Add PJSUA demo code with custom audio port.

main.cpp

@@ -40,7 +40,7 @@ void mumble_audio_callback(uint8_t *audio_data, uint32_t audio_data_size) {
         int64_t sessionId;
         int64_t sequenceNumber;
         int64_t opusDataLength;
-        bool lastPacket = false;
+        bool lastPacket;
 
         dataPointer += mumble_parse_variant(&sessionId, &audio_data[dataPointer]);
         dataPointer += mumble_parse_variant(&sequenceNumber, &audio_data[dataPointer]);

sim.c

@@ -0,0 +1,296 @@
+/* $Id: simple_pjsua.c 3553 2011-05-05 06:14:19Z nanang $ */
+/*
+ * Copyright (C) 2008-2011 Teluu Inc. (http://www.teluu.com)
+ * Copyright (C) 2003-2008 Benny Prijono <benny@prijono.org>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <pjsua-lib/pjsua.h>
+
+#include <pjsip.h>
+#include <pjmedia.h>
+#include <pjlib.h>
+
+#include <stdlib.h>     /* atoi() */
+#include <stdio.h>
+#include <math.h>       /* sin()  */
+
+#define THIS_FILE "APP"
+
+#define SIP_DOMAIN "sip.antisip.com"
+#define SIP_USER "melangtone"
+#define SIP_PASSWD "b8DU9AZXbd9tVCWg"
+
+
+/* Struct attached to sine generator */
+typedef struct {
+} port_data;
+
+pjmedia_port *sine_port;
+int slot;
+
+/* This callback is called to feed more samples */
+static pj_status_t sine_get_frame(pjmedia_port *port,
+                                  pjmedia_frame *frame) {
+    port_data *sine = port->port_data.pdata;
+    pj_int16_t *samples = frame->buf;
+    pj_size_t count;
+
+    if (PJMEDIA_PIA_CCNT(&port->info) != 1) {
+        //todo throw exception
+    }
+
+    /* Get number of samples */
+    count = frame->size / 2 / PJMEDIA_PIA_CCNT(&port->info);
+
+    for (int i = 0; i < count; ++i) {
+        samples[i] = 10000.8 * (i % 10);
+    }
+
+    /* Must set frame->type correctly, otherwise the sound device
+     * will refuse to play.
+     */
+    frame->type = PJMEDIA_FRAME_TYPE_AUDIO;
+
+    return PJ_SUCCESS;
+}
+
+static pj_status_t sine_put_frame(pjmedia_port *port,
+                                  pjmedia_frame *frame) {
+
+    int sum = 0;
+    pj_int16_t *samples = frame->buf;
+    for (int i = 0; i < frame->size / 2; i++) {
+        sum += samples[i];
+    }
+    printf("%d\n", sum);
+
+    return PJ_SUCCESS;
+}
+
+/*
+ * Create a media port to generate sine wave samples.
+ */
+static pj_status_t create_sine_port(pj_pool_t *pool,
+                                    unsigned sampling_rate,
+                                    unsigned channel_count,
+                                    pjmedia_port **p_port) {
+    pjmedia_port *port;
+    unsigned i;
+    unsigned count;
+    pj_str_t name;
+    port_data *sine;
+
+    PJ_ASSERT_RETURN(pool && channel_count > 0 && channel_count <= 2, PJ_EINVAL);
+
+    port = pj_pool_zalloc(pool, sizeof(pjmedia_port));
+    PJ_ASSERT_RETURN(port != NULL, PJ_ENOMEM);
+
+    /* Fill in port info. */
+    name = pj_str("sine generator");
+    pjmedia_port_info_init(&port->info, &name,
+                           PJMEDIA_SIG_CLASS_PORT_AUD('s', 'i'),
+                           sampling_rate,
+                           channel_count,
+                           16, sampling_rate * 20 / 1000 * channel_count);
+
+    /* Set the function to feed frame */
+    port->get_frame = &sine_get_frame;
+    port->put_frame = &sine_put_frame;
+
+    /* Create sine port data */
+    port->port_data.pdata = sine = pj_pool_zalloc(pool, sizeof(port_data));
+
+    *p_port = port;
+
+    return PJ_SUCCESS;
+}
+
+
+/* Callback called by the library upon receiving incoming call */
+static void on_incoming_call(pjsua_acc_id acc_id, pjsua_call_id call_id,
+                             pjsip_rx_data *rdata) {
+    pjsua_call_info ci;
+
+    PJ_UNUSED_ARG(acc_id);
+    PJ_UNUSED_ARG(rdata);
+
+    pjsua_call_get_info(call_id, &ci);
+
+    PJ_LOG(3, (THIS_FILE, "Incoming call from %.*s!!",
+            (int) ci.remote_info.slen,
+            ci.remote_info.ptr));
+
+    /* Automatically answer incoming calls with 200/OK */
+    pjsua_call_answer(call_id, 200, NULL, NULL);
+}
+
+/* Callback called by the library when call's state has changed */
+static void on_call_state(pjsua_call_id call_id, pjsip_event *e) {
+    pjsua_call_info ci;
+
+    PJ_UNUSED_ARG(e);
+
+    pjsua_call_get_info(call_id, &ci);
+    PJ_LOG(3, (THIS_FILE, "Call %d state=%.*s", call_id,
+            (int) ci.state_text.slen,
+            ci.state_text.ptr));
+}
+
+/* Callback called by the library when call's media state has changed */
+static void on_call_media_state(pjsua_call_id call_id) {
