mediasoup
Version:
Cutting Edge WebRTC Video Conferencing
25 lines • 1.89 kB
TypeScript
import * as flatbuffers from 'flatbuffers';
import { AudioLevelObserverOptions, AudioLevelObserverOptionsT } from '../../fbs/audio-level-observer/audio-level-observer-options';
export declare class CreateAudioLevelObserverRequest implements flatbuffers.IUnpackableObject<CreateAudioLevelObserverRequestT> {
bb: flatbuffers.ByteBuffer | null;
bb_pos: number;
__init(i: number, bb: flatbuffers.ByteBuffer): CreateAudioLevelObserverRequest;
static getRootAsCreateAudioLevelObserverRequest(bb: flatbuffers.ByteBuffer, obj?: CreateAudioLevelObserverRequest): CreateAudioLevelObserverRequest;
static getSizePrefixedRootAsCreateAudioLevelObserverRequest(bb: flatbuffers.ByteBuffer, obj?: CreateAudioLevelObserverRequest): CreateAudioLevelObserverRequest;
rtpObserverId(): string | null;
rtpObserverId(optionalEncoding: flatbuffers.Encoding): string | Uint8Array | null;
options(obj?: AudioLevelObserverOptions): AudioLevelObserverOptions | null;
static startCreateAudioLevelObserverRequest(builder: flatbuffers.Builder): void;
static addRtpObserverId(builder: flatbuffers.Builder, rtpObserverIdOffset: flatbuffers.Offset): void;
static addOptions(builder: flatbuffers.Builder, optionsOffset: flatbuffers.Offset): void;
static endCreateAudioLevelObserverRequest(builder: flatbuffers.Builder): flatbuffers.Offset;
unpack(): CreateAudioLevelObserverRequestT;
unpackTo(_o: CreateAudioLevelObserverRequestT): void;
}
export declare class CreateAudioLevelObserverRequestT implements flatbuffers.IGeneratedObject {
rtpObserverId: string | Uint8Array | null;
options: AudioLevelObserverOptionsT | null;
constructor(rtpObserverId?: string | Uint8Array | null, options?: AudioLevelObserverOptionsT | null);
pack(builder: flatbuffers.Builder): flatbuffers.Offset;
}
//# sourceMappingURL=create-audio-level-observer-request.d.ts.map