/* built-in console to talk to a phone * * (C) 2017 by Andreas Eversberg * All Rights Reserved * * 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 3 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 G* along with this program. If not, see . */ #include #include #include #include #include #include #include #include #include #include "../libsample/sample.h" #include "../libsamplerate/samplerate.h" #include "../libjitter/jitter.h" #include "../liblogging/logging.h" #include #include #include #include #include #include "testton.h" #include "console.h" #include "cause.h" #include "../libmobile/call.h" #ifdef HAVE_ALSA #include "../libsound/sound.h" #endif enum console_state { CONSOLE_IDLE = 0, /* IDLE */ CONSOLE_SETUP_RO, /* call from radio to console */ CONSOLE_SETUP_RT, /* call from console to radio */ CONSOLE_ALERTING_RO, /* call from radio to console */ CONSOLE_ALERTING_RT, /* call from console to radio */ CONSOLE_CONNECT, CONSOLE_DISCONNECT_RO, }; static const char *console_state_name[] = { "IDLE", "SETUP_RO", "SETUP_RT", "ALERTING_RO", "ALERTING_RT", "CONNECT", "DISCONNECT_RO", }; /* console call instance */ typedef struct console { osmo_cc_session_t *session; osmo_cc_session_codec_t *codec; uint32_t callref; enum console_state state; int disc_cause; /* cause that has been sent by transceiver instance for release */ char station_id[33]; char dialing[33]; char audiodev[64]; /* headphone interface, if used */ int samplerate; /* sample rate of headphone interface */ void *sound; /* headphone interface */ int buffer_size; /* sample buffer size at headphone interface */ samplerate_t srstate; /* patterns/announcement upsampling */ jitter_t dejitter; /* headphone audio dejittering */ int test_audio_pos; /* position for test tone toward mobile */ sample_t tx_buffer[160];/* transmit audio buffer */ int tx_buffer_pos; /* current position in transmit audio buffer */ const struct number_lengths *number_lengths;/* number of digits to be dialed */ int number_max_length; /* number of digits of the longest number to be dialed */ int loopback; /* loopback test for echo */ int echo_test; /* send echo back to mobile phone */ const char *digits; /* list of dialable digits */ } console_t; static console_t console; extern osmo_cc_endpoint_t *ep; static struct osmo_cc_helper_audio_codecs codecs[] = { { "L16", 8000, 1, encode_l16, decode_l16 }, { NULL, 0, 0, NULL, NULL}, }; /* stream test music */ int16_t *test_spl = NULL; int test_size = 0; int test_max = 0; static void get_test_patterns(int16_t *samples, int length) { const int16_t *spl; int size, max, pos; spl = test_spl; size = test_size; max = test_max; /* stream sample */ pos = console.test_audio_pos; while(length--) { if (pos >= size) *samples++ = 0; else *samples++ = spl[pos] >> 2; if (++pos == max) pos = 0; } console.test_audio_pos = pos; } static void console_new_state(enum console_state state) { LOGP(DCALL, LOGL_DEBUG, "Call state '%s' -> '%s'\n", console_state_name[console.state], console_state_name[state]); console.state = state; console.test_audio_pos = 0; } static void free_console(void) { if (console.session) { osmo_cc_free_session(console.session); console.session = NULL; } console.codec = NULL; console.callref = 0; } static void up_audio(struct osmo_cc_session_codec *codec, uint8_t marker, uint16_t sequence, uint32_t timestamp, uint32_t ssrc, uint8_t *payload, int payload_len) { /* save audio from transceiver to jitter buffer */ if (console.sound) { jitter_frame_t *jf; jf = jitter_frame_alloc(codec->decoder, &console, payload, payload_len, marker, sequence, timestamp, ssrc); if (!jf) return; jitter_save(&console.dejitter, jf); return; } /* if echo test is used, send echo back to mobile */ if (console.echo_test) { osmo_cc_rtp_send_ts(codec, payload, payload_len, marker, sequence, timestamp); return; } /* if no sound is used, send test tone to mobile */ if (console.state == CONSOLE_CONNECT) { int16_t spl[160]; uint8_t *payload; int payload_len; get_test_patterns(spl, 160); codec->encoder((uint8_t *)spl, 160 * 2, &payload, &payload_len, &console); osmo_cc_rtp_send(codec, payload, payload_len, 0, 1, 160); free(payload); return; } } static void request_setup(int callref, const char *dialing) { osmo_cc_msg_t *msg; msg = osmo_cc_new_msg(OSMO_CC_MSG_SETUP_REQ); /* called number */ if (dialing) osmo_cc_add_ie_called(msg, OSMO_CC_TYPE_UNKNOWN, OSMO_CC_PLAN_TELEPHONY, dialing); /* bearer capability */ osmo_cc_add_ie_bearer(msg, OSMO_CC_CODING_ITU_T, OSMO_CC_CAPABILITY_AUDIO, OSMO_CC_MODE_CIRCUIT); /* sdp offer */ console.session = osmo_cc_helper_audio_offer(&ep->session_config, NULL, codecs, up_audio, msg, 1); osmo_cc_ul_msg(ep, callref, msg); } static void request_answer(int callref, const char *connectid, const char *sdp) { osmo_cc_msg_t *msg; msg = osmo_cc_new_msg(OSMO_CC_MSG_SETUP_RSP); /* calling number */ if (connectid) osmo_cc_add_ie_calling(msg, OSMO_CC_TYPE_SUBSCRIBER, OSMO_CC_PLAN_TELEPHONY, OSMO_CC_PRESENT_ALLOWED, OSMO_CC_SCREEN_NETWORK, connectid); /* SDP */ if (sdp) osmo_cc_add_ie_sdp(msg, sdp); osmo_cc_ul_msg(ep, callref, msg); } static void request_answer_ack(int callref) { osmo_cc_msg_t *msg; msg = osmo_cc_new_msg(OSMO_CC_MSG_SETUP_COMP_REQ); osmo_cc_ul_msg(ep, callref, msg); } static void request_disconnect_release_reject(int callref, int cause, uint8_t msg_type) { osmo_cc_msg_t *msg; msg = osmo_cc_new_msg(msg_type); osmo_cc_add_ie_cause(msg, OSMO_CC_LOCATION_USER, cause, 0, 0); osmo_cc_ul_msg(ep, callref, msg); } void console_msg(osmo_cc_call_t *call, osmo_cc_msg_t *msg) { uint8_t location, isdn_cause, socket_cause; uint16_t sip_cause; uint8_t type, plan, present, screen; uint8_t progress, coding; char caller_id[33], number[33]; const char *sdp; int rc; if (msg->type != OSMO_CC_MSG_SETUP_IND && console.callref != call->callref) { LOGP(DCALL, LOGL_ERROR, "invalid call ref %u (msg=0x%02x).\n", call->callref, msg->type); request_disconnect_release_reject(call->callref, CAUSE_INVALCALLREF, OSMO_CC_MSG_REL_REQ); osmo_cc_free_msg(msg); return; } switch(msg->type) { case OSMO_CC_MSG_SETUP_IND: { /* caller id */ rc = osmo_cc_get_ie_calling(msg, 0, &type, &plan, &present, &screen, caller_id, sizeof(caller_id)); if (rc < 0) caller_id[0] = '\0'; /* dialing */ rc = osmo_cc_get_ie_called(msg, 0, &type, &plan, number, sizeof(number)); if (rc < 0) number[0] = '\0'; LOGP(DCALL, LOGL_INFO, "Incoming call from '%s'\n", caller_id); /* setup is also allowed on disconnected call */ if (console.state == CONSOLE_DISCONNECT_RO) { LOGP(DCALL, LOGL_INFO, "Releasing pending disconnected call\n"); if (console.callref) { request_disconnect_release_reject(console.callref, CAUSE_NORMAL, OSMO_CC_MSG_REL_REQ); free_console(); } console_new_state(CONSOLE_IDLE); } if (console.state != CONSOLE_IDLE) { LOGP(DCALL, LOGL_NOTICE, "We are busy, rejecting.\n"); request_disconnect_release_reject(console.callref, CAUSE_NORMAL, OSMO_CC_MSG_REJ_REQ); osmo_cc_free_msg(msg); return; } console.callref = call->callref; /* sdp accept */ sdp = osmo_cc_helper_audio_accept(&ep->session_config, NULL, codecs, up_audio, msg, &console.session, &console.codec, 0); if (!sdp) { LOGP(DCALL, LOGL_NOTICE, "Cannot accept codec, rejecting.