/* $Id$ */ /* * Copyright (C) 2008-2011 Teluu Inc. (http://www.teluu.com) * Copyright (C) 2003-2008 Benny Prijono * * 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 #include #if defined(PJSUA_MEDIA_HAS_PJMEDIA) && PJSUA_MEDIA_HAS_PJMEDIA != 0 #define THIS_FILE "pjsua_aud.c" /***************************************************************************** * * Prototypes */ /* Open sound dev */ static pj_status_t open_snd_dev(pjmedia_snd_port_param *param); /* Close existing sound device */ static void close_snd_dev(void); /* Create audio device param */ static pj_status_t create_aud_param(pjmedia_aud_param *param, pjmedia_aud_dev_index capture_dev, pjmedia_aud_dev_index playback_dev, unsigned clock_rate, unsigned channel_count, unsigned samples_per_frame, unsigned bits_per_sample); /***************************************************************************** * * Call API that are closely tied to PJMEDIA */ /* * Check if call has an active media session. */ PJ_DEF(pj_bool_t) pjsua_call_has_media(pjsua_call_id call_id) { pjsua_call *call = &pjsua_var.calls[call_id]; PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls, PJ_EINVAL); return call->audio_idx >= 0 && call->media[call->audio_idx].strm.a.stream; } /* * Get the conference port identification associated with the call. */ PJ_DEF(pjsua_conf_port_id) pjsua_call_get_conf_port(pjsua_call_id call_id) { pjsua_call *call; pjsua_conf_port_id port_id = PJSUA_INVALID_ID; PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls, PJ_EINVAL); /* Use PJSUA_LOCK() instead of acquire_call(): * https://trac.pjsip.org/repos/ticket/1371 */ PJSUA_LOCK(); if (!pjsua_call_is_active(call_id)) goto on_return; call = &pjsua_var.calls[call_id]; if (call->audio_idx >= 0) port_id = call->media[call->audio_idx].strm.a.conf_slot; on_return: PJSUA_UNLOCK(); return port_id; } /* * Get media stream info for the specified media index. */ PJ_DEF(pj_status_t) pjsua_call_get_stream_info( pjsua_call_id call_id, unsigned med_idx, pjsua_stream_info *psi) { pjsua_call *call; pjsua_call_media *call_med; pj_status_t status; PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls, PJ_EINVAL); PJ_ASSERT_RETURN(psi, PJ_EINVAL); PJSUA_LOCK(); call = &pjsua_var.calls[call_id]; if (med_idx >= call->med_cnt) { PJSUA_UNLOCK(); return PJ_EINVAL; } call_med = &call->media[med_idx]; psi->type = call_med->type; switch (call_med->type) { case PJMEDIA_TYPE_AUDIO: status = pjmedia_stream_get_info(call_med->strm.a.stream, &psi->info.aud); break; #if defined(PJMEDIA_HAS_VIDEO) && (PJMEDIA_HAS_VIDEO != 0) case PJMEDIA_TYPE_VIDEO: status = pjmedia_vid_stream_get_info(call_med->strm.v.stream, &psi->info.vid); break; #endif default: status = PJMEDIA_EINVALIMEDIATYPE; break; } PJSUA_UNLOCK(); return status; } /* * Get media stream statistic for the specified media index. */ PJ_DEF(pj_status_t) pjsua_call_get_stream_stat( pjsua_call_id call_id, unsigned med_idx, pjsua_stream_stat *stat) { pjsua_call *call; pjsua_call_media *call_med; pj_status_t status; PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls, PJ_EINVAL); PJ_ASSERT_RETURN(stat, PJ_EINVAL); PJSUA_LOCK(); call = &pjsua_var.calls[call_id]; if (med_idx >= call->med_cnt) { PJSUA_UNLOCK(); return PJ_EINVAL; } call_med = &call->media[med_idx]; switch (call_med->type) { case PJMEDIA_TYPE_AUDIO: status = pjmedia_stream_get_stat(call_med->strm.a.stream, &stat->rtcp); if (status == PJ_SUCCESS) status = pjmedia_stream_get_stat_jbuf(call_med->strm.a.stream, &stat->jbuf); break; #if defined(PJMEDIA_HAS_VIDEO) && (PJMEDIA_HAS_VIDEO != 0) case PJMEDIA_TYPE_VIDEO: status = pjmedia_vid_stream_get_stat(call_med->strm.v.stream, &stat->rtcp); if (status == PJ_SUCCESS) status = pjmedia_vid_stream_get_stat_jbuf(call_med->strm.v.stream, &stat->jbuf); break; #endif default: status = PJMEDIA_EINVALIMEDIATYPE; break; } PJSUA_UNLOCK(); return status; } /* * Send DTMF digits to remote using RFC 2833 payload formats. */ PJ_DEF(pj_status_t) pjsua_call_dial_dtmf( pjsua_call_id call_id, const pj_str_t *digits) { pjsua_call *call; pjsip_dialog *dlg = NULL; pj_status_t status; PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls, PJ_EINVAL); PJ_LOG(4,(THIS_FILE, "Call %d dialing DTMF %.*s", call_id, (int)digits->slen, digits->ptr)); pj_log_push_indent(); status = acquire_call("pjsua_call_dial_dtmf()", call_id, &call, &dlg); if (status != PJ_SUCCESS) goto on_return; if (!pjsua_call_has_media(call_id)) { PJ_LOG(3,(THIS_FILE, "Media is not established yet!")); status = PJ_EINVALIDOP; goto on_return; } status = pjmedia_stream_dial_dtmf( call->media[call->audio_idx].strm.a.stream, digits); on_return: if (dlg) pjsip_dlg_dec_lock(dlg); pj_log_pop_indent(); return status; } /***************************************************************************** * * Audio media with PJMEDIA backend */ /* Init pjmedia audio subsystem */ pj_status_t pjsua_aud_subsys_init() { pj_str_t codec_id = {NULL, 0}; unsigned opt; pjmedia_audio_codec_config codec_cfg; pj_status_t status; /* To suppress warning about unused var when all codecs are disabled */ PJ_UNUSED_ARG(codec_id); /* * Register all codecs */ pjmedia_audio_codec_config_default(&codec_cfg); codec_cfg.speex.quality = pjsua_var.media_cfg.quality; codec_cfg.speex.complexity = -1; codec_cfg.ilbc.mode = pjsua_var.media_cfg.ilbc_mode; #if PJMEDIA_HAS_PASSTHROUGH_CODECS /* Register passthrough codecs */ { unsigned aud_idx; unsigned ext_fmt_cnt = 0; pjmedia_format ext_fmts[32]; /* List extended formats supported by audio devices */ for (aud_idx = 0; aud_idx < pjmedia_aud_dev_count(); ++aud_idx) { pjmedia_aud_dev_info aud_info; unsigned i; status = pjmedia_aud_dev_get_info(aud_idx, &aud_info); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Error querying audio device info", status); goto on_error; } /* Collect extended formats supported by this audio device */ for (i = 0; i < aud_info.ext_fmt_cnt; ++i) { unsigned j; pj_bool_t is_listed = PJ_FALSE; /* See if this extended format is already in the list */ for (j = 0; j < ext_fmt_cnt && !is_listed; ++j) { if (ext_fmts[j].id == aud_info.ext_fmt[i].id && ext_fmts[j].det.aud.avg_bps == aud_info.ext_fmt[i].det.aud.avg_bps) { is_listed = PJ_TRUE; } } /* Put this format into the list, if it is not in the list */ if (!is_listed) ext_fmts[ext_fmt_cnt++] = aud_info.ext_fmt[i]; pj_assert(ext_fmt_cnt <= PJ_ARRAY_SIZE(ext_fmts)); } } /* Init the passthrough codec with supported formats only */ codec_cfg.passthrough.setting.fmt_cnt = ext_fmt_cnt; codec_cfg.passthrough.setting.fmts = ext_fmts; codec_cfg.passthrough.setting.ilbc_mode = pjsua_var.media_cfg.ilbc_mode; } #endif /* PJMEDIA_HAS_PASSTHROUGH_CODECS */ /* Register all codecs */ status = pjmedia_codec_register_audio_codecs(pjsua_var.med_endpt, &codec_cfg); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Error registering codecs", status); goto on_error; } /* Set speex/16000 to higher priority*/ codec_id = pj_str("speex/16000"); pjmedia_codec_mgr_set_codec_priority( pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt), &codec_id, PJMEDIA_CODEC_PRIO_NORMAL+2); /* Set speex/8000 