update to pion v3
This commit is contained in:
parent
00a785f4c5
commit
a362df4976
@ -2,7 +2,7 @@ import EventEmitter from 'eventemitter3'
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import { OPCODE } from './data'
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import { EVENT, WebSocketEvents } from './events'
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import { WebSocketMessages, WebSocketPayloads, SignalProvidePayload } from './messages'
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import { WebSocketMessages, WebSocketPayloads, SignalProvidePayload, SignalCandidatePayload } from './messages'
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export interface BaseEvents {
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info: (...message: any[]) => void
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@ -211,8 +211,8 @@ export abstract class BaseClient extends EventEmitter<BaseEvents> {
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}
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this._peer.ontrack = this.onTrack.bind(this)
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this._peer.addTransceiver('audio', { direction: 'recvonly' })
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this._peer.addTransceiver('video', { direction: 'recvonly' })
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this._peer.addTransceiver('audio', { direction: 'sendrecv' })
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this._peer.addTransceiver('video', { direction: 'sendrecv' })
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this._channel = this._peer.createDataChannel('data')
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this._channel.onerror = this.onError.bind(this)
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@ -246,6 +246,15 @@ export abstract class BaseClient extends EventEmitter<BaseEvents> {
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this.createPeer(sdp, lite, ice)
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return
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}
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if (event === EVENT.SIGNAL.CANDIDATE) {
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const { data } = payload as SignalCandidatePayload
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let candidate: RTCIceCandidate = JSON.parse(data)
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this._peer!.addIceCandidate(candidate)
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return
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}
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// @ts-ignore
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if (typeof this[event] === 'function') {
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@ -14,6 +14,7 @@ export const EVENT = {
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SIGNAL: {
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ANSWER: 'signal/answer',
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PROVIDE: 'signal/provide',
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CANDIDATE: 'signal/candidate'
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},
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MEMBER: {
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LIST: 'member/list',
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@ -78,7 +79,7 @@ export type ControlEvents =
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export type SystemEvents = typeof EVENT.SYSTEM.DISCONNECT
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export type MemberEvents = typeof EVENT.MEMBER.LIST | typeof EVENT.MEMBER.CONNECTED | typeof EVENT.MEMBER.DISCONNECTED
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export type SignalEvents = typeof EVENT.SIGNAL.ANSWER | typeof EVENT.SIGNAL.PROVIDE
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export type SignalEvents = typeof EVENT.SIGNAL.ANSWER | typeof EVENT.SIGNAL.PROVIDE | typeof EVENT.SIGNAL.CANDIDATE
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export type ChatEvents = typeof EVENT.CHAT.MESSAGE | typeof EVENT.CHAT.EMOTE
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export type ScreenEvents = typeof EVENT.SCREEN.CONFIGURATIONS | typeof EVENT.SCREEN.RESOLUTION | typeof EVENT.SCREEN.SET
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@ -15,6 +15,7 @@ export type WebSocketMessages =
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| WebSocketMessage
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| SignalProvideMessage
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| SignalAnswerMessage
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| SignalCandidateMessage
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| MemberListMessage
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| MemberConnectMessage
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| MemberDisconnectMessage
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@ -26,6 +27,7 @@ export type WebSocketMessages =
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export type WebSocketPayloads =
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| SignalProvidePayload
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| SignalAnswerPayload
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| SignalCandidatePayload
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| MemberListPayload
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| Member
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| ControlPayload
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@ -78,6 +80,14 @@ export interface SignalAnswerPayload {
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displayname: string
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}
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// signal/candidate
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export interface SignalCandidateMessage extends WebSocketMessage, SignalCandidatePayload {
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event: typeof EVENT.SIGNAL.CANDIDATE