\n"); request_disconnect_release_reject(console.callref, CAUSE_RESOURCE_UNAVAIL, OSMO_CC_MSG_REJ_REQ); osmo_cc_free_msg(msg); return; } if (caller_id[0]) { strncpy(console.station_id, caller_id, sizeof(console.station_id)); console.station_id[sizeof(console.station_id) - 1] = '\0'; } strncpy(console.dialing, number, sizeof(console.dialing) - 1); console.dialing[sizeof(console.dialing) - 1] = '\0'; console_new_state(CONSOLE_CONNECT); LOGP(DCALL, LOGL_INFO, "Call automatically answered\n"); request_answer(console.callref, number, sdp); break; } case OSMO_CC_MSG_SETUP_ACK_IND: case OSMO_CC_MSG_PROC_IND: osmo_cc_helper_audio_negotiate(msg, &console.session, &console.codec); break; case OSMO_CC_MSG_ALERT_IND: LOGP(DCALL, LOGL_INFO, "Call alerting\n"); osmo_cc_helper_audio_negotiate(msg, &console.session, &console.codec); console_new_state(CONSOLE_ALERTING_RT); break; case OSMO_CC_MSG_SETUP_CNF: { /* connected id */ rc = osmo_cc_get_ie_calling(msg, 0, &type, &plan, &present, &screen, caller_id, sizeof(caller_id)); if (rc < 0) caller_id[0] = '\0'; LOGP(DCALL, LOGL_INFO, "Call connected to '%s'\n", caller_id); osmo_cc_helper_audio_negotiate(msg, &console.session, &console.codec); console_new_state(CONSOLE_CONNECT); if (caller_id[0]) { strncpy(console.station_id, caller_id, sizeof(console.station_id)); console.station_id[sizeof(console.station_id) - 1] = '\0'; } request_answer_ack(console.callref); break; } case OSMO_CC_MSG_SETUP_COMP_IND: break; case OSMO_CC_MSG_DISC_IND: rc = osmo_cc_get_ie_cause(msg, 0, &location, &isdn_cause, &sip_cause, &socket_cause); if (rc < 0) isdn_cause = OSMO_CC_ISDN_CAUSE_NORM_CALL_CLEAR; rc = osmo_cc_get_ie_progress(msg, 0, &coding, &location, &progress); osmo_cc_helper_audio_negotiate(msg, &console.session, &console.codec); if (rc >= 0 && (progress == 1 || progress == 8)) { LOGP(DCALL, LOGL_INFO, "Call disconnected with audio (%s)\n", cause_name(isdn_cause)); console_new_state(CONSOLE_DISCONNECT_RO); console.disc_cause = isdn_cause; } else { LOGP(DCALL, LOGL_INFO, "Call disconnected without audio (%s)\n", cause_name(isdn_cause)); request_disconnect_release_reject(console.callref, isdn_cause, OSMO_CC_MSG_REL_REQ); console_new_state(CONSOLE_IDLE); free_console(); } break; case OSMO_CC_MSG_REL_IND: case OSMO_CC_MSG_REJ_IND: rc = osmo_cc_get_ie_cause(msg, 0, &location, &isdn_cause, &sip_cause, &socket_cause); if (rc < 0) isdn_cause = OSMO_CC_ISDN_CAUSE_NORM_CALL_CLEAR; LOGP(DCALL, LOGL_INFO, "Call released (%s)\n", cause_name(isdn_cause)); console_new_state(CONSOLE_IDLE); free_console(); break; } osmo_cc_free_msg(msg); } static char console_text[256]; static int console_len = 0; int console_init(const char *audiodev, int samplerate, int buffer, int loopback, int echo_test, const char *digits, const struct number_lengths *lengths, const char *station_id) { int rc = 0; int i; init_testton(); /* Put scrolling window one line above bottom. */ logging_limit_scroll_bottom(1); memset(&console, 0, sizeof(console)); strncpy(console.audiodev, audiodev, sizeof(console.audiodev) - 1); console.samplerate = samplerate; console.buffer_size = buffer * samplerate / 1000; console.loopback = loopback; console.echo_test = echo_test; console.digits = digits; console.number_lengths = lengths; if (lengths) { for (i = 0; lengths[i].usage; i++) { if (lengths[i].digits > console.number_max_length) console.number_max_length = lengths[i].digits; } } if (station_id) strncpy(console.station_id, station_id, sizeof(console.station_id) - 1); if (!audiodev[0]) return 0; rc = init_samplerate(&console.srstate, 8000.0, (double)samplerate, 3300.0); if (rc < 0) { LOGP(DSENDER, LOGL_ERROR, "Failed to init sample rate conversion!\n"); goto error; } rc = jitter_create(&console.dejitter, "console", 8000, 0.040, 0.200, JITTER_FLAG_NONE); if (rc < 0) { LOGP(DSENDER, LOGL_ERROR, "Failed to create and init dejitter buffer!