音频流仍然存在问题，后续debug

lib/audiodriver/wavheader.go

@@ -11,9 +11,10 @@ import (
 
 // Skip riff header and `fmt ` just 16 bytes
 const (
-	FmtHeaderOffset        int64 = 0x0c
-	FmtHeaderIDSize        int64 = 4
-	FmtHeaderChunkSizeSize int64 = 4
+	FmtHeaderOffset        = 0x0c
+	FmtHeaderIDSize        = 4
+	FmtHeaderChunkSizeSize = 4
+	FmtHeaderSizeDefault   = 16
 )
 
 type WavHeader struct {
@@ -35,6 +36,17 @@ func NewHeader(f io.Reader) (*WavHeader, error) {
 	return w, nil
 }
 
+func DefaultHeader() *WavHeader {
+	return &WavHeader{
+		Size:          uint32(FmtHeaderSizeDefault),
+		AudioFormat:   1,
+		NumChannels:   2,
+		SampleRate:    48000, // opus only support 48kHz
+		BlockAlign:    4,
+		BitsPerSample: 16,
+	}
+}
+
 func (w *WavHeader) Parse(f io.Reader) error {
 	// skip headers
 	var headers [FmtHeaderOffset]byte

lib/webrtcconnection/connection.go

@@ -24,7 +24,7 @@ func New(o *Options) (*Connection, error) {
 	connection := &Connection{
 		option: o,
 	}
-	codecSelector, err := setupCodec(o.Video.BPS, o.Audio.BPS)
+	codecSelector, err := setupCodec(o.Video.BPS)
 	if err != nil {
 		return nil, err
 	}
@@ -123,7 +123,7 @@ func (c *Connection) Regist(offer *webrtc.SessionDescription) (*webrtc.SessionDe
 	return rtc.LocalDescription(), nil
 }
 
-func setupCodec(videoBPS, audioBPS int) (*mediadevices.CodecSelector, error) {
+func setupCodec(videoBPS int) (*mediadevices.CodecSelector, error) {
 	x264Prarm, err := x264.NewParams()
 	if err != nil {
 		return nil, err
@@ -134,7 +134,6 @@ func setupCodec(videoBPS, audioBPS int) (*mediadevices.CodecSelector, error) {
 	if err != nil {
 		return nil, err
 	}
-	opusParam.BitRate = audioBPS
 
 	codecSelector := mediadevices.NewCodecSelector(
 		mediadevices.WithAudioEncoders(&opusParam),

lib/webrtcconnection/options.go

@@ -7,9 +7,6 @@ type Options struct {
 		Width  int `yaml:"width"`
 		BPS    int `yaml:"bps"`
 	} `yaml:"video"`
-	Audio struct {
-		BPS int `yaml:"bps"`
-	} `yaml:"audio"`
 }
 
 func ExampleOptions() *Options {
@@ -22,6 +19,5 @@ func ExampleOptions() *Options {
 	options.Video.BPS = 500_000
 	options.Video.Height = 768
 	options.Video.Width = 1024
-	options.Audio.BPS = 96_000
 	return options
 }

servers/qemuserver/server.go

@@ -15,15 +15,12 @@ import (
 	"github.com/sirupsen/logrus"
 )
 
-const waveHeaderSize = 16
+const waveHeaderSize = audiodriver.FmtHeaderSizeDefault
 
-var waveHeader = &audiodriver.WavHeader{
-	Size:          waveHeaderSize,
-	AudioFormat:   1,
-	NumChannels:   2,
-	SampleRate:    48000, // opus only support 48kHz
-	BlockAlign:    4,
-	BitsPerSample: 16,
+var waveHeader *audiodriver.WavHeader
+
+func init() {
+	waveHeader = audiodriver.DefaultHeader()
 }
 
 type Server struct {

web/src/components/AceScreen.vue

@@ -1,7 +1,10 @@
 <template>
   <div>
     <canvas id="vnc" />
-    <div id="data"></div>
+    <div id="data">
+      <video id="video" muted autoplay />
+      <audio id="audio" autoplay />
+    </div>
   </div>
 </template>
 
@@ -90,11 +93,11 @@ onMounted(() => {
     };
     pc.ontrack = (ev) => {
       console.log(ev);
-      const el = document.createElement(ev.track.kind);
-      el.id = ev.track.kind;
+      const el = document.querySelector(`${ev.track.kind}#${ev.track.kind}`);
       el.srcObject = ev.streams[0];
       el.autoplay = true;
-      el.controls = true;
+      el.controls = false;
+      el.oncontextmenu = () => false;
       document.getElementById("data").appendChild(el);
     };