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This commit is contained in:
hasagi 2019-04-15 19:20:47 +08:00
parent fdc1ad463e
commit 243d547632
4 changed files with 132 additions and 14 deletions
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83
main.go
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@ -19,7 +19,9 @@ import (
"github.com/gorilla/websocket"
pionRTC "github.com/pion/webrtc"
"gopkg.in/hraban/opus.v2"
// "gopkg.in/hraban/opus.v2"
"github.com/xlab/opus-go/opus"
"github.com/gordonklaus/portaudio"
)
const (
@ -311,22 +313,68 @@ func fanoutScreen(imageChannel chan *image.RGBA, roomID string) {
}
var cc chan float32 = make(chan float32, 100000)
func Callback(out []float32) {
var output float32
for i := range out {
if i % 2 == 0 {
select {
case sample := <-cc:
output = sample
default:
output = 0
}
out[i] = output
}
}
}
// fanoutAudio fanout outputs to all webrtc in the same room
func fanoutAudio(audioChannel chan float32, roomID string) {
var output float32
pcm := make([]float32, 240)
enc, err := opus.NewEncoder(48000, 2, opus.AppVoIP)
// ix, _ := enc.DTX() //false
// ix1, _ := enc.Bitrate() //120000
// ix2, _ := enc.Complexity() //9
// ix3, _ := enc.MaxBandwidth() //1105
// ix4, _ := enc.PacketLossPerc() //0
portaudio.Initialize()
defer portaudio.Terminate()
host, err := portaudio.DefaultHostApi()
if err != nil {
log.Println(err)
return
}
parameters := portaudio.HighLatencyParameters(nil, host.DefaultOutputDevice)
stream, err := portaudio.OpenStream(parameters, Callback)
if err != nil {
log.Println(err)
return
}
log.Println(parameters.SampleRate, parameters.Output.Channels)
stream.Start()
// dec, err := opus.NewDecoder(48000, 2)
// enc, err := opus.NewEncoder(48000, 2, opus.AppAudio)
// // ix, _ := enc.DTX() //false
// // ix1, _ := enc.Bitrate() //120000
// // ix2, _ := enc.Complexity() //9
// // ix3, _ := enc.MaxBandwidth() //1105
// // ix4, _ := enc.PacketLossPerc() //0
// enc.SetMaxBandwidth(opus.Fullband)
// enc.SetBitrateToAuto()
// enc.SetComplexity(10)
var err2 int32
dec := opus.DecoderCreate(48000, 2, &err2)
enc := opus.EncoderCreate(48000, 2, opus.ApplicationAudio, &err2)
enc.SetMaxBandwidth(opus.Fullband)
enc.SetBitrateToAuto()
enc.SetComplexity(10)
if err != nil {
log.Println("[!] Cannot create audio encoder")
return
@ -346,12 +394,13 @@ func fanoutAudio(audioChannel chan float32, roomID string) {
}
pcm[i] = output
// pcm[i] = <- audioChannel
}
}
data := make([]byte, 1000)
n, err := enc.EncodeFloat32(pcm, data)
// n, err := enc.EncodeFloat32(pcm, data)
n := opus.EncodeFloat(enc, pcm, 120, data, 1000)
if err != nil {
log.Println("[!] Failed to decode")
@ -359,6 +408,16 @@ func fanoutAudio(audioChannel chan float32, roomID string) {
}
data = data[:n]
pcm2 := make([]float32, 1000)
// n2, err := dec.DecodeFloat32(data, pcm2)
n2 := opus.DecodeFloat(dec, string(data), n, pcm2, 1000, 0)
pcm2 = pcm2[:n2]
for i := 0; i < int(n2); i++ {
cc <- pcm2[i]
}
log.Println(n, n2)
isRoomRunning := false
for _, webRTC := range rooms[roomID].rtcSessions {
// Client stopped

BIN
static/sinewave.raw Normal file
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Binary file not shown.

59
static/test.html Normal file
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@ -0,0 +1,59 @@
<html>
<body>
<button onclick="play()">hihi</button>
<script>
var audioCtx = new (window.AudioContext || window.webkitAudioContext)();
// Stereo
var channels = 1;
var sampleRate = 16000;
var bitDepth = 16;
// Create an empty two second stereo buffer at the
// sample rate of the AudioContext
var req = new XMLHttpRequest();
var sound;
req.open('GET', "/static/sinewave.raw", false);
req.overrideMimeType('text\/plain; charset=x-user-defined');
req.onreadystatechange = function (aEvt) {
if (req.readyState == 4) {
if(req.status == 200) {
sound = req.responseText;
console.log("loaded");
} else {
alert('failed loading audio');
}
}
};
req.send(null);
function play(){
frameCount = sound.length / 2;
myAudioBuffer = audioCtx.createBuffer(channels, frameCount, sampleRate);
for (var channel = 0; channel < channels; channel++) {
var nowBuffering = myAudioBuffer.getChannelData(channel, bitDepth, sampleRate);
for (var i = 0; i < frameCount; i++) {
// audio needs to be in [-1.0; 1.0]
// for this reason I also tried to divide it by 32767
// as my pcm sample is in 16-Bit. It plays still the
// same creepy sound less noisy.
var word = (sound.charCodeAt(i * 2) & 0xff) + ((sound.charCodeAt(i * 2 + 1) & 0xff) << 8);
nowBuffering[i] = ((word + 32768) % 65536 - 32768) / 32768.0;
}
}
// Get an AudioBufferSourceNode.
// This is the AudioNode to use when we want to play an AudioBuffer
var source = audioCtx.createBufferSource();
// set the buffer in the AudioBufferSourceNode
source.buffer = myAudioBuffer;
// connect the AudioBufferSourceNode to the
// destination so we can hear the sound
source.connect(audioCtx.destination);
// start the source playing
source.start();
}
</script>
</body>
</html>

4
webrtc/webrtc.go
View file

@ -281,8 +281,8 @@ func (w *WebRTC) startStreaming(vp8Track *webrtc.Track, opusTrack *webrtc.Track)
go func() {
for w.isConnected {
data := <-w.AudioChannel
// opusTrack.Write(data)
opusTrack.WriteSample(media.Sample{Data: data, Samples: uint32(len(data))})
opusTrack.Write(data)
// opusTrack.WriteSample(media.Sample{Data: data, Samples: uint32(len(data))})
}
}()