sip-lab 1.23.0 → 1.24.0

package/pocketsphinx/model/en-us/en-us/README

@@ -0,0 +1,34 @@
+/* ====================================================================
+ * Copyright (c) 2015 Alpha Cephei Inc. All rights
+ * reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright
+ *    notice, this list of conditions and the following disclaimer.
+ *
+ * 2. Redistributions in binary form must reproduce the above copyright
+ *    notice, this list of conditions and the following disclaimer in
+ *    the documentation and/or other materials provided with the
+ *    distribution.
+ *
+ * THIS SOFTWARE IS PROVIDED BY ALPHA CEPHEI INC. ``AS IS'' AND.
+ * ANY EXPRESSED OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO,.
+ * THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
+ * PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL ALPHA CEPHEI INC.
+ * NOR ITS EMPLOYEES BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT.
+ * LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,.
+ * DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY.
+ * THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT.
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE.
+ * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ *
+ * ====================================================================
+ *
+ */
+
+This directory contains generic US english acoustic model trained with
+latest sphinxtrain.

package/pocketsphinx/model/en-us/en-us/feat.params

@@ -0,0 +1,12 @@
+-lowerf 130
+-upperf 6800
+-nfilt 25
+-transform dct
+-lifter 22
+-feat 1s_c_d_dd
+-svspec 0-12/13-25/26-38
+-agc none
+-cmn batch
+-varnorm no
+-model ptm
+-remove_noise yes

package/pocketsphinx/model/en-us/en-us/noisedict

@@ -0,0 +1,5 @@
+<s> SIL
+</s> SIL
+<sil> SIL
+[NOISE] +NSN+
+[SPEECH] +SPN+

package/samples/play_wav_and_speech_recog.bad_transcript.pcmu8000.js

@@ -0,0 +1,182 @@
+var sip = require ('../index.js')
+var Zeq = require('@mayama/zeq')
+var z = new Zeq()
+var m = require('data-matching')
+var sip_msg = require('sip-matching')
+var sdp = require('sdp-matching')
+
+async function test() {
+    //sip.set_log_level(6)
+    sip.dtmf_aggregation_on(500)
+
+    sip.set_codecs("pcmu/8000/1:128")
+
+    z.trap_events(sip.event_source, 'event', (evt) => {
+        var e = evt.args[0]
+        return e
+    })
+
+    console.log(sip.start((data) => { console.log(data)} ))
+
+    t1 = sip.transport.create({address: "127.0.0.1", type: 'tcp'})
+    t2 = sip.transport.create({address: "127.0.0.1", type: 'tcp'})
+
+    console.log("t1", t1)
+    console.log("t2", t2)
+
+    oc = sip.call.create(t1.id, {
+        from_uri: '"abc"<sip:alice@test.com>',
+        to_uri: `sip:bob@${t2.address}:${t2.port}`,
+        headers: {
+            'X-MyHeader1': 'abc',
+            'X-MyHeader2': 'def',
+        },
+    })
+
+    await z.wait([
+        {
+            event: "incoming_call",
+            call_id: m.collect("call_id"),
+            msg: sip_msg({
+                $rm: 'INVITE',
+                $fU: 'alice',
+                $fd: 'test.com',
+                $tU: 'bob',
+                '$hdr(X-MyHeader1)': 'abc',
+                'hdr_x_myheader2': 'def',
+            }),
+        },
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '100',
+                $rr: 'Trying',
+                '$(hdrcnt(via))': 1,
+                'hdr_call_id': m.collect('sip_call_id'),
+                $fU: 'alice',
+                $fd: 'test.com',
+                $tU: 'bob',
+                '$hdr(l)': '0',
+            }),
+        },
+    ], 1000)
+
+    ic = {
+        id: z.store.call_id,
+        sip_call_id: z.store.sip_call_id,
+    }
+
+    sip.call.respond(ic.id, {
+        code: 200,
+        reason:'OK',
+        headers: {
+            'X-MyHeader3': 'ghi',
+            'X-MyHeader4': 'jkl',
+        },
+    })
+
+    await z.wait([
+        {
+            event: 'media_update',
+            call_id: oc.id,
+            status: 'ok',
+        },
+        {
+            event: 'media_update',
+            call_id: ic.id,
+            status: 'ok',
+        },
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '200',
+                $rr: 'OK',
+                '$(hdrcnt(v))': 1,
+                $fU: 'alice',
+                $fd: 'test.com',
+                $tU: 'bob',
+                '$hdr(content-type)': 'application/sdp',
+                $rb: '!{_}a=sendrecv',
+                '$hdr(X-MyHeader3)': 'ghi',
+                '$hdr(X-MyHeader4)': 'jkl',
+            }),
+        },
+    ], 1000)
+
+    await z.sleep(100)
+
+    sip.call.start_record_wav(oc.id, {file: './oc.wav'})
+    sip.call.start_record_wav(ic.id, {file: './ic.wav'})
+
+    sip.call.start_speech_recog(oc.id)
+    sip.call.start_speech_recog(ic.id)
+
+    await z.sleep(100)
+
+    sip.call.start_play_wav(oc.id, {file: 'samples/artifacts/hello_good_morning.wav', end_of_file_event: true, no_loop: true})
+    sip.call.start_play_wav(ic.id, {file: 'samples/artifacts/hello_good_morning.wav', end_of_file_event: true, no_loop: true})
+
+    await z.wait([
+        {
+            event: 'end_of_file',
+            call_id: ic.id,
+        },
+        {
+            event: 'end_of_file',
+            call_id: oc.id,
+        },
+        {
+            event: 'speech_transcript',
+            call_id: oc.id,
+            //transcript: 'hello good morning', // bad transcript (will not match)
+        },
+        {
+            event: 'speech_transcript',
+            call_id: ic.id,
+            //transcript: 'hello good morning', // bad transcript (will not match)
+        },
+    ], 4000)
+
+    sip.call.stop_record_wav(oc.id)
+    sip.call.stop_record_wav(ic.id)
+
+    sip.call.terminate(oc.id)
+
+    await z.wait([
+        {
+            event: 'call_ended',
+            call_id: oc.id,
+        },
+        {
+            event: 'call_ended',
+            call_id: ic.id,
+        },
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'BYE',
+            msg: sip_msg({
+                $rs: '200',
+                $rr: 'OK',
+            }),
+        },
+    ], 1000)
+
+    await z.sleep(100)
+
+    console.log("Success")
+
+    sip.stop()
+}
+
+
+test()
+.catch(e => {
+    console.error(e)
+    process.exit(1)
+})
+

