iridium/neko-custom
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update to pion v3

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This commit is contained in:
Marcel Battista 2021-02-14 16:30:24 +00:00
parent 00a785f4c5
commit a362df4976
14 changed files with 211 additions and 84 deletions
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15
client/src/neko/base.ts
View File

@ -2,7 +2,7 @@ import EventEmitter from 'eventemitter3'
import { OPCODE } from './data'
import { EVENT, WebSocketEvents } from './events'
import { WebSocketMessages, WebSocketPayloads, SignalProvidePayload } from './messages'
import { WebSocketMessages, WebSocketPayloads, SignalProvidePayload, SignalCandidatePayload } from './messages'
export interface BaseEvents {
info: (...message: any[]) => void
@ -211,8 +211,8 @@ export abstract class BaseClient extends EventEmitter<BaseEvents> {
}
this._peer.ontrack = this.onTrack.bind(this)
this._peer.addTransceiver('audio', { direction: 'recvonly' })
this._peer.addTransceiver('video', { direction: 'recvonly' })
this._peer.addTransceiver('audio', { direction: 'sendrecv' })
this._peer.addTransceiver('video', { direction: 'sendrecv' })
this._channel = this._peer.createDataChannel('data')
this._channel.onerror = this.onError.bind(this)
@ -246,6 +246,15 @@ export abstract class BaseClient extends EventEmitter<BaseEvents> {
this.createPeer(sdp, lite, ice)
return
}
if (event === EVENT.SIGNAL.CANDIDATE) {
const { data } = payload as SignalCandidatePayload
let candidate: RTCIceCandidate = JSON.parse(data)
this._peer!.addIceCandidate(candidate)
return
}
// @ts-ignore
if (typeof this[event] === 'function') {

3
client/src/neko/events.ts
View File

@ -14,6 +14,7 @@ export const EVENT = {
SIGNAL: {
ANSWER: 'signal/answer',
PROVIDE: 'signal/provide',
CANDIDATE: 'signal/candidate'
},
MEMBER: {
LIST: 'member/list',
@ -78,7 +79,7 @@ export type ControlEvents =
export type SystemEvents = typeof EVENT.SYSTEM.DISCONNECT
export type MemberEvents = typeof EVENT.MEMBER.LIST | typeof EVENT.MEMBER.CONNECTED | typeof EVENT.MEMBER.DISCONNECTED
export type SignalEvents = typeof EVENT.SIGNAL.ANSWER | typeof EVENT.SIGNAL.PROVIDE
export type SignalEvents = typeof EVENT.SIGNAL.ANSWER | typeof EVENT.SIGNAL.PROVIDE | typeof EVENT.SIGNAL.CANDIDATE
export type ChatEvents = typeof EVENT.CHAT.MESSAGE | typeof EVENT.CHAT.EMOTE
export type ScreenEvents = typeof EVENT.SCREEN.CONFIGURATIONS | typeof EVENT.SCREEN.RESOLUTION | typeof EVENT.SCREEN.SET

10
client/src/neko/messages.ts
View File

@ -15,6 +15,7 @@ export type WebSocketMessages =
| WebSocketMessage
| SignalProvideMessage
| SignalAnswerMessage
| SignalCandidateMessage
| MemberListMessage
| MemberConnectMessage
| MemberDisconnectMessage
@ -26,6 +27,7 @@ export type WebSocketMessages =
export type WebSocketPayloads =
| SignalProvidePayload
| SignalAnswerPayload
| SignalCandidatePayload
| MemberListPayload
| Member
| ControlPayload
@ -78,6 +80,14 @@ export interface SignalAnswerPayload {
displayname: string
}
// signal/candidate
export interface SignalCandidateMessage extends WebSocketMessage, SignalCandidatePayload {
event: typeof EVENT.SIGNAL.CANDIDATE
}
export interface SignalCandidatePayload {
data: string
}
/*
MEMBER MESSAGES/PAYLOADS
*/