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}
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export interface SignalCandidatePayload {
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data: string
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}
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/*
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MEMBER MESSAGES/PAYLOADS
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*/
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@ -10,10 +10,9 @@ import "C"
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import (
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"fmt"
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"sync"
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"time"
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"unsafe"
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"github.com/pion/webrtc/v2"
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"n.eko.moe/neko/internal/types"
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)
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@ -80,13 +79,13 @@ func CreateRTMPPipeline(pipelineDevice string, pipelineDisplay string, pipelineS
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}
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// CreateAppPipeline creates a GStreamer Pipeline
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func CreateAppPipeline(codecName string, pipelineDevice string, pipelineSrc string) (*Pipeline, error) {
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func CreateAppPipeline(codecName string, pipelineDevice string, pipelineSrc string, bitrate string) (*Pipeline, error) {
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pipelineStr := " ! appsink name=appsink"
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var clockRate float32
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switch codecName {
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case webrtc.VP8:
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case "VP8":
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// https://gstreamer.freedesktop.org/documentation/vpx/vp8enc.html?gi-language=c
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// gstreamer1.0-plugins-good
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// vp8enc error-resilient=partitions keyframe-max-dist=10 auto-alt-ref=true cpu-used=5 deadline=1
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@ -99,9 +98,9 @@ func CreateAppPipeline(codecName string, pipelineDevice string, pipelineSrc stri
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if pipelineSrc != "" {
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pipelineStr = fmt.Sprintf(pipelineSrc+pipelineStr, pipelineDevice)
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} else {
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pipelineStr = fmt.Sprintf(videoSrc+"vp8enc cpu-used=8 threads=2 deadline=1 error-resilient=partitions keyframe-max-dist=10 auto-alt-ref=true"+pipelineStr, pipelineDevice)
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pipelineStr = fmt.Sprintf(videoSrc+"vp8enc cpu-used=-5 threads=4 deadline=1 error-resilient=partitions keyframe-max-dist=30 auto-alt-ref=true"+pipelineStr, pipelineDevice)
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}
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case webrtc.VP9:
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case "VP9":
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// https://gstreamer.freedesktop.org/documentation/vpx/vp9enc.html?gi-language=c
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// gstreamer1.0-plugins-good
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// vp9enc
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@ -117,7 +116,7 @@ func CreateAppPipeline(codecName string, pipelineDevice string, pipelineSrc stri
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} else {
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pipelineStr = fmt.Sprintf(videoSrc+"vp9enc"+pipelineStr, pipelineDevice)
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}
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case webrtc.H264:
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case "H264":
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// https://gstreamer.freedesktop.org/documentation/openh264/openh264enc.html?gi-language=c#openh264enc
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// gstreamer1.0-plugins-bad
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// openh264enc multi-thread=4 complexity=high bitrate=3072000 max-bitrate=4096000
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@ -132,12 +131,19 @@ func CreateAppPipeline(codecName string, pipelineDevice string, pipelineSrc stri
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} else {
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var h264Str string
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h264Str = "openh264enc multi-thread=4 complexity=high bitrate=3072000 max-bitrate=4096000 ! video/x-h264,stream-format=byte-stream"
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if bitrate != "" {
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h264Str = "openh264enc multi-thread=4 complexity=high bitrate=" + bitrate + "000 max-bitrate=" + bitrate + "999 ! video/x-h264,stream-format=byte-stream"
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}
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// https://gstreamer.freedesktop.org/documentation/x264/index.html?gi-language=c
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// gstreamer1.0-plugins-ugly
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// video/x-raw,format=I420 ! x264enc bframes=0 key-int-max=60 byte-stream=true tune=zerolatency speed-preset=veryfast ! video/x-h264,stream-format=byte-stream
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if err := CheckPlugins([]string{"openh264"}); err != nil {
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h264Str = "video/x-raw,format=I420 ! x264enc bframes=0 key-int-max=60 byte-stream=true tune=zerolatency speed-preset=veryfast ! video/x-h264,stream-format=byte-stream"