+    pjsua_call_info ci;
+
+    pjsua_call_get_info(call_id, &ci);
+
+    if (ci.media_status == PJSUA_CALL_MEDIA_ACTIVE) {
+        // When media is active, connect call to sound device.
+        pjsua_conf_connect(ci.conf_slot, slot);
+        pjsua_conf_connect(slot, ci.conf_slot);
+    }
+}
+
+/* Display error and exit application */
+static void error_exit(const char *title, pj_status_t status) {
+    pjsua_perror(THIS_FILE, title, status);
+    pjsua_destroy();
+    exit(1);
+}
+
+/*
+ * main()
+ *
+ * argv[1] may contain URL to call.
+ */
+int main(int argc, char *argv[]) {
+    pjsua_acc_id acc_id;
+    pj_status_t status;
+    pj_pool_t *pool;
+
+    int channel_count = 1;
+
+    /* Create pjsua first! */
+    status = pjsua_create();
+    if (status != PJ_SUCCESS) error_exit("Error in pjsua_create()", status);
+
+    /* If argument is specified, it's got to be a valid SIP URL */
+    if (argc > 1) {
+        status = pjsua_verify_url(argv[1]);
+        if (status != PJ_SUCCESS) error_exit("Invalid URL in argv", status);
+    }
+
+    /* Init pjsua */
+    {
+        pjsua_config cfg;
+        pjsua_logging_config log_cfg;
+
+        pjsua_config_default(&cfg);
+        cfg.cb.on_incoming_call = &on_incoming_call;
+        cfg.cb.on_call_media_state = &on_call_media_state;
+        cfg.cb.on_call_state = &on_call_state;
+
+        pjsua_logging_config_default(&log_cfg);
+        log_cfg.console_level = 4;
+
+        status = pjsua_init(&cfg, &log_cfg, NULL);
+        if (status != PJ_SUCCESS) error_exit("Error in pjsua_init()", status);
+    }
+
+    /* Add UDP transport. */
+    {
+        pjsua_transport_config cfg;
+
+        pjsua_transport_config_default(&cfg);
+        cfg.port = 5060;
+        status = pjsua_transport_create(PJSIP_TRANSPORT_UDP, &cfg, NULL);
+        if (status != PJ_SUCCESS) error_exit("Error creating transport", status);
+    }
+
+    pj_caching_pool cp;
+    pj_caching_pool_init(&cp, &pj_pool_factory_default_policy, 0);
+
+    pool = pj_pool_create(&cp.factory,     /* pool factory         */
+                          "wav",           /* pool name.           */
+                          4000,            /* init size            */
+                          4000,            /* increment size       */
+                          NULL             /* callback on error    */
+    );
+
+    status = create_sine_port(pool,        /* memory pool          */
+                              8000,       /* sampling rate        */
+                              channel_count,/* # of channels       */
+                              &sine_port   /* returned port        */
+    );
+    if (status != PJ_SUCCESS) {
+        error_exit("Unable to create sine port", status);
+        return 1;
+    }
+
+    pjsua_set_null_snd_dev();
+
+    pjsua_conf_add_port(pool, sine_port, &slot);
+
+    /* Initialization is done, now start pjsua */
+    status = pjsua_start();
+    if (status != PJ_SUCCESS) error_exit("Error starting pjsua", status);
+
+    /* Register to SIP server by creating SIP account. */
+    {
+        pjsua_acc_config cfg;
+
+        pjsua_acc_config_default(&cfg);
+        cfg.id = pj_str("sip:" SIP_USER "@" SIP_DOMAIN);
+        cfg.reg_uri = pj_str("sip:" SIP_DOMAIN);
+        cfg.cred_count = 1;
+        cfg.cred_info[0].realm = pj_str(SIP_DOMAIN);
+        cfg.cred_info[0].scheme = pj_str("digest");
+        cfg.cred_info[0].username = pj_str(SIP_USER);
+        cfg.cred_info[0].data_type = PJSIP_CRED_DATA_PLAIN_PASSWD;
+        cfg.cred_info[0].data = pj_str(SIP_PASSWD);
+
+        status = pjsua_acc_add(&cfg, PJ_TRUE, &acc_id);
+        if (status != PJ_SUCCESS) error_exit("Error adding account", status);
+    }
+
+    /* If URL is specified, make call to the URL. */
+    if (argc > 1) {
+        pj_str_t uri = pj_str(argv[1]);
+        status = pjsua_call_make_call(acc_id, &uri, 0, NULL, NULL, NULL);
+        if (status != PJ_SUCCESS) error_exit("Error making call", status);
+    }
+
+    /* Wait until user press "q" to quit. */
+    for (; ;) {
+        char option[10];
+
+        puts("Press 'h' to hangup all calls, 'q' to quit");
+        if (fgets(option, sizeof(option), stdin) == NULL) {
+            puts("EOF while reading stdin, will quit now..");
+            break;
+        }
+
+        if (option[0] == 'q')
+            break;
+
+        if (option[0] == 'h')
+            pjsua_call_hangup_all();
+    }
+
+    /* Destroy pjsua */
+    pjsua_destroy();
+
+    return 0;
+}