package/samples/speech_synth_and_recog.speex16000.js

@@ -0,0 +1,186 @@
+var sip = require ('../index.js')
+var Zeq = require('@mayama/zeq')
+var z = new Zeq()
+var m = require('data-matching')
+var sip_msg = require('sip-matching')
+var sdp = require('sdp-matching')
+
+async function test() {
+    //sip.set_log_level(6)
+    sip.dtmf_aggregation_on(500)
+
+    sip.set_codecs("speex/16000/1:128")
+
+    z.trap_events(sip.event_source, 'event', (evt) => {
+        var e = evt.args[0]
+        return e
+    })
+
+    console.log(sip.start((data) => { console.log(data)} ))
+
+    t1 = sip.transport.create({address: "127.0.0.1", type: 'tcp'})
+    t2 = sip.transport.create({address: "127.0.0.1", type: 'tcp'})
+
+    console.log("t1", t1)
+    console.log("t2", t2)
+
+    oc = sip.call.create(t1.id, {
+        from_uri: '"abc"<sip:alice@test.com>',
+        to_uri: `sip:bob@${t2.address}:${t2.port}`,
+        headers: {
+            'X-MyHeader1': 'abc',
+            'X-MyHeader2': 'def',
+        },
+    })
+
+    await z.wait([
+        {
+            event: "incoming_call",
+            call_id: m.collect("call_id"),
+            msg: sip_msg({
+                $rm: 'INVITE',
+                $fU: 'alice',
+                $fd: 'test.com',
+                $tU: 'bob',
+                '$hdr(X-MyHeader1)': 'abc',
+                'hdr_x_myheader2': 'def',
+            }),
+        },
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '100',
+                $rr: 'Trying',
+                '$(hdrcnt(via))': 1,
+                'hdr_call_id': m.collect('sip_call_id'),
+                $fU: 'alice',
+                $fd: 'test.com',
+                $tU: 'bob',
+                '$hdr(l)': '0',
+            }),
+        },
+    ], 1000)
+
+    ic = {
+        id: z.store.call_id,
+        sip_call_id: z.store.sip_call_id,
+    }
+
+    sip.call.respond(ic.id, {
+        code: 200,
+        reason:'OK',
+        headers: {
+            'X-MyHeader3': 'ghi',
+            'X-MyHeader4': 'jkl',
+        },
+    })
+
+    await z.wait([
+        {
+            event: 'media_update',
+            call_id: oc.id,
+            status: 'ok',
+        },
+        {
+            event: 'media_update',
+            call_id: ic.id,
+            status: 'ok',
+        },
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '200',
+                $rr: 'OK',
+                '$(hdrcnt(v))': 1,
+                $fU: 'alice',
+                $fd: 'test.com',
+                $tU: 'bob',
+                '$hdr(content-type)': 'application/sdp',
+                $rb: '!{_}a=sendrecv',
+                '$hdr(X-MyHeader3)': 'ghi',
+                '$hdr(X-MyHeader4)': 'jkl',
+            }),
+        },
+    ], 1000)
+
+    await z.sleep(100)
+
+    sip.call.start_record_wav(oc.id, {file: './oc.wav'})
+    sip.call.start_record_wav(ic.id, {file: './ic.wav'})
+
+    sip.call.start_speech_recog(oc.id)
+    sip.call.start_speech_recog(ic.id)
+
+    await z.sleep(100)
+
+    sip.call.start_speech_synth(oc.id, {voice: 'kal16', text: 'Good morning.', end_of_speech_event: true, no_loop: true})
+    sip.call.start_speech_synth(ic.id, {voice: 'kal16', text: 'How are you?', end_of_speech_event: true, no_loop: true})
+
+    await z.wait([
+        {
+            event: 'end_of_speech',
+            call_id: ic.id,
+        },
+        {
+            event: 'end_of_speech',
+            call_id: oc.id,
+        },
+        {
+            event: 'speech_transcript',
+            call_id: oc.id,
+            transcript: 'how are you',
+        },
+        {
+            event: 'speech_transcript',
+            call_id: ic.id,
+            transcript: 'good morning',
+        },
+    ], 3000)
+
+    sip.call.stop_speech_synth(oc.id) // this is not actually necessary. It is used just to confirm the command works
+    sip.call.stop_speech_synth(ic.id) // this is not actually necessary. It is used just to confirm the command works
+
+
+    sip.call.stop_record_wav(oc.id)
+    sip.call.stop_record_wav(ic.id)
+
+    sip.call.terminate(oc.id)
+
+    await z.wait([
+        {
+            event: 'call_ended',
+            call_id: oc.id,
+        },
+        {
+            event: 'call_ended',
+            call_id: ic.id,
+        },
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'BYE',
+            msg: sip_msg({
+                $rs: '200',
+                $rr: 'OK',
+            }),
+        },
+    ], 1000)
+
+    await z.sleep(100)
+
+    console.log("Success")
+
+    sip.stop()
+}
+
+
+test()
+.catch(e => {
+    console.error(e)
+    process.exit(1)
+})
+