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h264Str = "video/x-raw,format=I420 ! x264enc threads=4 byte-stream=true tune=zerolatency speed-preset=veryfast ! video/x-h264,stream-format=byte-stream"
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if bitrate != "" {
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h264Str = "video/x-raw,format=I420 ! x264enc threads=4 bitrate=" + bitrate + " byte-stream=true tune=zerolatency speed-preset=veryfast ! video/x-h264,stream-format=byte-stream"
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}
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if err := CheckPlugins([]string{"x264"}); err != nil {
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return nil, err
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@ -145,7 +151,7 @@ func CreateAppPipeline(codecName string, pipelineDevice string, pipelineSrc stri
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}
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pipelineStr = fmt.Sprintf(videoSrc+h264Str+pipelineStr, pipelineDevice)
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}
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case webrtc.Opus:
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case "Opus":
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// https://gstreamer.freedesktop.org/documentation/opus/opusenc.html
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// gstreamer1.0-plugins-base
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// opusenc
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@ -160,7 +166,7 @@ func CreateAppPipeline(codecName string, pipelineDevice string, pipelineSrc stri
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} else {
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pipelineStr = fmt.Sprintf(audioSrc+"opusenc"+pipelineStr, pipelineDevice)
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}
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case webrtc.G722:
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case "G722":
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// https://gstreamer.freedesktop.org/documentation/libav/avenc_g722.html?gi-language=c
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// gstreamer1.0-libav
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// avenc_g722
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@ -175,7 +181,7 @@ func CreateAppPipeline(codecName string, pipelineDevice string, pipelineSrc stri
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} else {
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pipelineStr = fmt.Sprintf(audioSrc+"avenc_g722"+pipelineStr, pipelineDevice)
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}
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case webrtc.PCMU:
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case "PCMU":
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// https://gstreamer.freedesktop.org/documentation/mulaw/mulawenc.html?gi-language=c
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// gstreamer1.0-plugins-good
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// audio/x-raw, rate=8000 ! mulawenc
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@ -190,7 +196,7 @@ func CreateAppPipeline(codecName string, pipelineDevice string, pipelineSrc stri
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} else {
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pipelineStr = fmt.Sprintf(audioSrc+"audio/x-raw, rate=8000 ! mulawenc"+pipelineStr, pipelineDevice)
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}
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case webrtc.PCMA:
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case "PCMA":
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// https://gstreamer.freedesktop.org/documentation/alaw/alawenc.html?gi-language=c
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// gstreamer1.0-plugins-good
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// audio/x-raw, rate=8000 ! alawenc
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@ -270,8 +276,7 @@ func goHandlePipelineBuffer(buffer unsafe.Pointer, bufferLen C.int, duration C.i
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pipelinesLock.Unlock()
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if ok {
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samples := uint32(pipeline.ClockRate * (float32(duration) / 1000000000))
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pipeline.Sample <- types.Sample{Data: C.GoBytes(buffer, bufferLen), Samples: samples}
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pipeline.Sample <- types.Sample{Data: C.GoBytes(buffer, bufferLen), Timestamp: time.Now(), Duration: time.Duration(duration)}
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} else {
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fmt.Printf("discarding buffer, no pipeline with id %d", int(pipelineID))
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}
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@ -135,6 +135,7 @@ func (manager *RemoteManager) createPipelines() {
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manager.config.VideoCodec,
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manager.config.Display,
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manager.config.VideoParams,
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manager.config.Bitrate,
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)
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if err != nil {
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manager.logger.Panic().Err(err).Msg("unable to create video pipeline")
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@ -144,6 +145,7 @@ func (manager *RemoteManager) createPipelines() {
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manager.config.AudioCodec,
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manager.config.Device,
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manager.config.AudioParams,
