This CL removes code that supported the now removed
downstream dependencies in the support for using an
external audio processing module.
BUG=webrtc:7939
Review-Url: https://codereview.webrtc.org/2969213002
Cr-Commit-Position: refs/heads/master@{#18929}
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`
BUG=webrtc:7634
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
Review-Url: https://codereview.webrtc.org/2969623003
Cr-Commit-Position: refs/heads/master@{#18870}
[This CL is a rebase of an original CL by solenberg@:
https://codereview.webrtc.org/2948763002/ which in turn was a
rebase of an original CL by peah@:
https://chromium-review.googlesource.com/c/527032/]
Allow an external audio processing module to be used in WebRTC
This CL adds support for optionally using an externally created audio
processing module in a peerconnection. The ownership is shared
between the peerconnection and the external creator of the module.
As part of this the internal ownership of the audio processing module
is moved from VoiceEngine to WebRtcVoiceEngine.
BUG=webrtc:7775
Review-Url: https://codereview.webrtc.org/2961723004
Cr-Commit-Position: refs/heads/master@{#18837}
The "initialize audio/video" parameters are no longer needed, but
at the same time were required to be true, causing a lot of confusion.
This CL removes them, but leaves the old method signature around,
marked "deprecated".
BUG=webrtc:3416
TBR=solenberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2800353002
Cr-Commit-Position: refs/heads/master@{#17626}
- The legacy API is not used in WVoE/MC.
- Removed use of the API (along with StartReceive()) from unit tests.
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/2453243003
Cr-Commit-Position: refs/heads/master@{#14858}
This file defines webrtc::Config which was mostly used by modules/audio_processing. The files webrtc/common.h, webrtc/common.cc and webrtc/test/common_unittests.cc are moved to modules/audio_processing and the few remaining uses of webrtc::Config are replaced with simpler code.
- For NetEq and pacing configuration, a VoEBase::ChannelConfig is passed to VoEBase::CreateChannel().
- Removes the need for VoiceEngine::Create(const Config& config). No need to store the webrtc::Config in VoE shared state.
BUG=webrtc:5879
Review-Url: https://codereview.webrtc.org/2307533004
Cr-Commit-Position: refs/heads/master@{#14109}
VoEBase is plumbed to optionally take an AudioDecoderFactory, or create
a builtin factory if none is provided.
Retained the CreateChannel interfaces in Channel and ChannelManager
and added variants for injecting an AudioDecoderFactory. The
"old-style" variants call CreateBuiltinAudioDecoderFactory to get a
factory to use.
(Just realized this means each channel uses a separate factory with the
old-style calls. Probably ok.)
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/1993783002
Cr-Commit-Position: refs/heads/master@{#12961}
- Remove WVoE::SetAudioDeviceModule() - the ADM is now supplied in ctor.
- Remove WVoE::Init() and WVoE::Terminate().
- Remove MediaEngineInterface::Terminate().
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1830213002
Cr-Commit-Position: refs/heads/master@{#12173}
BUG=4690
I have removed methods in VoE interfaces that were marked to be removed. I have removed them also in fake and mock implementations. I have also updated the callers in various ways:
1. Project win_test had some calls to the removed methods, but it turned out that the project is not used anymore, so I removed it entirely.
2. There were some calls to removed methods in jni methods. I have removed couple of jni methods as now they seem to do nothing.
3. With the remaining callers I just removed the calls to removed methods.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/53519004
Cr-Commit-Position: refs/heads/master@{#9281}
This CL adds support for unittests of the AudioDeviceModule on Android using both Java and C++. The new framework uses ::testing::TesWithParam to support both Java-based audio and OpenSL ES based audio. However, given existing issues in our OpenSL ES implementation, the list of test parameters only contains Java in this first version. Open SL ES will be enabled as soon as the backend has been refactored.
It also:
- Removes the redundant JNIEnv* argument in webrtc::VoiceEngine::SetAndroidObjects().
- Modifies usage of enable_android_opensl and the WEBRTC_ANDROID_OPENSLES define.
- Adds kAndroidJavaAudio and kAndroidOpenSLESAudio to AudioLayer enumerator.
- Fixes some bugs which were discovered when running the tests.
BUG=NONE
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40069004
Cr-Commit-Position: refs/heads/master@{#8651}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8651 4adac7df-926f-26a2-2b94-8c16560cd09d
The complexity of the last ChannelManager and potentially usage of it as well caused race conditions and deadlocks in loopback voe_auto_test. This ref-counted solution takes no long-term locks, uses less locks overall and is significantly easier to understand.
ScopedChannel has been split up into a ChannelOwner with a reference to a channel and an Iterator over ChannelManager. Previous code was really used for both things. ChannelOwner is used as a shared pointer to a channel object, while an Iterator should work as expected.
BUG=2081
R=tommi@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1802004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4502 4adac7df-926f-26a2-2b94-8c16560cd09d
This associates the two types instead of incorrectly reinterpret casting
VoiceEngineImpl* to VoiceEngine* (since these types were previously unrelated).
Please see more details in the bug for how this is currently causing problems
with security tools.
BUG=38612
Review URL: https://webrtc-codereview.appspot.com/1099013
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3520 4adac7df-926f-26a2-2b94-8c16560cd09d
