[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}
19 lines
395 B
Python
19 lines
395 B
Python
include_rules = [
|
|
"+webrtc/api",
|
|
"+webrtc/base",
|
|
"+webrtc/call",
|
|
"+webrtc/logging/rtc_event_log",
|
|
"+webrtc/media",
|
|
"+webrtc/modules/audio_coding",
|
|
"+webrtc/modules/audio_processing",
|
|
"+webrtc/p2p",
|
|
"+webrtc/pc",
|
|
|
|
"+webrtc/modules/rtp_rtcp",
|
|
"+webrtc/system_wrappers",
|
|
|
|
"+webrtc/modules/audio_device",
|
|
"+webrtc/modules/video_coding",
|
|
"+webrtc/modules/video_render",
|
|
]
|