to next higher priority*/ codec_id = pj_str("speex/8000"); pjmedia_codec_mgr_set_codec_priority( pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt), &codec_id, PJMEDIA_CODEC_PRIO_NORMAL+1); /* Disable ALL L16 codecs */ codec_id = pj_str("L16"); pjmedia_codec_mgr_set_codec_priority( pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt), &codec_id, PJMEDIA_CODEC_PRIO_DISABLED); /* Save additional conference bridge parameters for future * reference. */ pjsua_var.mconf_cfg.channel_count = pjsua_var.media_cfg.channel_count; pjsua_var.mconf_cfg.bits_per_sample = 16; pjsua_var.mconf_cfg.samples_per_frame = pjsua_var.media_cfg.clock_rate * pjsua_var.mconf_cfg.channel_count * pjsua_var.media_cfg.audio_frame_ptime / 1000; /* Init options for conference bridge. */ opt = PJMEDIA_CONF_NO_DEVICE; if (pjsua_var.media_cfg.quality >= 3 && pjsua_var.media_cfg.quality <= 4) { opt |= PJMEDIA_CONF_SMALL_FILTER; } else if (pjsua_var.media_cfg.quality < 3) { opt |= PJMEDIA_CONF_USE_LINEAR; } /* Init conference bridge. */ status = pjmedia_conf_create(pjsua_var.pool, pjsua_var.media_cfg.max_media_ports, pjsua_var.media_cfg.clock_rate, pjsua_var.mconf_cfg.channel_count, pjsua_var.mconf_cfg.samples_per_frame, pjsua_var.mconf_cfg.bits_per_sample, opt, &pjsua_var.mconf); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Error creating conference bridge", status); goto on_error; } /* Are we using the audio switchboard (a.k.a APS-Direct)? */ pjsua_var.is_mswitch = pjmedia_conf_get_master_port(pjsua_var.mconf) ->info.signature == PJMEDIA_CONF_SWITCH_SIGNATURE; /* Create null port just in case user wants to use null sound. */ status = pjmedia_null_port_create(pjsua_var.pool, pjsua_var.media_cfg.clock_rate, pjsua_var.mconf_cfg.channel_count, pjsua_var.mconf_cfg.samples_per_frame, pjsua_var.mconf_cfg.bits_per_sample, &pjsua_var.null_port); PJ_ASSERT_RETURN(status == PJ_SUCCESS, status); return status; on_error: return status; } /* Check if sound device is idle. */ void pjsua_check_snd_dev_idle() { unsigned call_cnt; /* Check if the sound device auto-close feature is disabled. */ if (pjsua_var.media_cfg.snd_auto_close_time < 0) return; /* Check if the sound device is currently closed. */ if (!pjsua_var.snd_is_on) return; /* Get the call count, we shouldn't close the sound device when there is * any calls active. */ call_cnt = pjsua_call_get_count(); /* When this function is called from pjsua_media_channel_deinit() upon * disconnecting call, actually the call count hasn't been updated/ * decreased. So we put additional check here, if there is only one * call and it's in DISCONNECTED state, there is actually no active * call. */ if (call_cnt == 1) { pjsua_call_id call_id; pj_status_t status; status = pjsua_enum_calls(&call_id, &call_cnt); if (status == PJ_SUCCESS && call_cnt > 0 && !pjsua_call_is_active(call_id)) { call_cnt = 0; } } /* Activate sound device auto-close timer if sound device is idle. * It is idle when there is no port connection in the bridge and * there is no active call. */ if (pjsua_var.snd_idle_timer.id == PJ_FALSE && call_cnt == 0 && pjmedia_conf_get_connect_count(pjsua_var.mconf) == 0) { pj_time_val delay; delay.msec = 0; delay.sec = pjsua_var.media_cfg.snd_auto_close_time; pjsua_var.snd_idle_timer.id = PJ_TRUE; pjsip_endpt_schedule_timer(pjsua_var.endpt, &pjsua_var.snd_idle_timer, &delay); } } /* Timer callback to close sound device */ static void close_snd_timer_cb( pj_timer_heap_t *th, pj_timer_entry *entry) { PJ_UNUSED_ARG(th); PJSUA_LOCK(); if (entry->id) { PJ_LOG(4,(THIS_FILE,"Closing sound device after idle for %d second(s)", pjsua_var.media_cfg.snd_auto_close_time)); entry->id = PJ_FALSE; close_snd_dev(); } PJSUA_UNLOCK(); } pj_status_t pjsua_aud_subsys_start(void) { pj_status_t status = PJ_SUCCESS; pj_timer_entry_init(&pjsua_var.snd_idle_timer, PJ_FALSE, NULL, &close_snd_timer_cb); pjsua_check_snd_dev_idle(); return status; } pj_status_t pjsua_aud_subsys_destroy() { unsigned i; close_snd_dev(); if (pjsua_var.mconf) { pjmedia_conf_destroy(pjsua_var.mconf); pjsua_var.mconf = NULL; } if (pjsua_var.null_port) { pjmedia_port_destroy(pjsua_var.null_port); pjsua_var.null_port = NULL; } /* Destroy file players */ for (i=0; istrm.a.stream; pjmedia_rtcp_stat stat; if (strm) { /* Unsubscribe from stream events */ pjmedia_event_unsubscribe(NULL, &call_media_on_event, call_med, strm); pjmedia_stream_send_rtcp_bye(strm); if (call_med->strm.a.conf_slot != PJSUA_INVALID_ID) { if (pjsua_var.mconf) { pjsua_conf_remove_port(call_med->strm.a.conf_slot); } call_med->strm.a.conf_slot = PJSUA_INVALID_ID; } if ((call_med->dir & PJMEDIA_DIR_ENCODING) && (pjmedia_stream_get_stat(strm, &stat) == PJ_SUCCESS) && stat.tx.pkt) { /* Save RTP timestamp & sequence, so when media session is * restarted, those values will be restored as the initial * RTP timestamp & sequence of the new media session. So in * the same call session, RTP timestamp and sequence are * guaranteed to be contigue. */ call_med->rtp_tx_seq_ts_set = 1 | (1 << 1); call_med->rtp_tx_seq = stat.rtp_tx_last_seq; call_med->rtp_tx_ts = stat.rtp_tx_last_ts; } if (pjsua_var.ua_cfg.cb.on_stream_destroyed) { pjsua_var.ua_cfg.cb.on_stream_destroyed(call_med->call->index, strm, call_med->idx); } if (call_med->strm.a.media_port) { if (call_med->strm.a.destroy_port) pjmedia_port_destroy(call_med->strm.a.media_port); call_med->strm.a.media_port = NULL; } pjmedia_stream_destroy(strm); call_med->strm.a.stream = NULL; } pjsua_check_snd_dev_idle(); } /* * DTMF callback from the stream. */ static void dtmf_callback(pjmedia_stream *strm, void *user_data, int digit) { PJ_UNUSED_ARG(strm); pj_log_push_indent(); if (pjsua_var.ua_cfg.cb.on_dtmf_digit2) { pjsua_call_id call_id; pjsua_dtmf_info info; call_id = (pjsua_call_id)(pj_ssize_t)user_data; info.method = PJSUA_DTMF_METHOD_RFC2833; info.digit = digit; (*pjsua_var.ua_cfg.cb.on_dtmf_digit2)(call_id, &info); } else if (pjsua_var.ua_cfg.cb.on_dtmf_digit) { /* For discussions about call mutex protection related to this * callback, please see ticket #460: * http://trac.pjsip.org/repos/ticket/460#comment:4 */ pjsua_call_id call_id; call_id = (pjsua_call_id)(pj_ssize_t)user_data; (*pjsua_var.ua_cfg.cb.on_dtmf_digit)(call_id, digit); } pj_log_pop_indent(); } /* Internal function: update audio channel after SDP negotiation. * Warning: do not use temporary/flip-flop pool, e.g: inv->pool_prov, * for creating stream, etc, as after SDP negotiation and when * the SDP media is not changed, the stream should remain running * while the temporary/flip-flop pool may be released. */ pj_status_t pjsua_aud_channel_update(pjsua_call_media *call_med, pj_pool_t *tmp_pool, pjmedia_stream_info *si, const pjmedia_sdp_session *local_sdp, const pjmedia_sdp_session *remote_sdp) { pjsua_call *call = call_med->call; unsigned strm_idx = call_med->idx; pj_status_t status = PJ_SUCCESS; PJ_UNUSED_ARG(tmp_pool); PJ_UNUSED_ARG(local_sdp); PJ_UNUSED_ARG(remote_sdp); PJ_LOG(4,(THIS_FILE,"Audio channel update..")); pj_log_push_indent(); si->rtcp_sdes_bye_disabled = pjsua_var.media_cfg.no_rtcp_sdes_bye; /* Check if no media is active */ if (local_sdp->media[strm_idx]->desc.port != 0) { /* Optionally, application may modify other stream settings here * (such as jitter buffer parameters, codec ptime, etc.) */ si->jb_init = pjsua_var.media_cfg.jb_init; si->jb_min_pre = pjsua_var.media_cfg.jb_min_pre; si->jb_max_pre = pjsua_var.media_cfg.jb_max_pre; si->jb_max = pjsua_var.media_cfg.jb_max; /* Set SSRC and CNAME */ si->ssrc = call_med->ssrc; si->cname = call->cname; /* Set RTP timestamp & sequence, normally these value are intialized * automatically when stream session created, but for some cases (e.g: * call reinvite, call update) timestamp and sequence need to be kept * contigue. */ si->rtp_ts = call_med->rtp_tx_ts; si->rtp_seq = call_med->rtp_tx_seq; si->rtp_seq_ts_set = call_med->rtp_tx_seq_ts_set; #if defined(PJMEDIA_STREAM_ENABLE_KA) && PJMEDIA_STREAM_ENABLE_KA!