\n"); goto error; } return 0; error: console_cleanup(); return rc; } int console_open_audio(int __attribute__((unused)) buffer_size, double __attribute__((unused)) interval) { if (!console.audiodev[0]) return 0; #ifdef HAVE_ALSA /* open sound device for call control */ /* use factor 1.4 of speech level for complete range of sound card */ console.sound = sound_open(SOUND_DIR_DUPLEX, console.audiodev, NULL, NULL, NULL, 1, 0.0, console.samplerate, buffer_size, interval, 1.4, 4000.0, 2.0); if (!console.sound) { LOGP(DSENDER, LOGL_ERROR, "No sound device!\n"); return -EIO; } #else LOGP(DSENDER, LOGL_ERROR, "No sound card support compiled in!\n"); return -ENOTSUP; #endif return 0; } int console_start_audio(void) { if (!console.audiodev[0]) return 0; #ifdef HAVE_ALSA return sound_start(console.sound); #else return -EINVAL; #endif } void console_cleanup(void) { #ifdef HAVE_ALSA /* close sound devoice */ if (console.sound) { sound_close(console.sound); console.sound = NULL; } #endif jitter_destroy(&console.dejitter); if (console.session) { osmo_cc_free_session(console.session); console.session = NULL; } } /* process input from console * it is not called at loopback mode * calling this implies that the console.number_lengths is set */ static void process_ui(int c) { char text[256] = ""; int len, w, h; int i; switch (console.state) { case CONSOLE_IDLE: if (c > 0) { if ((int)strlen(console.station_id) < console.number_max_length) { for (i = 0; i < (int)strlen(console.digits); i++) { if (c == console.digits[i]) { console.station_id[strlen(console.station_id) + 1] = '\0'; console.station_id[strlen(console.station_id)] = c; } } } if ((c == 8 || c == 127) && strlen(console.station_id)) console.station_id[strlen(console.station_id) - 1] = '\0'; dial_after_hangup: len = strlen(console.station_id); for (i = 0; console.number_lengths[i].usage; i++) { if (len == console.number_lengths[i].digits) break; } if (c == 'd' && console.number_lengths[i].usage) { LOGP(DCALL, LOGL_INFO, "Outgoing call to '%s'\n", console.station_id); console.dialing[0] = '\0'; console_new_state(CONSOLE_SETUP_RT); console.callref = osmo_cc_new_callref(); request_setup(console.callref, console.station_id); } } sprintf(text, "on-hook: %s%s ", console.station_id, "................................" + 32 - console.number_max_length + strlen(console.station_id)); len = strlen(console.station_id); for (i = 0; console.number_lengths[i].usage; i++) { if (len == console.number_lengths[i].digits) break; } if (console.number_lengths[i].usage) { if (console.number_lengths[i + 1].usage) sprintf(strchr(text, '\0'), "(enter digits %s or press d=dial)", console.digits); else sprintf(strchr(text, '\0'), "(press d=dial)"); } else sprintf(strchr(text, '\0'), "(enter digits %s)", console.digits); break; case CONSOLE_SETUP_RO: case CONSOLE_SETUP_RT: case CONSOLE_ALERTING_RO: case CONSOLE_ALERTING_RT: case CONSOLE_CONNECT: case CONSOLE_DISCONNECT_RO: if (c > 0) { if (c == 'h' || (c == 'd' && console.state == CONSOLE_DISCONNECT_RO)) { LOGP(DCALL, LOGL_INFO, "Call hangup\n"); if (console.callref) { if (console.state == CONSOLE_SETUP_RO) request_disconnect_release_reject(console.callref, CAUSE_NORMAL, OSMO_CC_MSG_REJ_REQ); else request_disconnect_release_reject(console.callref, CAUSE_NORMAL, OSMO_CC_MSG_REL_REQ); free_console(); } console_new_state(CONSOLE_IDLE); if (c == 'd') goto dial_after_hangup; } } if (console.state == CONSOLE_SETUP_RT) sprintf(text, "call setup: %s (press h=hangup)", console.station_id); if (console.state == CONSOLE_ALERTING_RT) sprintf(text, "call ringing: %s (press h=hangup)", console.station_id); if (console.state == CONSOLE_CONNECT) { if (console.dialing[0]) sprintf(text, "call active: %s->%s (press h=hangup)", console.station_id, console.dialing); else sprintf(text, "call active: %s (press