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"",
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)
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if err != nil {
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manager.logger.Panic().Err(err).Msg("unable to create audio pipeline")
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@ -174,6 +176,7 @@ func (manager *RemoteManager) ChangeResolution(width int, height int, rate int)
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manager.config.VideoCodec,
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manager.config.Display,
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manager.config.VideoParams,
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manager.config.Bitrate,
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)
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if err != nil {
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manager.logger.Panic().Err(err).Msg("unable to create new video pipeline")
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@ -120,6 +120,16 @@ func (session *Session) SignalAnswer(sdp string) error {
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return session.peer.SignalAnswer(sdp)
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}
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func (session *Session) SignalCandidate(data string) error {
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if session.socket == nil {
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return nil
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}
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return session.socket.Send(&message.SignalCandidate{
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Event: event.SIGNAL_CANDIDATE,
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Data: data,
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});
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}
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func (session *Session) destroy() error {
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if session.socket != nil {
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if err := session.socket.Destroy(); err != nil {
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@ -4,7 +4,6 @@ import (
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"regexp"
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"strconv"
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"github.com/pion/webrtc/v2"
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"github.com/spf13/cobra"
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"github.com/spf13/viper"
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)
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@ -19,6 +18,7 @@ type Remote struct {
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ScreenWidth int
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ScreenHeight int
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ScreenRate int
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Bitrate string
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}
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func (Remote) Init(cmd *cobra.Command) error {
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@ -47,6 +47,12 @@ func (Remote) Init(cmd *cobra.Command) error {
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return err
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}
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cmd.PersistentFlags().String("bitrate", "", "set this video bitrate when possible")
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if err := viper.BindPFlag("bitrate", cmd.PersistentFlags().Lookup("bitrate")); err != nil {
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return err
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}
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// video codecs
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cmd.PersistentFlags().Bool("vp8", false, "use VP8 video codec")
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if err := viper.BindPFlag("vp8", cmd.PersistentFlags().Lookup("vp8")); err != nil {
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@ -88,24 +94,24 @@ func (Remote) Init(cmd *cobra.Command) error {
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}
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func (s *Remote) Set() {
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videoCodec := webrtc.VP8
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videoCodec := "VP8"
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if viper.GetBool("vp8") {
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videoCodec = webrtc.VP8
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videoCodec = "VP8"
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} else if viper.GetBool("vp9") {
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videoCodec = webrtc.VP9
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videoCodec = "VP9"
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} else if viper.GetBool("h264") {
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videoCodec = webrtc.H264
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videoCodec = "H264"
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}
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audioCodec := webrtc.Opus
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audioCodec := "Opus"
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if viper.GetBool("opus") {
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audioCodec = webrtc.Opus
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audioCodec = "Opus"
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} else if viper.GetBool("g722") {
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audioCodec = webrtc.G722
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audioCodec = "G722"
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} else if viper.GetBool("pcmu") {
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audioCodec = webrtc.PCMU