package/samples/tcp_and_extra_headers.js

@@ -126,7 +126,6 @@ async function test() {
         },
     ], 2000)
 
-
     sip.call.reinvite(oc.id)
 
     await z.wait([
@@ -339,6 +338,50 @@ async function test() {
 
     await z.sleep(2000)
 
+    sip.call.reinvite(oc.id)
+
+    await z.wait([
+        {
+            event: 'reinvite',
+            call_id: ic.id
+        },
+    ], 1000)
+
+    sip.call.respond(ic.id, {code: 200, reason: 'OK'})
+
+    await z.wait([
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '100',
+            }),
+        },
+        {
+            event: 'response',
+            call_id: oc.id,
+            method: 'INVITE',
+            msg: sip_msg({
+                $rs: '200',
+                $rr: 'OK',
+                $rb: '!{_}a=sendrecv',
+            }),
+        },
+        {
+            event: 'media_update',
+            call_id: oc.id,
+            status: 'ok',
+        },
+        {
+            event: 'media_update',
+            call_id: ic.id,
+            status: 'ok',
+        },
+    ], 500)
+
+    await z.sleep(2000)
+
     stat1 = sip.call.get_stream_stat(oc.id, {media_id: 0})
     stat2 = sip.call.get_stream_stat(ic.id, {media_id: 0})
 

package/src/addon.cpp

@@ -603,6 +603,41 @@ Napi::Value call_start_speech_synth(const Napi::CallbackInfo &info) {
   return env.Null();
 }
 
+Napi::Value call_start_speech_recog(const Napi::CallbackInfo &info) {
+  Napi::Env env = info.Env();
+
+  if (info.Length() != 2) {
+    Napi::Error::New(env,
+                     "Wrong number of arguments. Expected: call_id, params.")
+        .ThrowAsJavaScriptException();
+    return env.Null();
+  }
+
+  if (!info[0].IsNumber()) {
+    Napi::TypeError::New(env, "call_id must be number.")
+        .ThrowAsJavaScriptException();
+    return env.Null();
+  }
+  int call_id = info[0].As<Napi::Number>().Int32Value();
+
+  if (!info[1].IsString()) {
+    Napi::TypeError::New(env, "params must be a JSON string.")
+        .ThrowAsJavaScriptException();
+    return env.Null();
+  }
+  const string json = info[1].As<Napi::String>().Utf8Value();
+
+  int res = pjw_call_start_speech_recog(call_id, json.c_str());
+
+  if (res != 0) {
+    Napi::Error::New(env, pjw_get_error()).ThrowAsJavaScriptException();
+    return env.Null();
+  }
+
+  return env.Null();
+}
+
+
 Napi::Value call_stop_record_wav(const Napi::CallbackInfo &info) {
   Napi::Env env = info.Env();
 