=0 /* Enable/disable stream keep-alive and NAT hole punch. */ si->use_ka = pjsua_var.acc[call->acc_id].cfg.use_stream_ka; #endif /* Create session based on session info. */ status = pjmedia_stream_create(pjsua_var.med_endpt, NULL, si, call_med->tp, NULL, &call_med->strm.a.stream); if (status != PJ_SUCCESS) { goto on_return; } /* Start stream */ status = pjmedia_stream_start(call_med->strm.a.stream); if (status != PJ_SUCCESS) { goto on_return; } if (call_med->prev_state == PJSUA_CALL_MEDIA_NONE) pjmedia_stream_send_rtcp_sdes(call_med->strm.a.stream); /* If DTMF callback is installed by application, install our * callback to the session. */ if (pjsua_var.ua_cfg.cb.on_dtmf_digit || pjsua_var.ua_cfg.cb.on_dtmf_digit2) { pjmedia_stream_set_dtmf_callback(call_med->strm.a.stream, &dtmf_callback, (void*)(pj_ssize_t)(call->index)); } /* Get the port interface of the first stream in the session. * We need the port interface to add to the conference bridge. */ pjmedia_stream_get_port(call_med->strm.a.stream, &call_med->strm.a.media_port); /* Notify application about stream creation. * Note: application may modify media_port to point to different * media port */ if (pjsua_var.ua_cfg.cb.on_stream_created2) { pjsua_on_stream_created_param prm; prm.stream = call_med->strm.a.stream; prm.stream_idx = strm_idx; prm.destroy_port = PJ_FALSE; prm.port = call_med->strm.a.media_port; (*pjsua_var.ua_cfg.cb.on_stream_created2)(call->index, &prm); call_med->strm.a.destroy_port = prm.destroy_port; call_med->strm.a.media_port = prm.port; } else if (pjsua_var.ua_cfg.cb.on_stream_created) { (*pjsua_var.ua_cfg.cb.on_stream_created)(call->index, call_med->strm.a.stream, strm_idx, &call_med->strm.a.media_port); } /* * Add the call to conference bridge. */ { char tmp[PJSIP_MAX_URL_SIZE]; pj_str_t port_name; port_name.ptr = tmp; port_name.slen = pjsip_uri_print(PJSIP_URI_IN_REQ_URI, call->inv->dlg->remote.info->uri, tmp, sizeof(tmp)); if (port_name.slen < 1) { port_name = pj_str("call"); } status = pjmedia_conf_add_port(pjsua_var.mconf, call->inv->pool, call_med->strm.a.media_port, &port_name, (unsigned*) &call_med->strm.a.conf_slot); if (status != PJ_SUCCESS) { goto on_return; } } /* Subscribe to stream events */ pjmedia_event_subscribe(NULL, &call_media_on_event, call_med, call_med->strm.a.stream); } on_return: pj_log_pop_indent(); return status; } PJ_DEF(void) pjsua_snd_dev_param_default(pjsua_snd_dev_param *prm) { pj_bzero(prm, sizeof(*prm)); prm->capture_dev = PJMEDIA_AUD_DEFAULT_CAPTURE_DEV; prm->playback_dev = PJMEDIA_AUD_DEFAULT_PLAYBACK_DEV; } PJ_DEF(void) pjsua_conf_connect_param_default(pjsua_conf_connect_param *prm) { pj_bzero(prm, sizeof(*prm)); prm->level = 1.0; } /* * Get maxinum number of conference ports. */ PJ_DEF(unsigned) pjsua_conf_get_max_ports(void) { return pjsua_var.media_cfg.max_media_ports; } /* * Get current number of active ports in the bridge. */ PJ_DEF(unsigned) pjsua_conf_get_active_ports(void) { unsigned ports[PJSUA_MAX_CONF_PORTS]; unsigned count = PJ_ARRAY_SIZE(ports); pj_status_t status; status = pjmedia_conf_enum_ports(pjsua_var.mconf, ports, &count); if (status != PJ_SUCCESS) count = 0; return count; } /* * Enumerate all conference ports. */ PJ_DEF(pj_status_t) pjsua_enum_conf_ports(pjsua_conf_port_id id[], unsigned *count) { return pjmedia_conf_enum_ports(pjsua_var.mconf, (unsigned*)id, count); } /* * Get information about the specified conference port */ PJ_DEF(pj_status_t) pjsua_conf_get_port_info( pjsua_conf_port_id id, pjsua_conf_port_info *info) { pjmedia_conf_port_info cinfo; unsigned i; pj_status_t status; status = pjmedia_conf_get_port_info( pjsua_var.mconf, id, &cinfo); if (status != PJ_SUCCESS) return status; pj_bzero(info, sizeof(*info)); info->slot_id = id; info->name = cinfo.name; pjmedia_format_copy(&info->format, &cinfo.format); info->clock_rate = cinfo.clock_rate; info->channel_count = cinfo.channel_count; info->samples_per_frame = cinfo.samples_per_frame; info->bits_per_sample = cinfo.bits_per_sample; info->tx_level_adj = ((float)cinfo.tx_adj_level) / 128 + 1; info->rx_level_adj = ((float)cinfo.rx_adj_level) / 128 + 1; /* Build array of listeners */ info->listener_cnt = cinfo.listener_cnt; for (i=0; ilisteners[i] = cinfo.listener_slots[i]; } return PJ_SUCCESS; } /* * Add arbitrary media port to PJSUA's conference bridge. */ PJ_DEF(pj_status_t) pjsua_conf_add_port( pj_pool_t *pool, pjmedia_port *port, pjsua_conf_port_id *p_id) { pj_status_t status; status = pjmedia_conf_add_port(pjsua_var.mconf, pool, port, NULL, (unsigned*)p_id); if (status != PJ_SUCCESS) { if (p_id) *p_id = PJSUA_INVALID_ID; } return status; } /* * Remove arbitrary slot from the conference bridge. */ PJ_DEF(pj_status_t) pjsua_conf_remove_port(pjsua_conf_port_id id) { pj_status_t status; status = pjmedia_conf_remove_port(pjsua_var.mconf, (unsigned)id); pjsua_check_snd_dev_idle(); return status; } /* * Establish unidirectional media flow from souce to sink. */ PJ_DEF(pj_status_t) pjsua_conf_connect( pjsua_conf_port_id source, pjsua_conf_port_id sink) { pjsua_conf_connect_param prm; pjsua_conf_connect_param_default(&prm); return pjsua_conf_connect2(source, sink, &prm); } /* * Establish unidirectional media flow from souce to sink, with signal * level adjustment. */ PJ_DEF(pj_status_t) pjsua_conf_connect2( pjsua_conf_port_id source, pjsua_conf_port_id sink, const pjsua_conf_connect_param *prm) { pj_status_t status = PJ_SUCCESS; PJ_LOG(4,(THIS_FILE, "%s connect: %d --> %d", (pjsua_var.is_mswitch ? "Switch" : "Conf"), source, sink)); pj_log_push_indent(); PJSUA_LOCK(); /* If sound device idle timer is active, cancel it first. */ if (pjsua_var.snd_idle_timer.id) { pjsip_endpt_cancel_timer(pjsua_var.endpt, &pjsua_var.snd_idle_timer); pjsua_var.snd_idle_timer.id = PJ_FALSE; } /* For audio switchboard (i.e. APS-Direct): * Check if sound device need to be reopened, i.e: its attributes * (format, clock rate, channel count) must match to peer's. * Note that sound device can be reopened only if it doesn't have * any connection. */ if (pjsua_var.is_mswitch) { pjmedia_conf_port_info port0_info; pjmedia_conf_port_info peer_info; unsigned peer_id; pj_bool_t need_reopen = PJ_FALSE; peer_id = (source!