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webrtc_m130/webrtc/ortc/DEPS

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Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. This CL adds the following interfaces: * RtpTransportController * RtpTransport * RtpSender * RtpReceiver They're implemented on top of the "BaseChannel" object, which is normally used in a PeerConnection, and roughly corresponds to an SDP "m=" section. As a result of this, there are several limitations: * You can only have one of each type of sender and receiver (audio/video) on top of the same transport controller. * The sender/receiver with the same media type must use the same RTP transport. * You can't change the transport after creating the sender or receiver. * Some of the parameters aren't supported. Later, these "adapter" objects will be gradually replaced by real objects that don't have these limitations, as "BaseChannel", "MediaChannel" and related code is restructured. In this CL, we essentially have: ORTC adapter objects -> BaseChannel -> Media engine PeerConnection -> BaseChannel -> Media engine And later we hope to have simply: PeerConnection -> "Real" ORTC objects -> Media engine See the linked bug for more context. BUG=webrtc:7013 TBR=stefan@webrtc.org Review-Url: https://codereview.webrtc.org/2675173003 Cr-Commit-Position: refs/heads/master@{#16842}
2017-02-25 18:15:09 -08:00
include_rules = [
"+webrtc/api",
"+webrtc/base",
"+webrtc/call",
"+webrtc/logging/rtc_event_log",
"+webrtc/media",
"+webrtc/modules/audio_coding",
Allow an external audio processing module to be used in WebRTC [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}
2017-06-29 08:32:09 -07:00
"+webrtc/modules/audio_processing",
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. This CL adds the following interfaces: * RtpTransportController * RtpTransport * RtpSender * RtpReceiver They're implemented on top of the "BaseChannel" object, which is normally used in a PeerConnection, and roughly corresponds to an SDP "m=" section. As a result of this, there are several limitations: * You can only have one of each type of sender and receiver (audio/video) on top of the same transport controller. * The sender/receiver with the same media type must use the same RTP transport. * You can't change the transport after creating the sender or receiver. * Some of the parameters aren't supported. Later, these "adapter" objects will be gradually replaced by real objects that don't have these limitations, as "BaseChannel", "MediaChannel" and related code is restructured. In this CL, we essentially have: ORTC adapter objects -> BaseChannel -> Media engine PeerConnection -> BaseChannel -> Media engine And later we hope to have simply: PeerConnection -> "Real" ORTC objects -> Media engine See the linked bug for more context. BUG=webrtc:7013 TBR=stefan@webrtc.org Review-Url: https://codereview.webrtc.org/2675173003 Cr-Commit-Position: refs/heads/master@{#16842}
2017-02-25 18:15:09 -08:00
"+webrtc/p2p",
"+webrtc/pc",
"+webrtc/modules/rtp_rtcp",
"+webrtc/system_wrappers",
"+webrtc/modules/audio_device",
"+webrtc/modules/video_coding",
"+webrtc/modules/video_render",
]
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