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Add PJSUA demo code with custom audio port.

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This commit is contained in:
Michał Słomkowski 2015-09-27 20:11:15 +02:00
parent 0a4f9555e7
commit 65bdb17180
2 changed files with 297 additions and 1 deletions
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2
main.cpp
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@ -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]);

296
sim.c Normal file
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@ -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;
}