37
server/internal/gst/gst.go
View File

@ -10,10 +10,9 @@ import "C"
import (
"fmt"
"sync"
"time"
"unsafe"
"github.com/pion/webrtc/v2"
"n.eko.moe/neko/internal/types"
)
@ -80,13 +79,13 @@ func CreateRTMPPipeline(pipelineDevice string, pipelineDisplay string, pipelineS
}
// CreateAppPipeline creates a GStreamer Pipeline
func CreateAppPipeline(codecName string, pipelineDevice string, pipelineSrc string) (*Pipeline, error) {
func CreateAppPipeline(codecName string, pipelineDevice string, pipelineSrc string, bitrate string) (*Pipeline, error) {
pipelineStr := " ! appsink name=appsink"
var clockRate float32
switch codecName {
case webrtc.VP8:
case "VP8":
// https://gstreamer.freedesktop.org/documentation/vpx/vp8enc.html?gi-language=c
// gstreamer1.0-plugins-good
// vp8enc error-resilient=partitions keyframe-max-dist=10 auto-alt-ref=true cpu-used=5 deadline=1
@ -99,9 +98,9 @@ func CreateAppPipeline(codecName string, pipelineDevice string, pipelineSrc stri
if pipelineSrc != "" {
pipelineStr = fmt.Sprintf(pipelineSrc+pipelineStr, pipelineDevice)
} else {
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)
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)
}
case webrtc.VP9:
case "VP9":
// https://gstreamer.freedesktop.org/documentation/vpx/vp9enc.html?gi-language=c
// gstreamer1.0-plugins-good
// vp9enc
@ -117,7 +116,7 @@ func CreateAppPipeline(codecName string, pipelineDevice string, pipelineSrc stri
} else {
pipelineStr = fmt.Sprintf(videoSrc+"vp9enc"+pipelineStr, pipelineDevice)
}
case webrtc.H264:
case "H264":
// https://gstreamer.freedesktop.org/documentation/openh264/openh264enc.html?gi-language=c#openh264enc
// gstreamer1.0-plugins-bad
// openh264enc multi-thread=4 complexity=high bitrate=3072000 max-bitrate=4096000
@ -130,14 +129,21 @@ func CreateAppPipeline(codecName string, pipelineDevice string, pipelineSrc stri
if pipelineSrc != "" {
pipelineStr = fmt.Sprintf(pipelineSrc+pipelineStr, pipelineDevice)
} else {
var h264Str string
h264Str = "openh264enc multi-thread=4 complexity=high bitrate=3072000 max-bitrate=4096000 ! video/x-h264,stream-format=byte-stream"
var h264Str string
h264Str = "openh264enc multi-thread=4 complexity=high bitrate=3072000 max-bitrate=4096000 ! video/x-h264,stream-format=byte-stream"
if bitrate != "" {
h264Str = "openh264enc multi-thread=4 complexity=high bitrate=" + bitrate + "000 max-bitrate=" + bitrate + "999 ! video/x-h264,stream-format=byte-stream"
}
// https://gstreamer.freedesktop.org/documentation/x264/index.html?gi-language=c
// gstreamer1.0-plugins-ugly
// 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
if err := CheckPlugins([]string{"openh264"}); err != nil {
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"
h264Str = "video/x-raw,format=I420 ! x264enc threads=4 byte-stream=true tune=zerolatency speed-preset=veryfast ! video/x-h264,stream-format=byte-stream"
if bitrate != "" {
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"
}
if err := CheckPlugins([]string{"x264"}); err != nil {
return nil, err
@ -145,7 +151,7 @@ func CreateAppPipeline(codecName string, pipelineDevice string, pipelineSrc stri
}
pipelineStr = fmt.Sprintf(videoSrc+h264Str+pipelineStr, pipelineDevice)
}
case webrtc.Opus:
case "Opus":
// https://gstreamer.freedesktop.org/documentation/opus/opusenc.html
// gstreamer1.0-plugins-base
// opusenc
@ -160,7 +166,7 @@ func CreateAppPipeline(codecName string, pipelineDevice string, pipelineSrc stri
} else {
pipelineStr = fmt.Sprintf(audioSrc+"opusenc"+pipelineStr, pipelineDevice)
}
case webrtc.G722:
case "G722":
// https://gstreamer.freedesktop.org/documentation/libav/avenc_g722.html?gi-language=c
// gstreamer1.0-libav
// avenc_g722
@ -175,7 +181,7 @@ func CreateAppPipeline(codecName string, pipelineDevice string, pipelineSrc stri
} else {
pipelineStr = fmt.Sprintf(audioSrc+"avenc_g722"+pipelineStr, pipelineDevice)
}
case webrtc.PCMU:
case "PCMU":
// https://gstreamer.freedesktop.org/documentation/mulaw/mulawenc.html?gi-language=c
// gstreamer1.0-plugins-good
// audio/x-raw, rate=8000 ! mulawenc
@ -190,7 +196,7 @@ func CreateAppPipeline(codecName string, pipelineDevice string, pipelineSrc stri
} else {
pipelineStr = fmt.Sprintf(audioSrc+"audio/x-raw, rate=8000 ! mulawenc"+pipelineStr, pipelineDevice)
}
case webrtc.PCMA:
case "PCMA":
// https://gstreamer.freedesktop.org/documentation/alaw/alawenc.html?gi-language=c
// gstreamer1.0-plugins-good
// audio/x-raw, rate=8000 ! alawenc
@ -270,8 +276,7 @@ func goHandlePipelineBuffer(buffer unsafe.Pointer, bufferLen C.int, duration C.i
pipelinesLock.Unlock()
if ok {
samples := uint32(pipeline.ClockRate * (float32(duration) / 1000000000))
pipeline.Sample <- types.Sample{Data: C.GoBytes(buffer, bufferLen), Samples: samples}
pipeline.Sample <- types.Sample{Data: C.GoBytes(buffer, bufferLen), Timestamp: time.Now(), Duration: time.Duration(duration)}
} else {
fmt.Printf("discarding buffer, no pipeline with id %d", int(pipelineID))
}