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audioCodec = "PCMU"
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} else if viper.GetBool("pcma") {
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audioCodec = webrtc.PCMA
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audioCodec = "PCMA"
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}
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s.Device = viper.GetString("device")
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@ -114,6 +120,7 @@ func (s *Remote) Set() {
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s.Display = viper.GetString("display")
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s.VideoCodec = videoCodec
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s.VideoParams = viper.GetString("video")
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s.Bitrate = viper.GetString("bitrate")
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s.ScreenWidth = 1280
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s.ScreenHeight = 720
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@ -6,7 +6,9 @@ const (
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const (
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SIGNAL_ANSWER = "signal/answer"
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SIGNAL_OFFER = "signal/offer"
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SIGNAL_PROVIDE = "signal/provide"
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SIGNAL_CANDIDATE = "signal/candidate"
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)
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const (
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@ -27,6 +27,11 @@ type SignalAnswer struct {
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SDP string `json:"sdp"`
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}
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type SignalCandidate struct {
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Event string `json:"event"`
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Data string `json:"data"`
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}
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type MembersList struct {
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Event string `json:"event"`
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Memebers []*types.Member `json:"members"`
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@ -24,6 +24,7 @@ type Session interface {
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Write(v interface{}) error
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Send(v interface{}) error
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SignalAnswer(sdp string) error
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SignalCandidate(data string) error
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}
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type SessionManager interface {
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@ -1,8 +1,13 @@
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package types
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import (
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"time"
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)
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type Sample struct {
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Data []byte
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Samples uint32
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Timestamp time.Time
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Duration time.Duration
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}
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type WebRTCManager interface {
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@ -5,7 +5,7 @@ import (
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"encoding/binary"
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"strconv"
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"github.com/pion/webrtc/v2"
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"github.com/pion/webrtc/v3"
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)
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const OP_MOVE = 0x01
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@ -3,7 +3,7 @@ package webrtc
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import (
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"sync"
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"github.com/pion/webrtc/v2"
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"github.com/pion/webrtc/v3"
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)
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type Peer struct {
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|
@ -1,13 +1,15 @@
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package webrtc
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import (
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"encoding/json"
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"fmt"
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"io"
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"math/rand"
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"strings"
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"github.com/pion/webrtc/v2"
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"github.com/pion/webrtc/v2/pkg/media"
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"github.com/pion/interceptor"
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"github.com/pion/rtcp"
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"github.com/pion/webrtc/v3"
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"github.com/pion/webrtc/v3/pkg/media"
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"github.com/rs/zerolog"
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"github.com/rs/zerolog/log"
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@ -26,10 +28,10 @@ func New(sessions types.SessionManager, remote types.RemoteManager, config *conf
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type WebRTCManager struct {