@@ -1336,6 +1371,8 @@ Napi::Object init(Napi::Env env, Napi::Object exports) {
 
   exports.Set("call_start_speech_synth", Napi::Function::New(env, call_start_speech_synth));
 
+  exports.Set("call_start_speech_recog", Napi::Function::New(env, call_start_speech_recog));
+
   exports.Set("call_stop_record_wav",
               Napi::Function::New(env, call_stop_record_wav));
   exports.Set("call_stop_play_wav",

package/src/event_templates.cpp

@@ -111,6 +111,12 @@ int make_evt_end_of_speech(char *dest, int size, long call_id) {
       "{\"event\": \"end_of_speech\", \"call_id\": %ld}", call_id);
 }
 
+int make_evt_speech_transcript(char *dest, int size, long call_id, char* transcript) {
+  return snprintf(
+      dest, size,
+      "{\"event\": \"speech_transcript\", \"call_id\": %ld, \"transcript\": \"%s\"}", call_id, transcript);
+}
+
 int make_evt_tcp_msg(char *dest, int size, long call_id, const char *protocol, char *data, int data_len) {
   return snprintf(
       dest, size,

package/src/event_templates.hpp

@@ -38,6 +38,8 @@ int make_evt_end_of_file(char *dest, int size, long call_id);
 
 int make_evt_end_of_speech(char *dest, int size, long call_id);
 
+int make_evt_speech_transcript(char *dest, int size, long call_id, char* transcript);
+
 int make_evt_tcp_msg(char *dest, int size, long call_id, const char *protocol, char *data, int data_len);
 
 #endif

package/src/pjmedia/include/pjmedia/pocketsphinx_port.h

@@ -0,0 +1,19 @@
+#ifndef __POCKETSPHINX_PORT_H__
+#define __POCKETSPHINX_PORT_H__
+
+#include <pjmedia/port.h>
+
+PJ_BEGIN_DECL
+
+PJ_DEF(pj_status_t) pjmedia_pocketsphinx_port_create( pj_pool_t *pool,
+				unsigned clock_rate,
+				unsigned channel_count,
+				unsigned samples_per_frame,
+				unsigned bits_per_sample,
+                void (*cb)(pjmedia_port*, void *user_data, char *transcript),
+                void *cb_user_data,
+				pjmedia_port **p_port);
+
+PJ_END_DECL
+
+#endif	/* __POCKETSPHINX_PORT_H__ */

package/src/pjmedia/src/pjmedia/flite_port.c

@@ -143,10 +143,7 @@ PJ_DEF(pj_status_t) pjmedia_flite_port_create( pj_pool_t *pool,
 	if (!strcasecmp(voice, "awb")) {
 		flite->v = globals.awb;
 	} else if (!strcasecmp(voice, "kal")) {
-/*  "kal" is 8kHz and the native rate is set to 16kHz
- *  so kal talks a little bit too fast ...
- *  for now: "symlink" kal to kal16
- */		flite->v = globals.kal16;
+		flite->v = globals.kal; // this uses SamplingRate of 8000. All others use 16000
 	} else if (!strcasecmp(voice, "rms")) {
 		flite->v = globals.rms;
 	} else if (!strcasecmp(voice, "slt")) {
@@ -194,12 +191,12 @@ static pj_status_t flite_get_frame(pjmedia_port *port,
     struct flite_t *flite = (struct flite_t*)port;
 
     if(!flite->w) {
-        printf("flite no data\n");
+        //printf("flite no data\n");
         frame->type = PJMEDIA_FRAME_TYPE_NONE;
         return PJ_SUCCESS;
     }
 
-    printf("written_samples=%i num_samples=%i\n", flite->written_samples, flite->w->num_samples);
+    //printf("written_samples=%i num_samples=%i\n", flite->written_samples, flite->w->num_samples);
     if (flite->written_samples + PJMEDIA_PIA_SPF(&port->info) > (unsigned)flite->w->num_samples) {
         printf("flite end of speech\n");
 
@@ -235,7 +232,7 @@ static pj_status_t flite_get_frame(pjmedia_port *port,
     memcpy(frame->buf, flite->w->samples + flite->written_samples, PJMEDIA_PIA_SPF(&port->info)*2);
     flite->written_samples += PJMEDIA_PIA_SPF(&port->info);
     frame->type = PJMEDIA_FRAME_TYPE_AUDIO;
-    printf("flite data written samples=%i\n", PJMEDIA_PIA_SPF(&port->info));
+    //printf("flite data written samples=%i\n", PJMEDIA_PIA_SPF(&port->info));
 
     return PJ_SUCCESS;
 }