googlesource.com/11980
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20322}
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}
In order to enable errorprone [1] we have to fix this finding.
third_party/errorprone has been recently enabled in Chromium and in
order to unblock the Chromium Roll we have to fix these errors.
[1] - https://cs.chromium.org/chromium/src/third_party/errorprone/
Bug: webrtc:8390
Change-Id: Ic737def5ae2a8c6ad1216d9b485af59987fe511c
Tbr: magjed@webrtc.org
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/9161
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20273}
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 b23ed7f1af.
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}