3
server/internal/remote/manager.go
View File

@ -135,6 +135,7 @@ func (manager *RemoteManager) createPipelines() {
manager.config.VideoCodec,
manager.config.Display,
manager.config.VideoParams,
manager.config.Bitrate,
)
if err != nil {
manager.logger.Panic().Err(err).Msg("unable to create video pipeline")
@ -144,6 +145,7 @@ func (manager *RemoteManager) createPipelines() {
manager.config.AudioCodec,
manager.config.Device,
manager.config.AudioParams,
"",
)
if err != nil {
manager.logger.Panic().Err(err).Msg("unable to create audio pipeline")
@ -174,6 +176,7 @@ func (manager *RemoteManager) ChangeResolution(width int, height int, rate int)
manager.config.VideoCodec,
manager.config.Display,
manager.config.VideoParams,
manager.config.Bitrate,
)
if err != nil {
manager.logger.Panic().Err(err).Msg("unable to create new video pipeline")

10
server/internal/session/session.go
View File

@ -120,6 +120,16 @@ func (session *Session) SignalAnswer(sdp string) error {
return session.peer.SignalAnswer(sdp)
}
func (session *Session) SignalCandidate(data string) error {
if session.socket == nil {
return nil
}
return session.socket.Send(&message.SignalCandidate{
Event: event.SIGNAL_CANDIDATE,
Data: data,
});
}
func (session *Session) destroy() error {
if session.socket != nil {
if err := session.socket.Destroy(); err != nil {

27
server/internal/types/config/remote.go
View File

@ -4,7 +4,6 @@ import (
"regexp"
"strconv"
"github.com/pion/webrtc/v2"
"github.com/spf13/cobra"
"github.com/spf13/viper"
)
@ -19,6 +18,7 @@ type Remote struct {
ScreenWidth int
ScreenHeight int
ScreenRate int
Bitrate string
}
func (Remote) Init(cmd *cobra.Command) error {
@ -47,6 +47,12 @@ func (Remote) Init(cmd *cobra.Command) error {
return err
}
cmd.PersistentFlags().String("bitrate", "", "set this video bitrate when possible")
if err := viper.BindPFlag("bitrate", cmd.PersistentFlags().Lookup("bitrate")); err != nil {
return err
}
// video codecs
cmd.PersistentFlags().Bool("vp8", false, "use VP8 video codec")
if err := viper.BindPFlag("vp8", cmd.PersistentFlags().Lookup("vp8")); err != nil {
@ -88,24 +94,24 @@ func (Remote) Init(cmd *cobra.Command) error {
}
func (s *Remote) Set() {
videoCodec := webrtc.VP8
videoCodec := "VP8"
if viper.GetBool("vp8") {
videoCodec = webrtc.VP8
videoCodec = "VP8"
} else if viper.GetBool("vp9") {
videoCodec = webrtc.VP9
videoCodec = "VP9"
} else if viper.GetBool("h264") {
videoCodec = webrtc.H264
videoCodec = "H264"
}
audioCodec := webrtc.Opus
audioCodec := "Opus"
if viper.GetBool("opus") {
audioCodec = webrtc.Opus
audioCodec = "Opus"
} else if viper.GetBool("g722") {
audioCodec = webrtc.G722
audioCodec = "G722"
} else if viper.GetBool("pcmu") {
audioCodec = webrtc.PCMU
audioCodec = "PCMU"
} else if viper.GetBool("pcma") {
audioCodec = webrtc.PCMA
audioCodec = "PCMA"
}
s.Device = viper.GetString("device")
@ -114,6 +120,7 @@ func (s *Remote) Set() {
s.Display = viper.GetString("display")
s.VideoCodec = videoCodec
s.VideoParams = viper.GetString("video")
s.Bitrate = viper.GetString("bitrate")
s.ScreenWidth = 1280
s.ScreenHeight = 720