h=hangup)", console.station_id); } if (console.state == CONSOLE_DISCONNECT_RO) sprintf(text, "call disconnected: %s (press h=hangup d=redial)", cause_name(console.disc_cause)); break; } /* skip if nothing has changed */ len = strlen(text); if (console_len == len && !memcmp(console_text, text, len)) return; /* lock logging */ lock_logging(); /* disable window */ enable_limit_scroll(false); /* geht height */ get_win_size(&w, &h); /* save cursor go to bottom, use white color */ printf("\0337\033[%d;1H\033[1;37m", h); /* copy text and pad with spaces */ console_len = len; memcpy(console_text, text, console_len); if (console_len < (int)MIN(sizeof(console_text), w)) memset(console_text + console_len, ' ', MIN(sizeof(console_text), w) - console_len); /* write text */ fwrite(console_text, MIN(sizeof(console_text), w), 1, stdout); /* reset color, go back to previous line, flush */ printf("\033[0;39m\0338"); /* flush output */ fflush(stdout); /* enable window */ enable_limit_scroll(true); /* unlock logging */ unlock_logging(); } /* get keys from keyboard to control call via console * returns 1 on exit (ctrl+c) */ void process_console(int c) { if (!console.loopback && console.number_max_length) process_ui(c); if (!console.sound) return; #ifdef HAVE_ALSA /* handle audio, if sound device is used */ sample_t samples[console.buffer_size + 10], *samples_list[1]; uint8_t *power_list[1]; int count, input_num; int rc; count = sound_get_tosend(console.sound, console.buffer_size); if (count < 0) { LOGP(DSENDER, LOGL_ERROR, "Failed to get samples in buffer (rc = %d)!\n", count); if (count == -EPIPE) LOGP(DSENDER, LOGL_ERROR, "Trying to recover.\n"); return; } if (count > 0) { /* load and upsample */ input_num = samplerate_upsample_input_num(&console.srstate, count); { int16_t spl[input_num]; jitter_load_samples(&console.dejitter, (uint8_t *)spl, input_num, sizeof(*spl), jitter_conceal_s16, NULL); int16_to_samples_speech(samples, spl, input_num); } samplerate_upsample(&console.srstate, samples, input_num, samples, count); /* write to sound device */ samples_list[0] = samples; power_list[0] = NULL; rc = sound_write(console.sound, samples_list, power_list, count, NULL, NULL, 1); if (rc < 0) { LOGP(DSENDER, LOGL_ERROR, "Failed to write TX data to sound device (rc = %d)\n", rc); if (rc == -EPIPE) LOGP(DSENDER, LOGL_ERROR, "Trying to recover.\n"); return; } } samples_list[0] = samples; count = sound_read(console.sound, samples_list, console.buffer_size, 1, NULL); if (count < 0) { LOGP(DSENDER, LOGL_ERROR, "Failed to read from sound device (rc = %d)!\n", count); if (count == -EPIPE) LOGP(DSENDER, LOGL_ERROR, "Trying to recover.\n"); return; } if (count) { int i; count = samplerate_downsample(&console.srstate, samples, count); /* put samples into ring buffer */ for (i = 0; i < count; i++) { console.tx_buffer[console.tx_buffer_pos] = samples[i]; /* if ring buffer wraps, deliver data down to call process */ if (++console.tx_buffer_pos == 160) { console.tx_buffer_pos = 0; /* only if we have a call */ if (console.callref && console.codec) { int16_t spl[160]; uint8_t *payload; int payload_len; samples_to_int16_speech(spl, console.tx_buffer, 160); console.codec->encoder((uint8_t *)spl, 160 * 2, &payload, &payload_len, &console); osmo_cc_rtp_send(console.codec, payload, payload_len, 0, 1, 160); } } } } #endif } /* Call this for every inscription. If the console's dial string is empty, it is set to the number that has been inscribed. */ int console_inscription(const char *station_id) { if (console.loopback || !console.number_max_length) return -EINVAL; if (console.station_id[0]) return 1; strncpy(console.station_id, station_id, sizeof(console.station_id) - 1); process_ui(-1); return 0; }