diff --git a/.gitignore b/.gitignore index 9ed3b07..d8ad1dd 100644 --- a/.gitignore +++ b/.gitignore @@ -1 +1,2 @@ *.test +/fone/fone diff --git a/fone/main.go b/fone/main.go new file mode 100644 index 0000000..db04e4a --- /dev/null +++ b/fone/main.go @@ -0,0 +1,179 @@ +package main + +// #cgo pkg-config: libpulse-simple +// #include +// #include +// #include +import "C" + +import ( + "errors" + "flag" + "github.com/jart/gosip/dsp" + "github.com/jart/gosip/rtp" + "github.com/jart/gosip/sdp" + "github.com/jart/gosip/sip" + "github.com/jart/gosip/util" + "net" + "os" + "time" + "unsafe" +) + +const ( + hz = 8000 + chans = 1 + ptime = 20 + ssize = 2 + psamps = hz / (1000 / ptime) * chans + pbytes = psamps * ssize + filename = "/var/lib/asterisk/sounds/en/cc-yougotpranked.s16" +) + +var ( + address = flag.String("sipAddress", ":9020", "Listen address") + paServerFlag = flag.String("paServer", "", "Pulse Audio server name") + paSinkFlag = flag.String("paSink", "", "Pulse Audio device or sink name") + paName = C.CString("fone") +) + +func main() { + pa, err := makePulseAudio(C.PA_STREAM_PLAYBACK, filename) + if err != nil { + panic(err) + } + defer C.pa_simple_free(pa) + defer C.pa_simple_flush(pa, nil) + + f, err := os.Open(filename) + if err != nil { + panic(err) + } + defer f.Close() + + tick := time.NewTicker(ptime * time.Millisecond) + defer tick.Stop() + func() { + for { + var buf [pbytes]byte + select { + case <-tick.C: + got, _ := f.Read(buf[:]) + if got < pbytes { + return + } + var paerr C.int + if C.pa_simple_write(pa, unsafe.Pointer(&buf[0]), pbytes, &paerr) != 0 { + panic(C.GoString(C.pa_strerror(paerr))) + } + } + } + }() + os.Exit(0) + + // Create RTP audio session. + rs, err := rtp.NewSession("") + if err != nil { + panic(err) + } + defer rs.Close() + rtpPort := uint16(rs.Sock.LocalAddr().(*net.UDPAddr).Port) + + invite := &sip.Msg{ + Method: sip.MethodInvite, + Request: &sip.URI{User: "echo", Host: "127.0.0.1", Port: 5060}, + Payload: &sdp.SDP{ + Origin: sdp.Origin{ID: util.GenerateOriginID()}, + Audio: &sdp.Media{ + Port: rtpPort, + Codecs: []sdp.Codec{sdp.ULAWCodec, sdp.DTMFCodec}, + }, + }, + } + + // Create a SIP phone call. + dl, err := sip.NewDialog(invite) + if err != nil { + panic(err) + } + + // We're going to send white noise every 20ms. + var frame rtp.Frame + awgn := dsp.NewAWGN(-45.0) + ticker := time.NewTicker(20 * time.Millisecond) + defer ticker.Stop() + + // Hangup after 200ms. + death := time.After(200 * time.Millisecond) + + // Let's GO! + var answered bool + for { + select { + case <-ticker.C: + for n := 0; n < 160; n++ { + frame[n] = awgn.Get() + } + if err := rs.Send(&frame); err != nil { + panic("RTP send failed: " + err.Error()) + } + case err := <-dl.OnErr: + panic(err) + case state := <-dl.OnState: + switch state { + case sip.DialogAnswered: + answered = true + case sip.DialogHangup: + if !answered { + panic("Call didn't get answered!") + } + return + } + case rs.Peer = <-dl.OnPeer: + case frame := <-rs.C: + rs.R <- frame + case err := <-rs.E: + panic("RTP recv failed: " + err.Error()) + rs.CloseAfterError() + dl.Hangup <- true + case <-death: + dl.Hangup <- true + } + } +} + +func makePulseAudio(direction C.pa_stream_direction_t, streamName string) (*C.pa_simple, error) { + var ss C.pa_sample_spec + ss.format = C.PA_SAMPLE_S16NE + ss.rate = hz + ss.channels = chans + + var ba C.pa_buffer_attr + ba.maxlength = pbytes * 4 + ba.tlength = pbytes + ba.prebuf = pbytes * 2 + ba.minreq = pbytes + ba.fragsize = 0xffffffff + + var paServer *C.char + if *paServerFlag != "" { + paServer = C.CString(*paServerFlag) + defer C.free(unsafe.Pointer(paServer)) + } + + var paSink *C.char + if *paSinkFlag != "" { + paSink = C.CString(*paSinkFlag) + defer C.free(unsafe.Pointer(paSink)) + } + + paStreamName := C.CString(streamName) + defer C.free(unsafe.Pointer(paStreamName)) + + var paerr C.int + pa := C.pa_simple_new(paServer, paName, direction, paSink, paStreamName, &ss, nil, &ba, &paerr) + if pa == nil { + return nil, errors.New(C.GoString(C.pa_strerror(paerr))) + } + return pa, nil +}