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logger zerolog.Logger
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videoTrack *webrtc.Track
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audioTrack *webrtc.Track
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videoCodec *webrtc.RTPCodec
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audioCodec *webrtc.RTPCodec
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videoTrack *webrtc.TrackLocalStaticSample
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audioTrack *webrtc.TrackLocalStaticSample
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videoCodec webrtc.RTPCodecParameters
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audioCodec webrtc.RTPCodecParameters
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sessions types.SessionManager
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remote types.RemoteManager
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config *config.WebRTC
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@ -97,39 +99,31 @@ func (manager *WebRTCManager) CreatePeer(id string, session types.Session) (stri
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settings.SetEphemeralUDPPortRange(manager.config.EphemeralMin, manager.config.EphemeralMax)
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settings.SetNAT1To1IPs(manager.config.NAT1To1IPs, webrtc.ICECandidateTypeHost)
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settings.SetSRTPReplayProtectionWindow(512)
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// Create MediaEngine based off sdp
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engine := webrtc.MediaEngine{}
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engine.RegisterCodec(manager.audioCodec)
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engine.RegisterCodec(manager.videoCodec)
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engine.RegisterCodec(manager.audioCodec, webrtc.RTPCodecTypeAudio)
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engine.RegisterCodec(manager.videoCodec, webrtc.RTPCodecTypeVideo)
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i := &interceptor.Registry{}
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if err := webrtc.RegisterDefaultInterceptors(&engine, i); err != nil {
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return "", manager.config.ICELite, manager.config.ICEServers, err
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}
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// Create API with MediaEngine and SettingEngine
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api := webrtc.NewAPI(webrtc.WithMediaEngine(engine), webrtc.WithSettingEngine(settings))
|
||||
api := webrtc.NewAPI(webrtc.WithMediaEngine(&engine), webrtc.WithSettingEngine(settings), webrtc.WithInterceptorRegistry(i))
|
||||
|
||||
// Create new peer connection
|
||||
connection, err := api.NewPeerConnection(*configuration)
|
||||
if err != nil {
|
||||
return "", manager.config.ICELite, manager.config.ICEServers, err
|
||||
}
|
||||
|
||||
if _, err = connection.AddTransceiverFromTrack(manager.videoTrack, webrtc.RtpTransceiverInit{
|
||||
Direction: webrtc.RTPTransceiverDirectionSendonly,
|
||||
}); err != nil {
|
||||
return "", manager.config.ICELite, manager.config.ICEServers, err
|
||||
}
|
||||
|
||||
if _, err = connection.AddTransceiverFromTrack(manager.audioTrack, webrtc.RtpTransceiverInit{
|
||||
Direction: webrtc.RTPTransceiverDirectionSendonly,
|
||||
}); err != nil {
|
||||
return "", manager.config.ICELite, manager.config.ICEServers, err
|
||||
}
|
||||
|
||||
description, err := connection.CreateOffer(nil)
|
||||
if err != nil {
|
||||
return "", manager.config.ICELite, manager.config.ICEServers, err
|
||||
}
|
||||
|
||||
negotiated := true
|
||||
connection.CreateDataChannel("data", &webrtc.DataChannelInit{
|
||||
Negotiated: &negotiated,
|
||||
})
|
||||
connection.OnDataChannel(func(d *webrtc.DataChannel) {
|
||||
d.OnMessage(func(msg webrtc.DataChannelMessage) {
|
||||
if err = manager.handle(id, msg); err != nil {
|
||||
@ -138,7 +132,31 @@ func (manager *WebRTCManager) CreatePeer(id string, session types.Session) (stri
|
||||
})
|
||||
})
|
||||
|
||||
connection.SetLocalDescription(description)
|
||||
// Set the handler for ICE connection state
|
||||
// This will notify you when the peer has connected/disconnected
|
||||
connection.OnICEConnectionStateChange(func(connectionState webrtc.ICEConnectionState) {
|
||||
fmt.Printf("Connection State has changed %s \n", connectionState.String())
|
||||
})
|
||||
|
||||
rtpSender, viderr := connection.AddTrack(manager.videoTrack)
|
||||
if viderr != nil {
|
||||
return "", manager.config.ICELite, manager.config.ICEServers, viderr
|
||||
}
|
||||
|
||||
if _, err = connection.AddTrack(manager.audioTrack); err != nil {
|
||||
return "", manager.config.ICELite, manager.config.ICEServers, err
|
||||
}
|
||||
|
||||
description, err := connection.CreateOffer(nil)
|
||||
if err != nil {
|
||||
return "", manager.config.ICELite, manager.config.ICEServers, err
|
||||
}
|
||||
|
||||
err = connection.SetLocalDescription(description)
|
||||
if err != nil {
|
||||
panic(err)
|
||||
}
|
||||
|
||||
connection.OnConnectionStateChange(func(state webrtc.PeerConnectionState) {
|
||||
switch state {
|
||||
case webrtc.PeerConnectionStateDisconnected:
|
||||
@ -156,6 +174,47 @@ func (manager *WebRTCManager) CreatePeer(id string, session types.Session) (stri
|
||||
}
|
||||
})
|
||||
|
||||
connection.OnICECandidate(func(i *webrtc.ICECandidate) {
|
||||
if i != nil {
|
||||
candidateString, err := json.Marshal(i.ToJSON())
|
||||
if err != nil {
|
||||
manager.logger.Info().Msg("error")
|
||||
return
|
||||
}
|
||||
|
||||
if err = session.SignalCandidate(string(candidateString));err != nil {
|
||||
manager.logger.Info().Msg("err")
|
||||
return
|
||||
}
|
||||
}
|
||||
})
|
||||
|
||||
|
||||
// Read incoming RTCP packets
|
||||
// Before these packets are retuned they are processed by interceptors. For things
|
||||
// like NACK this needs to be called.