6
server/internal/types/event/events.go
View File

@ -5,8 +5,10 @@ const (
)
const (
SIGNAL_ANSWER = "signal/answer"
SIGNAL_PROVIDE = "signal/provide"
SIGNAL_ANSWER = "signal/answer"
SIGNAL_OFFER = "signal/offer"
SIGNAL_PROVIDE = "signal/provide"
SIGNAL_CANDIDATE = "signal/candidate"
)
const (

5
server/internal/types/message/messages.go
View File

@ -27,6 +27,11 @@ type SignalAnswer struct {
SDP string `json:"sdp"`
}
type SignalCandidate struct {
Event string `json:"event"`
Data string `json:"data"`
}
type MembersList struct {
Event string `json:"event"`
Memebers []*types.Member `json:"members"`

1
server/internal/types/session.go
View File

@ -24,6 +24,7 @@ type Session interface {
Write(v interface{}) error
Send(v interface{}) error
SignalAnswer(sdp string) error
SignalCandidate(data string) error
}
type SessionManager interface {

9
server/internal/types/webrtc.go
View File

@ -1,8 +1,13 @@
package types
import (
"time"
)
type Sample struct {
Data []byte
Samples uint32
Data []byte
Timestamp time.Time
Duration time.Duration
}
type WebRTCManager interface {

2
server/internal/webrtc/handle.go
View File

@ -5,7 +5,7 @@ import (
"encoding/binary"
"strconv"
"github.com/pion/webrtc/v2"
"github.com/pion/webrtc/v3"
)
const OP_MOVE = 0x01

2
server/internal/webrtc/peer.go
View File

@ -3,7 +3,7 @@ package webrtc
import (
"sync"
"github.com/pion/webrtc/v2"
"github.com/pion/webrtc/v3"
)
type Peer struct {

165
server/internal/webrtc/webrtc.go
View File

@ -1,13 +1,15 @@
package webrtc
import (
"encoding/json"
"fmt"
"io"
"math/rand"
"strings"
"github.com/pion/webrtc/v2"
"github.com/pion/webrtc/v2/pkg/media"
"github.com/pion/interceptor"
"github.com/pion/rtcp"
"github.com/pion/webrtc/v3"
"github.com/pion/webrtc/v3/pkg/media"
"github.com/rs/zerolog"
"github.com/rs/zerolog/log"
@ -26,10 +28,10 @@ func New(sessions types.SessionManager, remote types.RemoteManager, config *conf
type WebRTCManager struct {
logger zerolog.Logger
videoTrack *webrtc.Track
audioTrack *webrtc.Track
videoCodec *webrtc.RTPCodec
audioCodec *webrtc.RTPCodec
videoTrack *webrtc.TrackLocalStaticSample
audioTrack *webrtc.TrackLocalStaticSample
videoCodec webrtc.RTPCodecParameters
audioCodec webrtc.RTPCodecParameters
sessions types.SessionManager
remote types.RemoteManager
config *config.WebRTC
@ -97,39 +99,31 @@ func (manager *WebRTCManager) CreatePeer(id string, session types.Session) (stri
settings.SetEphemeralUDPPortRange(manager.config.EphemeralMin, manager.config.EphemeralMax)
settings.SetNAT1To1IPs(manager.config.NAT1To1IPs, webrtc.ICECandidateTypeHost)
settings.SetSRTPReplayProtectionWindow(512)
// Create MediaEngine based off sdp
engine := webrtc.MediaEngine{}
engine.RegisterCodec(manager.audioCodec)
engine.RegisterCodec(manager.videoCodec)
engine.RegisterCodec(manager.audioCodec, webrtc.RTPCodecTypeAudio)
engine.RegisterCodec(manager.videoCodec, webrtc.RTPCodecTypeVideo)
i := &interceptor.Registry{}
if err := webrtc.RegisterDefaultInterceptors(&engine, i); err != nil {
return "", manager.config.ICELite, manager.config.ICEServers, err
}
// Create API with MediaEngine and SettingEngine
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