=0)? source : sink; status = pjmedia_conf_get_port_info(pjsua_var.mconf, peer_id, &peer_info); pj_assert(status == PJ_SUCCESS); status = pjmedia_conf_get_port_info(pjsua_var.mconf, 0, &port0_info); pj_assert(status == PJ_SUCCESS); /* Check if sound device is instantiated. */ need_reopen = (pjsua_var.snd_port==NULL && pjsua_var.null_snd==NULL && !pjsua_var.no_snd); /* Check if sound device need to reopen because it needs to modify * settings to match its peer. Sound device must be idle in this case * though. */ if (!need_reopen && port0_info.listener_cnt==0 && port0_info.transmitter_cnt==0) { need_reopen = (peer_info.format.id != port0_info.format.id || peer_info.format.det.aud.avg_bps != port0_info.format.det.aud.avg_bps || peer_info.clock_rate != port0_info.clock_rate || peer_info.channel_count!=port0_info.channel_count); } if (need_reopen) { if (pjsua_var.cap_dev != PJSUA_SND_NULL_DEV) { pjmedia_snd_port_param param; pjmedia_snd_port_param_default(¶m); param.ec_options = pjsua_var.media_cfg.ec_options; /* Create parameter based on peer info */ status = create_aud_param(¶m.base, pjsua_var.cap_dev, pjsua_var.play_dev, peer_info.clock_rate, peer_info.channel_count, peer_info.samples_per_frame, peer_info.bits_per_sample); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Error opening sound device", status); goto on_return; } /* And peer format */ if (peer_info.format.id != PJMEDIA_FORMAT_PCM) { param.base.flags |= PJMEDIA_AUD_DEV_CAP_EXT_FORMAT; param.base.ext_fmt = peer_info.format; } param.options = 0; status = open_snd_dev(¶m); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Error opening sound device", status); goto on_return; } } else { /* Null-audio */ status = pjsua_set_snd_dev(pjsua_var.cap_dev, pjsua_var.play_dev); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Error opening sound device", status); goto on_return; } } } else if (pjsua_var.no_snd) { if (!pjsua_var.snd_is_on) { pjsua_var.snd_is_on = PJ_TRUE; /* Notify app */ if (pjsua_var.ua_cfg.cb.on_snd_dev_operation) { (*pjsua_var.ua_cfg.cb.on_snd_dev_operation)(1); } } } } else { /* The bridge version */ /* Create sound port if none is instantiated */ if (pjsua_var.snd_port==NULL && pjsua_var.null_snd==NULL && !pjsua_var.no_snd) { status = pjsua_set_snd_dev(pjsua_var.cap_dev, pjsua_var.play_dev); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Error opening sound device", status); goto on_return; } } else if (pjsua_var.no_snd && !pjsua_var.snd_is_on) { pjsua_var.snd_is_on = PJ_TRUE; /* Notify app */ if (pjsua_var.ua_cfg.cb.on_snd_dev_operation) { (*pjsua_var.ua_cfg.cb.on_snd_dev_operation)(1); } } } on_return: PJSUA_UNLOCK(); if (status == PJ_SUCCESS) { pjsua_conf_connect_param cc_param; if (!prm) pjsua_conf_connect_param_default(&cc_param); else pj_memcpy(&cc_param, prm, sizeof(cc_param)); status = pjmedia_conf_connect_port(pjsua_var.mconf, source, sink, (int)((cc_param.level-1) * 128)); } pj_log_pop_indent(); return status; } /* * Disconnect media flow from the source to destination port. */ PJ_DEF(pj_status_t) pjsua_conf_disconnect( pjsua_conf_port_id source, pjsua_conf_port_id sink) { pj_status_t status; PJ_LOG(4,(THIS_FILE, "%s disconnect: %d -x- %d", (pjsua_var.is_mswitch ? "Switch" : "Conf"), source, sink)); pj_log_push_indent(); status = pjmedia_conf_disconnect_port(pjsua_var.mconf, source, sink); pjsua_check_snd_dev_idle(); pj_log_pop_indent(); return status; } /* * Adjust the signal level to be transmitted from the bridge to the * specified port by making it louder or quieter. */ PJ_DEF(pj_status_t) pjsua_conf_adjust_tx_level(pjsua_conf_port_id slot, float level) { return pjmedia_conf_adjust_tx_level(pjsua_var.mconf, slot, (int)((level-1) * 128)); } /* * Adjust the signal level to be received from the specified port (to * the bridge) by making it louder or quieter. */ PJ_DEF(pj_status_t) pjsua_conf_adjust_rx_level(pjsua_conf_port_id slot, float level) { return pjmedia_conf_adjust_rx_level(pjsua_var.mconf, slot, (int)((level-1) * 128)); } /* * Get last signal level transmitted to or received from the specified port. */ PJ_DEF(pj_status_t) pjsua_conf_get_signal_level(pjsua_conf_port_id slot, unsigned *tx_level, unsigned *rx_level) { return pjmedia_conf_get_signal_level(pjsua_var.mconf, slot, tx_level, rx_level); } /***************************************************************************** * File player. */ static char* get_basename(const char *path, unsigned len) { char *p = ((char*)path) + len; if (len==0) return p; for (--p; p!=path && *p!='/' && *p!='\\'; ) --p; return (p==path) ? p : p+1; } /* * Create a file player, and automatically connect this player to * the conference bridge. */ PJ_DEF(pj_status_t) pjsua_player_create( const pj_str_t *filename, unsigned options, pjsua_player_id *p_id) { unsigned slot, file_id; char path[PJ_MAXPATH]; pj_pool_t *pool = NULL; pjmedia_port *port; pj_status_t status = PJ_SUCCESS; if (pjsua_var.player_cnt >= PJ_ARRAY_SIZE(pjsua_var.player)) return PJ_ETOOMANY; PJ_LOG(4,(THIS_FILE, "Creating file player: %.*s..", (int)filename->slen, filename->ptr)); pj_log_push_indent(); PJSUA_LOCK(); for (file_id=0; file_idptr, filename->slen); path[filename->slen] = '\0'; pool = pjsua_pool_create(get_basename(path, (unsigned)filename->slen), 1000, 1000); if (!pool) { status = PJ_ENOMEM; goto on_error; } status = pjmedia_wav_player_port_create( pool, path, pjsua_var.mconf_cfg.samples_per_frame * 1000 / pjsua_var.media_cfg.channel_count / pjsua_var.media_cfg.clock_rate, options, 0, &port); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to open file for playback", status); goto on_error; } status = pjmedia_conf_add_port(pjsua_var.mconf, pool, port, filename, &slot); if (status != PJ_SUCCESS) { pjmedia_port_destroy(port); pjsua_perror(THIS_FILE, "Unable to add file to conference bridge", status); goto on_error; } pjsua_var.player[file_id].type = 0; pjsua_var.player[file_id].pool = pool; pjsua_var.player[file_id].port = port; pjsua_var.player[file_id].slot = slot; if (p_id) *p_id = file_id; ++pjsua_var.player_cnt; PJSUA_UNLOCK(); PJ_LOG(4,(THIS_FILE, "Player created, id=%d, slot=%d", file_id, slot)); pj_log_pop_indent(); return PJ_SUCCESS; on_error: PJSUA_UNLOCK(); if (pool) pj_pool_release(pool); pj_log_pop_indent(); return status; } /* * Create a file playlist media port, and automatically add the port * to the conference bridge. */ PJ_DEF(pj_status_t) pjsua_playlist_create( const pj_str_t file_names[], unsigned file_count, const pj_str_t *label, unsigned options, pjsua_player_id *p_id) { unsigned slot, file_id, ptime; pj_pool_t *pool = NULL; pjmedia_port *port; pj_status_t status = PJ_SUCCESS; if (pjsua_var.player_cnt >= PJ_ARRAY_SIZE(pjsua_var.player)) return PJ_ETOOMANY; PJ_LOG(4,(THIS_FILE, "Creating playlist with %d file(s)..", file_count)); pj_log_push_indent(); PJSUA_LOCK(); for (file_id=0; file_idinfo.name, &slot); if (status != PJ_SUCCESS) { pjmedia_port_destroy(port); pjsua_perror(THIS_FILE, "Unable to add port", status); goto on_error; } pjsua_var.player[file_id].type = 1; pjsua_var.player[file_id].pool = pool; pjsua_var.player[file_id].port = port; pjsua_var.player[file_id].slot = slot; if (p_id) *p_id = file_id; ++pjsua_var.player_cnt; PJSUA_UNLOCK(); PJ_LOG(4,(THIS_FILE, "Playlist created, id=%d, slot=%d", file_id, slot)); pj_log_pop_indent(); return PJ_SUCCESS; on_error: PJSUA_UNLOCK(); if (pool) pj_pool_release(pool); pj_log_pop_indent(); return status; } /* * Get conference port ID associated with player. */ PJ_DEF(pjsua_conf_port_id) pjsua_player_get_conf_port(pjsua_player_id id) { PJ_ASSERT_RETURN(id>=0&&id<(int)PJ_ARRAY_SIZE(pjsua_var.player),PJ_EINVAL); PJ_ASSERT_RETURN(pjsua_var.player[id].port != NULL, PJ_EINVAL); return pjsua_var.player[id].slot; } /* * Get the media port for the player. */ PJ_DEF(pj_status_t) pjsua_player_get_port( pjsua_player_id id, pjmedia_port **p_port) { PJ_ASSERT_RETURN(id>=0&&id<(int)PJ_ARRAY_SIZE(pjsua_var.player),PJ_EINVAL); PJ_ASSERT_RETURN(pjsua_var.player[id].port != NULL, PJ_EINVAL); PJ_ASSERT_RETURN(p_port != NULL, PJ_EINVAL); *p_port = pjsua_var.player[id].port; return PJ_SUCCESS; } /* * Get player info. */ PJ_DEF(pj_status_t) pjsua_player_get_info(pjsua_player_id id, pjmedia_wav_player_info *info) { PJ_ASSERT_RETURN(id>=0&&id<(int)PJ_ARRAY_SIZE(pjsua_var.player), -PJ_EINVAL); PJ_ASSERT_RETURN(pjsua_var.player[id].port != NULL, PJ_EINVAL); PJ_ASSERT_RETURN(pjsua_var.player[id].type == 0, PJ_EINVAL); return pjmedia_wav_player_get_info(pjsua_var.player[id].port, info); } /* * Get playback position. */ PJ_DEF(pj_ssize_t) pjsua_player_get_pos( pjsua_player_id id ) { pj_ssize_t pos_bytes; pjmedia_wav_player_info info; pj_status_t status; PJ_ASSERT_RETURN(id>=0&&id<(int)PJ_ARRAY_SIZE(pjsua_var.player), -PJ_EINVAL); PJ_ASSERT_RETURN(pjsua_var.player[id].port != NULL, -PJ_EINVAL); PJ_ASSERT_RETURN(pjsua_var.player[id].type == 0, -PJ_EINVAL); pos_bytes = pjmedia_wav_player_port_get_pos(pjsua_var.player[id].port); if (pos_bytes < 0) return pos_bytes; status = pjmedia_wav_player_get_info(pjsua_var.player[id].port, &info); if (status != PJ_SUCCESS) return -status; return pos_bytes / (info.payload_bits_per_sample / 8); } /* * Set playback position. */ PJ_DEF(pj_status_t) pjsua_player_set_pos( pjsua_player_id id, pj_uint32_t samples) { pjmedia_wav_player_info info; pj_uint32_t pos_bytes; pj_status_t status; PJ_ASSERT_RETURN(id>=0&&id<(int)PJ_ARRAY_SIZE(pjsua_var.player),PJ_EINVAL); PJ_ASSERT_RETURN(pjsua_var.player[id].port != NULL, PJ_EINVAL); PJ_ASSERT_RETURN(pjsua_var.player[id].type == 0, PJ_EINVAL); status = pjmedia_wav_player_get_info(pjsua_var.player[id].port, &info); if (status != PJ_SUCCESS) return status; pos_bytes = samples * (info.payload_bits_per_sample / 8); return pjmedia_wav_player_port_set_pos(pjsua_var.player[id].port, pos_bytes); } /* * Close the file, remove the player from the bridge, and free * resources associated with the file player. */ PJ_DEF(pj_status_t) pjsua_player_destroy(pjsua_player_id id) { PJ_ASSERT_RETURN(id>=0&&id<(int)PJ_ARRAY_SIZE(pjsua_var.player),PJ_EINVAL); PJ_ASSERT_RETURN(pjsua_var.player[id].port != NULL, PJ_EINVAL); PJ_LOG(4,(THIS_FILE, "Destroying player %d..", id)); pj_log_push_indent(); PJSUA_LOCK(); if (pjsua_var.player[id].port) { pjsua_conf_remove_port(pjsua_var.player[id].slot); pjmedia_port_destroy(pjsua_var.player[id].port); pjsua_var.player[id].port = NULL; pjsua_var.player[id].slot = 0xFFFF; pj_pool_release(pjsua_var.player[id].pool); pjsua_var.player[id].pool = NULL; pjsua_var.player_cnt--; } PJSUA_UNLOCK(); pj_log_pop_indent(); return PJ_SUCCESS; } /***************************************************************************** * File recorder. */ /* * Create a file recorder, and automatically connect this recorder to * the conference bridge. */ PJ_DEF(pj_status_t) pjsua_recorder_create( const pj_str_t *filename, unsigned enc_type, void *enc_param, pj_ssize_t max_size, unsigned options, pjsua_recorder_id *p_id) { enum Format { FMT_UNKNOWN, FMT_WAV, FMT_MP3, }; unsigned slot, file_id; char path[PJ_MAXPATH]; pj_str_t ext; int file_format; pj_pool_t *pool = NULL; pjmedia_port *port; pj_status_t status = PJ_SUCCESS; /* Filename must present */ PJ_ASSERT_RETURN(filename != NULL, PJ_EINVAL); /* Don't support max_size at present */ PJ_ASSERT_RETURN(max_size == 0 || max_size == -1, PJ_EINVAL); /* Don't support encoding type at present */ PJ_ASSERT_RETURN(enc_type == 0, PJ_EINVAL); PJ_LOG(4,(THIS_FILE, "Creating recorder %.*s..", (int)filename->slen, filename->ptr)); pj_log_push_indent(); if (pjsua_var.rec_cnt >= PJ_ARRAY_SIZE(pjsua_var.recorder)) { pj_log_pop_indent(); return PJ_ETOOMANY; } /* Determine the file format */ ext.ptr = filename->ptr + filename->slen - 4; ext.slen = 4; if (pj_stricmp2(&ext, ".wav") == 0) file_format = FMT_WAV; else if (pj_stricmp2(&ext, ".mp3") == 0) file_format = FMT_MP3; else { PJ_LOG(1,(THIS_FILE, "pjsua_recorder_create() error: unable to " "determine file format for %.*s", (int)filename->slen, filename->ptr)); pj_log_pop_indent(); return PJ_ENOTSUP; } PJSUA_LOCK(); for (file_id=0; file_idptr, filename->slen); path[filename->slen] = '\0'; pool = pjsua_pool_create(get_basename(path, (unsigned)filename->slen), 1000, 1000); if (!pool) { status = PJ_ENOMEM; goto on_return; } if (file_format == FMT_WAV) { status = pjmedia_wav_writer_port_create(pool, path, pjsua_var.media_cfg.clock_rate, pjsua_var.mconf_cfg.channel_count, pjsua_var.mconf_cfg.samples_per_frame, pjsua_var.mconf_cfg.bits_per_sample, options, 0, &port); } else { PJ_UNUSED_ARG(enc_param); port = NULL; status = PJ_ENOTSUP; } if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to open file for recording", status); goto on_return; } status = pjmedia_conf_add_port(pjsua_var.mconf, pool, port, filename, &slot); if (status != PJ_SUCCESS) { pjmedia_port_destroy(port); goto on_return; } pjsua_var.recorder[file_id].port = port; pjsua_var.recorder[file_id].slot = slot; pjsua_var.recorder[file_id].pool = pool; if (p_id) *p_id = file_id; ++pjsua_var.rec_cnt; PJSUA_UNLOCK(); PJ_LOG(4,(THIS_FILE, "Recorder created, id=%d, slot=%d", file_id, slot)); pj_log_pop_indent(); return PJ_SUCCESS; on_return: PJSUA_UNLOCK(); if (pool) pj_pool_release(pool); pj_log_pop_indent(); return status; } /* * Get conference port associated with recorder. */ PJ_DEF(pjsua_conf_port_id) pjsua_recorder_get_conf_port(pjsua_recorder_id id) { PJ_ASSERT_RETURN(id>=0 && id<(int)PJ_ARRAY_SIZE(pjsua_var.recorder), PJ_EINVAL); PJ_ASSERT_RETURN(pjsua_var.recorder[id].port != NULL, PJ_EINVAL); return pjsua_var.recorder[id].slot; } /* * Get the media port for the recorder. */ PJ_DEF(pj_status_t) pjsua_recorder_get_port( pjsua_recorder_id id, pjmedia_port **p_port) { PJ_ASSERT_RETURN(id>=0 && id<(int)PJ_ARRAY_SIZE(pjsua_var.recorder), PJ_EINVAL); PJ_ASSERT_RETURN(pjsua_var.recorder[id].port != NULL, PJ_EINVAL); PJ_ASSERT_RETURN(p_port != NULL, PJ_EINVAL); *p_port = pjsua_var.recorder[id].port; return PJ_SUCCESS; } /* * Destroy recorder (this will complete recording). */ PJ_DEF(pj_status_t) pjsua_recorder_destroy(pjsua_recorder_id id) { PJ_ASSERT_RETURN(id>=0 && id<(int)PJ_ARRAY_SIZE(pjsua_var.recorder), PJ_EINVAL); PJ_ASSERT_RETURN(pjsua_var.recorder[id].port != NULL, PJ_EINVAL); PJ_LOG(4,(THIS_FILE, "Destroying recorder %d..", id)); pj_log_push_indent(); PJSUA_LOCK(); if (pjsua_var.recorder[id].port) { pjsua_conf_remove_port(pjsua_var.recorder[id].slot); pjmedia_port_destroy(pjsua_var.recorder[id].port); pjsua_var.recorder[id].port = NULL; pjsua_var.recorder[id].slot = 0xFFFF; pj_pool_release(pjsua_var.recorder[id].pool); pjsua_var.recorder[id].pool = NULL; pjsua_var.rec_cnt--; } PJSUA_UNLOCK(); pj_log_pop_indent(); return PJ_SUCCESS; } /***************************************************************************** * Sound devices. */ /* * Enum sound devices. */ PJ_DEF(pj_status_t) pjsua_enum_aud_devs( pjmedia_aud_dev_info info[], unsigned *count) { unsigned i, dev_count; dev_count = pjmedia_aud_dev_count(); if (dev_count > *count) dev_count = *count; for (i=0; i *count) dev_count = *count; pj_bzero(info, dev_count * sizeof(pjmedia_snd_dev_info)); for (i=0; idir = PJMEDIA_DIR_CAPTURE_PLAYBACK; param->rec_id = capture_dev; param->play_id = playback_dev; param->clock_rate = clock_rate; param->channel_count = channel_count; param->samples_per_frame = samples_per_frame; param->bits_per_sample = bits_per_sample; /* Update the setting with user preference */ #define update_param(cap, field) \ if (pjsua_var.aud_param.flags & cap) { \ param->flags |= cap; \ param->field = pjsua_var.aud_param.field; \ } update_param( PJMEDIA_AUD_DEV_CAP_INPUT_VOLUME_SETTING, input_vol); update_param( PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING, output_vol); update_param( PJMEDIA_AUD_DEV_CAP_INPUT_ROUTE, input_route); update_param( PJMEDIA_AUD_DEV_CAP_OUTPUT_ROUTE, output_route); #undef update_param /* Latency settings */ param->flags |= (PJMEDIA_AUD_DEV_CAP_INPUT_LATENCY | PJMEDIA_AUD_DEV_CAP_OUTPUT_LATENCY); param->input_latency_ms = pjsua_var.media_cfg.snd_rec_latency; param->output_latency_ms = pjsua_var.media_cfg.snd_play_latency; /* EC settings */ if (pjsua_var.media_cfg.ec_tail_len) { param->flags |= (PJMEDIA_AUD_DEV_CAP_EC | PJMEDIA_AUD_DEV_CAP_EC_TAIL); param->ec_enabled = PJ_TRUE; param->ec_tail_ms = pjsua_var.media_cfg.ec_tail_len; } else { param->flags &= ~(PJMEDIA_AUD_DEV_CAP_EC|PJMEDIA_AUD_DEV_CAP_EC_TAIL); } /* VAD settings */ if (pjsua_var.media_cfg.no_vad) { param->flags &= ~PJMEDIA_AUD_DEV_CAP_VAD; } else { param->flags |= PJMEDIA_AUD_DEV_CAP_VAD; param->vad_enabled = PJ_TRUE; } return PJ_SUCCESS; } /* Internal: the first time the audio device is opened (during app * startup), retrieve the audio settings such as volume level * so that aud_get_settings() will work. */ static pj_status_t update_initial_aud_param() { pjmedia_aud_stream *strm; pjmedia_aud_param param; pj_status_t status; PJ_ASSERT_RETURN(pjsua_var.snd_port != NULL, PJ_EBUG); strm = pjmedia_snd_port_get_snd_stream(pjsua_var.snd_port); status = pjmedia_aud_stream_get_param(strm, ¶m); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Error audio stream " "device parameters", status); return status; } #define update_saved_param(cap, field) \ if (param.flags & cap) { \ pjsua_var.aud_param.flags |= cap; \ pjsua_var.aud_param.field = param.field; \ } update_saved_param(PJMEDIA_AUD_DEV_CAP_INPUT_VOLUME_SETTING, input_vol); update_saved_param(PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING, output_vol); update_saved_param(PJMEDIA_AUD_DEV_CAP_INPUT_ROUTE, input_route); update_saved_param(PJMEDIA_AUD_DEV_CAP_OUTPUT_ROUTE, output_route); #undef update_saved_param return PJ_SUCCESS; } /* Get format name */ static const char *get_fmt_name(pj_uint32_t id) { static char name[8]; if (id == PJMEDIA_FORMAT_L16) return "PCM"; pj_memcpy(name, &id, 4); name[4] = '\0'; return name; } static pj_status_t on_aud_prev_play_frame(void *user_data, pjmedia_frame *frame) { PJ_UNUSED_ARG(user_data); (*pjsua_var.media_cfg.on_aud_prev_play_frame)(frame); return PJ_SUCCESS; } static pj_status_t on_aud_prev_rec_frame(void *user_data, pjmedia_frame *frame) { PJ_UNUSED_ARG(user_data); (*pjsua_var.media_cfg.on_aud_prev_rec_frame)(frame); return PJ_SUCCESS; } /* Open sound device with the setting. */ static pj_status_t open_snd_dev(pjmedia_snd_port_param *param) { pjmedia_port *conf_port; pj_status_t status; pj_bool_t speaker_only = (pjsua_var.snd_mode & PJSUA_SND_DEV_SPEAKER_ONLY); PJ_ASSERT_RETURN(param, PJ_EINVAL); /* Check if NULL sound device is used */ if (PJSUA_SND_NULL_DEV==param->base.rec_id || PJSUA_SND_NULL_DEV==param->base.play_id) { return pjsua_set_null_snd_dev(); } /* Close existing sound port */ close_snd_dev(); /* Save the device IDs */ pjsua_var.cap_dev = param->base.rec_id; pjsua_var.play_dev = param->base.play_id; /* Notify app */ if (pjsua_var.ua_cfg.cb.on_snd_dev_operation) { (*pjsua_var.ua_cfg.cb.on_snd_dev_operation)(1); } /* Create memory pool for sound device. */ pjsua_var.snd_pool = pjsua_pool_create("pjsua_snd", 4000, 4000); PJ_ASSERT_RETURN(pjsua_var.snd_pool, PJ_ENOMEM); /* Setup preview callbacks, if configured */ if (pjsua_var.media_cfg.on_aud_prev_play_frame) param->on_play_frame = &on_aud_prev_play_frame; if (pjsua_var.media_cfg.on_aud_prev_rec_frame) param->on_rec_frame = &on_aud_prev_rec_frame; PJ_LOG(4,(THIS_FILE, "Opening sound device (%s) %s@%d/%d/%dms", speaker_only?"speaker only":"speaker + mic", get_fmt_name(param->base.ext_fmt.id), param->base.clock_rate, param->base.channel_count, param->base.samples_per_frame / param->base.channel_count * 1000 / param->base.clock_rate)); pj_log_push_indent(); if (speaker_only) { status = pjmedia_snd_port_create_player(pjsua_var.snd_pool, -1, param->base.clock_rate, param->base.channel_count, param->base.samples_per_frame, param->base.bits_per_sample, 0, &pjsua_var.snd_port); } else { status = pjmedia_snd_port_create2(pjsua_var.snd_pool, param, &pjsua_var.snd_port); } if (status != PJ_SUCCESS) goto on_error; /* Get the port0 of the conference bridge. */ conf_port = pjmedia_conf_get_master_port(pjsua_var.mconf); pj_assert(conf_port != NULL); /* For conference bridge, resample if necessary if the bridge's * clock rate is different than the sound device's clock rate. */ if (!pjsua_var.is_mswitch && param->base.ext_fmt.id == PJMEDIA_FORMAT_PCM && PJMEDIA_PIA_SRATE(&conf_port->info) != param->base.clock_rate) { pjmedia_port *resample_port; unsigned resample_opt = 0; if (pjsua_var.media_cfg.quality >= 3 && pjsua_var.media_cfg.quality <= 4) { resample_opt |= PJMEDIA_RESAMPLE_USE_SMALL_FILTER; } else if (pjsua_var.media_cfg.quality < 3) { resample_opt |= PJMEDIA_RESAMPLE_USE_LINEAR; } status = pjmedia_resample_port_create(pjsua_var.snd_pool, conf_port, param->base.clock_rate, resample_opt, &resample_port); if (status != PJ_SUCCESS) { char errmsg[PJ_ERR_MSG_SIZE]; pj_strerror(status, errmsg, sizeof(errmsg)); PJ_LOG(4, (THIS_FILE, "Error creating resample port: %s", errmsg)); close_snd_dev(); goto on_error; } conf_port = resample_port; } /* Otherwise for audio switchboard, the switch's port0 setting is * derived from the sound device setting, so update the setting. */ if (pjsua_var.is_mswitch) { if (param->base.flags & PJMEDIA_AUD_DEV_CAP_EXT_FORMAT) { conf_port->info.fmt = param->base.ext_fmt; } else { unsigned bps, ptime_usec; bps = param->base.clock_rate * param->base.bits_per_sample; ptime_usec = param->base.samples_per_frame / param->base.channel_count * 1000000 / param->base.clock_rate; pjmedia_format_init_audio(&conf_port->info.fmt, PJMEDIA_FORMAT_PCM, param->base.clock_rate, param->base.channel_count, param->base.bits_per_sample, ptime_usec, bps, bps); } } /* Connect sound port to the bridge */ status = pjmedia_snd_port_connect(pjsua_var.snd_port, conf_port ); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to connect conference port to " "sound device", status); pjmedia_snd_port_destroy(pjsua_var.snd_port); pjsua_var.snd_port = NULL; goto on_error; } /* Update sound device name. */ { pjmedia_aud_dev_info rec_info; pjmedia_aud_stream *strm; pjmedia_aud_param si; pj_str_t tmp; strm = pjmedia_snd_port_get_snd_stream(pjsua_var.snd_port); status = pjmedia_aud_stream_get_param(strm, &si); if (status == PJ_SUCCESS) status = pjmedia_aud_dev_get_info(si.rec_id, &rec_info); if (status==PJ_SUCCESS) { if (param->base.clock_rate != pjsua_var.media_cfg.clock_rate) { char tmp_buf[128]; int tmp_buf_len; tmp_buf_len = pj_ansi_snprintf(tmp_buf, sizeof(tmp_buf), "%s (%dKHz)", rec_info.name, param->base.clock_rate/1000); if (tmp_buf_len < 1 || tmp_buf_len >= (int)sizeof(tmp_buf)) tmp_buf_len = sizeof(tmp_buf) - 1; pj_strset(&tmp, tmp_buf, tmp_buf_len); pjmedia_conf_set_port0_name(pjsua_var.mconf, &tmp); } else { pjmedia_conf_set_port0_name(pjsua_var.mconf, pj_cstr(&tmp, rec_info.name)); } } /* Any error is not major, let it through */ status = PJ_SUCCESS; } /* If this is the first time the audio device is open, retrieve some * settings from the device (such as volume settings) so that the * pjsua_snd_get_setting() work. */ if (pjsua_var.aud_open_cnt == 0) { update_initial_aud_param(); ++pjsua_var.aud_open_cnt; } pjsua_var.snd_is_on = PJ_TRUE; /* Subscribe to audio device events */ pjmedia_event_subscribe(NULL, &on_media_event, NULL, pjmedia_snd_port_get_snd_stream(pjsua_var.snd_port)); pj_log_pop_indent(); return PJ_SUCCESS; on_error: pj_log_pop_indent(); return status; } /* Close existing sound device */ static void close_snd_dev(void) { pj_log_push_indent(); /* Notify app */ if (pjsua_var.snd_is_on && pjsua_var.ua_cfg.cb.on_snd_dev_operation) { (*pjsua_var.ua_cfg.cb.on_snd_dev_operation)(0); } /* Close sound device */ if (pjsua_var.snd_port) { pjmedia_aud_dev_info cap_info, play_info; pjmedia_aud_stream *strm; pjmedia_aud_param param; strm = pjmedia_snd_port_get_snd_stream(pjsua_var.snd_port); pjmedia_aud_stream_get_param(strm, ¶m); if (pjmedia_aud_dev_get_info(param.rec_id, &cap_info) != PJ_SUCCESS) cap_info.name[0] = '\0'; if (pjmedia_aud_dev_get_info(param.play_id, &play_info) != PJ_SUCCESS) play_info.name[0] = '\0'; PJ_LOG(4,(THIS_FILE, "Closing %s sound playback device and " "%s sound capture device", play_info.name, cap_info.name)); /* Unsubscribe from audio device events */ pjmedia_event_unsubscribe(NULL, &on_media_event, NULL, strm); pjmedia_snd_port_disconnect(pjsua_var.snd_port); pjmedia_snd_port_destroy(pjsua_var.snd_port); pjsua_var.snd_port = NULL; } /* Close null sound device */ if (pjsua_var.null_snd) { PJ_LOG(4,(THIS_FILE, "Closing null sound device..")); pjmedia_master_port_destroy(pjsua_var.null_snd, PJ_FALSE); pjsua_var.null_snd = NULL; } if (pjsua_var.snd_pool) pj_pool_release(pjsua_var.snd_pool); pjsua_var.snd_pool = NULL; pjsua_var.snd_is_on = PJ_FALSE; pj_log_pop_indent(); } PJ_DEF(pj_status_t) pjsua_set_snd_dev(int capture_dev, int playback_dev) { pjsua_snd_dev_param param; pjsua_snd_dev_param_default(¶m); param.capture_dev = capture_dev; param.playback_dev = playback_dev; /* Always open the sound device. */ param.mode = 0; return pjsua_set_snd_dev2(¶m); } /* * Select or change sound device. Application may call this function at * any time to replace current sound device. */ PJ_DEF(pj_status_t) pjsua_set_snd_dev2(pjsua_snd_dev_param *snd_param) { unsigned alt_cr_cnt = 1; unsigned alt_cr[] = {0, 44100, 48000, 32000, 16000, 8000}; unsigned i; pj_status_t status = -1; unsigned orig_snd_dev_mode = pjsua_var.snd_mode; pj_bool_t no_change = (pjsua_var.snd_is_on || (!pjsua_var.snd_is_on && (snd_param->mode & PJSUA_SND_DEV_NO_IMMEDIATE_OPEN))); PJ_LOG(4,(THIS_FILE, "Set sound device: capture=%d, playback=%d", snd_param->capture_dev, snd_param->playback_dev)); pj_log_push_indent(); PJSUA_LOCK(); if (pjsua_var.cap_dev == snd_param->capture_dev && pjsua_var.play_dev == snd_param->playback_dev && pjsua_var.snd_mode == snd_param->mode && !pjsua_var.no_snd && no_change) { PJ_LOG(4, (THIS_FILE, "No changes in capture and playback devices")); PJSUA_UNLOCK(); pj_log_pop_indent(); return PJ_SUCCESS; } /* Null-sound */ if (snd_param->capture_dev == PJSUA_SND_NULL_DEV && snd_param->playback_dev == PJSUA_SND_NULL_DEV) { PJSUA_UNLOCK(); status = pjsua_set_null_snd_dev(); pj_log_pop_indent(); return status; } pjsua_var.snd_mode = snd_param->mode; if (!pjsua_var.no_snd && !pjsua_var.snd_is_on && (snd_param->mode & PJSUA_SND_DEV_NO_IMMEDIATE_OPEN)) { pjsua_var.cap_dev = snd_param->capture_dev; pjsua_var.play_dev = snd_param->playback_dev; PJSUA_UNLOCK(); pj_log_pop_indent(); return PJ_SUCCESS; } /* Set default clock rate */ alt_cr[0] = pjsua_var.media_cfg.snd_clock_rate; if (alt_cr[0] == 0) alt_cr[0] = pjsua_var.media_cfg.clock_rate; /* Allow retrying of different clock rate if we're using conference * bridge (meaning audio format is always PCM), otherwise lock on * to one clock rate. */ if (pjsua_var.is_mswitch) { alt_cr_cnt = 1; } else { alt_cr_cnt = PJ_ARRAY_SIZE(alt_cr); } /* Attempts to open the sound device with different clock rates */ for (i=0; icapture_dev, snd_param->playback_dev, alt_cr[i], pjsua_var.media_cfg.channel_count, samples_per_frame, 16); if (status != PJ_SUCCESS) goto on_error; /* Open! */ param.options = 0; status = open_snd_dev(¶m); if (status == PJ_SUCCESS) break; } if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to open sound device", status); goto on_error; } pjsua_var.no_snd = PJ_FALSE; pjsua_var.snd_is_on = PJ_TRUE; PJSUA_UNLOCK(); pj_log_pop_indent(); return PJ_SUCCESS; on_error: pjsua_var.snd_mode = orig_snd_dev_mode; PJSUA_UNLOCK(); pj_log_pop_indent(); return status; } /* * Get currently active sound devices. If sound devices has not been created * (for example when pjsua_start() is not called), it is possible that * the function returns PJ_SUCCESS with -1 as device IDs. */ PJ_DEF(pj_status_t) pjsua_get_snd_dev(int *capture_dev, int *playback_dev) { PJSUA_LOCK(); if (capture_dev) { *capture_dev = pjsua_var.cap_dev; } if (playback_dev) { *playback_dev = pjsua_var.play_dev; } PJSUA_UNLOCK(); return PJ_SUCCESS; } /* * Use null sound device. */ PJ_DEF(pj_status_t) pjsua_set_null_snd_dev(void) { pjmedia_port *conf_port; pj_status_t status; PJ_LOG(4,(THIS_FILE, "Setting null sound device..")); pj_log_push_indent(); PJSUA_LOCK(); /* Close existing sound device */ close_snd_dev(); pjsua_var.cap_dev = PJSUA_SND_NULL_DEV; pjsua_var.play_dev = PJSUA_SND_NULL_DEV; /* Notify app */ if (pjsua_var.ua_cfg.cb.on_snd_dev_operation) { (*pjsua_var.ua_cfg.cb.on_snd_dev_operation)(1); } /* Create memory pool for sound device. */ pjsua_var.snd_pool = pjsua_pool_create("pjsua_snd", 4000, 4000); PJ_ASSERT_RETURN(pjsua_var.snd_pool, PJ_ENOMEM); PJ_LOG(4,(THIS_FILE, "Opening null sound device..")); /* Get the port0 of the conference bridge. */ conf_port = pjmedia_conf_get_master_port(pjsua_var.mconf); pj_assert(conf_port != NULL); /* Create master port, connecting port0 of the conference bridge to * a null port. */ status = pjmedia_master_port_create(pjsua_var.snd_pool, pjsua_var.null_port, conf_port, 0, &pjsua_var.null_snd); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to create null sound device", status); PJSUA_UNLOCK(); pj_log_pop_indent(); return status; } /* Start the master port */ status = pjmedia_master_port_start(pjsua_var.null_snd); PJ_ASSERT_RETURN(status == PJ_SUCCESS, status); pjsua_var.no_snd = PJ_FALSE; pjsua_var.snd_is_on = PJ_TRUE; PJSUA_UNLOCK(); pj_log_pop_indent(); return PJ_SUCCESS; } /* * Use no device! */ PJ_DEF(pjmedia_port*) pjsua_set_no_snd_dev(void) { PJSUA_LOCK(); /* Close existing sound device */ close_snd_dev(); pjsua_var.no_snd = PJ_TRUE; pjsua_var.cap_dev = PJSUA_SND_NO_DEV; pjsua_var.play_dev = PJSUA_SND_NO_DEV; PJSUA_UNLOCK(); return pjmedia_conf_get_master_port(pjsua_var.mconf); } /* * Configure the AEC settings of the sound port. */ PJ_DEF(pj_status_t) pjsua_set_ec(unsigned tail_ms, unsigned options) { pj_status_t status = PJ_SUCCESS; PJSUA_LOCK(); pjsua_var.media_cfg.ec_tail_len = tail_ms; pjsua_var.media_cfg.ec_options = options; if (pjsua_var.snd_port) status = pjmedia_snd_port_set_ec(pjsua_var.snd_port, pjsua_var.pool, tail_ms, options); PJSUA_UNLOCK(); return status; } /* * Get current AEC tail length. */ PJ_DEF(pj_status_t) pjsua_get_ec_tail(unsigned *p_tail_ms) { *p_tail_ms = pjsua_var.media_cfg.ec_tail_len; return PJ_SUCCESS; } /* * Get echo canceller statistics. */ PJ_DEF(pj_status_t) pjsua_get_ec_stat(pjmedia_echo_stat *p_stat) { if (pjsua_var.snd_port) { return pjmedia_snd_port_get_ec_stat(pjsua_var.snd_port, p_stat); } else { return PJ_ENOTFOUND; } } /* * Check whether the sound device is currently active. */ PJ_DEF(pj_bool_t) pjsua_snd_is_active(void) { return pjsua_var.snd_port != NULL; } /* * Configure sound device setting to the sound device being used. */ PJ_DEF(pj_status_t) pjsua_snd_set_setting( pjmedia_aud_dev_cap cap, const void *pval, pj_bool_t keep) { pj_status_t status; /* Check if we are allowed to set the cap */ if ((cap & pjsua_var.aud_svmask) == 0) { return PJMEDIA_EAUD_INVCAP; } PJSUA_LOCK(); /* If sound is active, set it immediately */ if (pjsua_snd_is_active()) { pjmedia_aud_stream *strm; strm = pjmedia_snd_port_get_snd_stream(pjsua_var.snd_port); status = pjmedia_aud_stream_set_cap(strm, cap, pval); } else { status = PJ_SUCCESS; } if (status != PJ_SUCCESS) { PJSUA_UNLOCK(); return status; } /* Save in internal param for later device open */ if (keep) { status = pjmedia_aud_param_set_cap(&pjsua_var.aud_param, cap, pval); } PJSUA_UNLOCK(); return status; } /* * Retrieve a sound device setting. */ PJ_DEF(pj_status_t) pjsua_snd_get_setting( pjmedia_aud_dev_cap cap, void *pval) { pj_status_t status; PJSUA_LOCK(); /* If sound device has never been opened before, open it to * retrieve the initial setting from the device (e.g. audio * volume) */ if (pjsua_var.aud_open_cnt==0) { PJ_LOG(4,(THIS_FILE, "Opening sound device to get initial settings")); pjsua_set_snd_dev(pjsua_var.cap_dev, pjsua_var.play_dev); close_snd_dev(); } if (pjsua_snd_is_active()) { /* Sound is active, retrieve from device directly */ pjmedia_aud_stream *strm; strm = pjmedia_snd_port_get_snd_stream(pjsua_var.snd_port); status = pjmedia_aud_stream_get_cap(strm, cap, pval); } else { /* Otherwise retrieve from internal param */ status = pjmedia_aud_param_get_cap(&pjsua_var.aud_param, cap, pval); } PJSUA_UNLOCK(); return status; } /* * Extra sound device */ struct pjsua_ext_snd_dev { pj_pool_t *pool; pjmedia_port *splitcomb; pjmedia_port *rev_port; pjmedia_snd_port *snd_port; pjsua_conf_port_id port_id; }; /* * Create an extra sound device and register it to conference bridge. */ PJ_DEF(pj_status_t) pjsua_ext_snd_dev_create( pjmedia_snd_port_param *param, pjsua_ext_snd_dev **p_snd) { pjsua_ext_snd_dev *snd = NULL; pj_pool_t *pool; pj_status_t status; PJ_ASSERT_RETURN(param && p_snd, PJ_EINVAL); PJ_ASSERT_RETURN(param->base.channel_count == 1, PJMEDIA_ENCCHANNEL); pool = pjsua_pool_create("extsnd%p", 512, 512); if (!pool) return PJ_ENOMEM; snd = PJ_POOL_ZALLOC_T(pool, pjsua_ext_snd_dev); if (!snd) { pj_pool_release(pool); return PJ_ENOMEM; } snd->pool = pool; snd->port_id = PJSUA_INVALID_ID; /* Create mono splitter/combiner */ status = pjmedia_splitcomb_create( pool, param->base.clock_rate, param->base.channel_count, param->base.samples_per_frame, param->base.bits_per_sample, 0, /* options */ &snd->splitcomb); if (status != PJ_SUCCESS) goto on_return; /* Create reverse channel */ status = pjmedia_splitcomb_create_rev_channel( pool, snd->splitcomb, 0 /* channel #1 */, 0 /* options */, &snd->rev_port); if (status != PJ_SUCCESS) goto on_return; /* And register it to conference bridge */ status = pjsua_conf_add_port(pool, snd->rev_port, &snd->port_id); if (status != PJ_SUCCESS) goto on_return; /* Create sound device */ status = pjmedia_snd_port_create2(pool, param, &snd->snd_port); if (status != PJ_SUCCESS) goto on_return; /* Connect the splitter to the sound device */ status = pjmedia_snd_port_connect(snd->snd_port, snd->splitcomb); if (status != PJ_SUCCESS) goto on_return; /* Finally */ *p_snd = snd; PJ_LOG(4,(THIS_FILE, "Extra sound device created")); on_return: if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Failed creating extra sound device", status); pjsua_ext_snd_dev_destroy(snd); } return status; } /* * Destroy an extra sound device and unregister it from conference bridge. */ PJ_DEF(pj_status_t) pjsua_ext_snd_dev_destroy(pjsua_ext_snd_dev *snd) { PJ_ASSERT_RETURN(snd, PJ_EINVAL); /* Unregister from the conference bridge */ if (snd->port_id != PJSUA_INVALID_ID) { pjsua_conf_remove_port(snd->port_id); snd->port_id = PJSUA_INVALID_ID; } /* Destroy all components */ if (snd->snd_port) { pjmedia_snd_port_disconnect(snd->snd_port); pjmedia_snd_port_destroy(snd->snd_port); snd->snd_port = NULL; } if (snd->rev_port) { pjmedia_port_destroy(snd->rev_port); snd->rev_port = NULL; } if (snd->splitcomb) { pjmedia_port_destroy(snd->splitcomb); snd->splitcomb = NULL; } /* Finally */ pj_pool_safe_release(&snd->pool); PJ_LOG(4,(THIS_FILE, "Extra sound device destroyed")); return PJ_SUCCESS; } /* * Get sound port instance of an extra sound device. */ PJ_DEF(pjmedia_snd_port*) pjsua_ext_snd_dev_get_snd_port( pjsua_ext_snd_dev *snd) { PJ_ASSERT_RETURN(snd, NULL); return snd->snd_port; } /* * Get conference port ID of an extra sound device. */ PJ_DEF(pjsua_conf_port_id) pjsua_ext_snd_dev_get_conf_port( pjsua_ext_snd_dev *snd) { PJ_ASSERT_RETURN(snd, PJSUA_INVALID_ID); return snd->port_id; } #endif /* PJSUA_MEDIA_HAS_PJMEDIA */