|
||||
go func() {
|
||||
rtcpBuf := make([]byte, 1500)
|
||||
for {
|
||||
n, _, rtcpErr := rtpSender.Read(rtcpBuf)
|
||||
if rtcpErr != nil {
|
||||
return
|
||||
}
|
||||
ps, err := rtcp.Unmarshal(rtcpBuf[:n])
|
||||
if err != nil {
|
||||
log.Printf("Unmarshal RTCP: %v", err)
|
||||
continue
|
||||
}
|
||||
for _, p := range ps {
|
||||
switch p.(type) {
|
||||
case *rtcp.TransportLayerNack:
|
||||
manager.logger.Info().Msg("got a nack")
|
||||
}
|
||||
}
|
||||
}
|
||||
}()
|
||||
|
||||
|
||||
if err := session.SetPeer(&Peer{
|
||||
id: id,
|
||||
api: api,
|
||||
@ -171,30 +230,40 @@ func (manager *WebRTCManager) CreatePeer(id string, session types.Session) (stri
|
||||
return description.SDP, manager.config.ICELite, manager.config.ICEServers, nil
|
||||
}
|
||||
|
||||
func (m *WebRTCManager) createTrack(codecName string) (*webrtc.Track, *webrtc.RTPCodec, error) {
|
||||
var codec *webrtc.RTPCodec
|
||||
func (m *WebRTCManager) createTrack(codecName string) (*webrtc.TrackLocalStaticSample, webrtc.RTPCodecParameters, error) {
|
||||
var codec webrtc.RTPCodecParameters
|
||||
var fb []webrtc.RTCPFeedback
|
||||
var fba []webrtc.RTCPFeedback
|
||||
fb = []webrtc.RTCPFeedback{
|
||||
{"goog-remb", ""},
|
||||
{"nack", ""},
|
||||
{"nack", "pli"},
|
||||
{"ccm", "fir"},
|
||||
}
|
||||
fba = []webrtc.RTCPFeedback{}
|
||||
|
||||
switch codecName {
|
||||
case webrtc.VP8:
|
||||
codec = webrtc.NewRTPVP8Codec(webrtc.DefaultPayloadTypeVP8, 90000)
|
||||
case webrtc.VP9:
|
||||
codec = webrtc.NewRTPVP9Codec(webrtc.DefaultPayloadTypeVP9, 90000)
|
||||
case webrtc.H264:
|
||||
codec = webrtc.NewRTPH264Codec(webrtc.DefaultPayloadTypeH264, 90000)
|
||||
case webrtc.Opus:
|
||||
codec = webrtc.NewRTPOpusCodec(webrtc.DefaultPayloadTypeOpus, 48000)
|
||||
case webrtc.G722:
|
||||
codec = webrtc.NewRTPG722Codec(webrtc.DefaultPayloadTypeG722, 8000)
|
||||
case webrtc.PCMU:
|
||||
codec = webrtc.NewRTPPCMUCodec(webrtc.DefaultPayloadTypePCMU, 8000)
|
||||
case webrtc.PCMA:
|
||||
codec = webrtc.NewRTPPCMACodec(webrtc.DefaultPayloadTypePCMA, 8000)
|
||||
case "VP8":
|
||||
codec = webrtc.RTPCodecParameters{ RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "video/VP8", ClockRate: 90000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fb}, PayloadType: 96,}
|
||||
case "VP9":
|
||||
codec = webrtc.RTPCodecParameters{ RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "video/VP9", ClockRate: 90000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fb}, PayloadType: 98,}
|
||||
case "H264":
|
||||
codec = webrtc.RTPCodecParameters{ RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "video/H264", ClockRate: 90000, Channels: 0, SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f", RTCPFeedback: fb}, PayloadType: 102,}
|
||||
case "Opus":
|
||||
codec = webrtc.RTPCodecParameters{ RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "audio/opus", ClockRate: 48000, Channels: 2, SDPFmtpLine: "", RTCPFeedback: fba}, PayloadType: 111,}
|
||||
case "G722":
|
||||
codec = webrtc.RTPCodecParameters{ RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "audio/G722", ClockRate: 8000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fba}, PayloadType: 9,}
|
||||
case "PCMU":
|
||||
codec = webrtc.RTPCodecParameters{ RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "audio/PCMU", ClockRate: 8000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fba}, PayloadType: 0,}
|
||||
case "PCMA":
|
||||
codec = webrtc.RTPCodecParameters{ RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "audio/PCMA", ClockRate: 8000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fba}, PayloadType: 8,}
|
||||
default:
|
||||
return nil, nil, fmt.Errorf("unknown codec %s", codecName)
|
||||
return nil, codec, fmt.Errorf("unknown codec %s", codecName)
|
||||
}
|
||||
|
||||
track, err := webrtc.NewTrack(codec.PayloadType, rand.Uint32(), "stream", "stream", codec)
|
||||
track, err := webrtc.NewTrackLocalStaticSample(codec.RTPCodecCapability, "stream", "stream")
|
||||
if err != nil {
|
||||
return nil, nil, err
|
||||
return nil, codec, err
|
||||
}
|
||||
|
||||
return track, codec, nil
|
||||
|
Reference in New Issue
Block a user