diff options
Diffstat (limited to 'pjsip/src/pjsua-lib/pjsua_aud.c')
| -rw-r--r-- | pjsip/src/pjsua-lib/pjsua_aud.c | 2148 |
1 files changed, 2148 insertions, 0 deletions
diff --git a/pjsip/src/pjsua-lib/pjsua_aud.c b/pjsip/src/pjsua-lib/pjsua_aud.c new file mode 100644 index 0000000..557a147 --- /dev/null +++ b/pjsip/src/pjsua-lib/pjsua_aud.c @@ -0,0 +1,2148 @@ +/* $Id: pjsua_aud.c 4145 2012-05-22 23:13:22Z bennylp $ */ +/* + * 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 <pjsua-lib/pjsua_internal.h> + +#if defined(PJSUA_MEDIA_HAS_PJMEDIA) && PJSUA_MEDIA_HAS_PJMEDIA != 0 + +#define THIS_FILE "pjsua_aud.c" +#define NULL_SND_DEV_ID -99 + +/***************************************************************************** + * + * 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]; + 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 = 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) { + PJ_PERROR(1,(THIS_FILE, status, "Error registering codecs")); + 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); + + 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; i<PJ_ARRAY_SIZE(pjsua_var.player); ++i) { + if (pjsua_var.player[i].port) { + pjmedia_port_destroy(pjsua_var.player[i].port); + pjsua_var.player[i].port = NULL; + } + } + + /* Destroy file recorders */ + for (i=0; i<PJ_ARRAY_SIZE(pjsua_var.recorder); ++i) { + if (pjsua_var.recorder[i].port) { + pjmedia_port_destroy(pjsua_var.recorder[i].port); + pjsua_var.recorder[i].port = NULL; + } + } + + return PJ_SUCCESS; +} + +void pjsua_aud_stop_stream(pjsua_call_media *call_med) +{ + pjmedia_stream *strm = call_med->strm.a.stream; + pjmedia_rtcp_stat stat; + + if (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)) + { + /* 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); + } + + 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(); + + /* For discussions about call mutex protection related to this + * callback, please see ticket #460: + * http://trac.pjsip.org/repos/ticket/460#comment:4 + */ + if (pjsua_var.ua_cfg.cb.on_dtmf_digit) { + pjsua_call_id call_id; + + call_id = (pjsua_call_id)(long)user_data; + pjsua_var.ua_cfg.cb.on_dtmf_digit(call_id, digit); + } + + pj_log_pop_indent(); +} + + +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; + pjmedia_port *media_port; + 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 = PJ_TRUE; + + /* Check if no media is active */ + if (si->dir != PJMEDIA_DIR_NONE) { + + /* Override ptime, if this option is specified. */ + if (pjsua_var.media_cfg.ptime != 0) { + si->param->setting.frm_per_pkt = (pj_uint8_t) + (pjsua_var.media_cfg.ptime / si->param->info.frm_ptime); + if (si->param->setting.frm_per_pkt == 0) + si->param->setting.frm_per_pkt = 1; + } + + /* Disable VAD, if this option is specified. */ + if (pjsua_var.media_cfg.no_vad) { + si->param->setting.vad = 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 */ + si->ssrc = call_med->ssrc; + + /* 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) { + pjmedia_stream_set_dtmf_callback(call_med->strm.a.stream, + &dtmf_callback, + (void*)(long)(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, &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_created) { + pjsua_var.ua_cfg.cb.on_stream_created(call->index, + call_med->strm.a.stream, + strm_idx, &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_prov, + media_port, + &port_name, + (unsigned*) + &call_med->strm.a.conf_slot); + if (status != PJ_SUCCESS) { + goto on_return; + } + } + } + +on_return: + pj_log_pop_indent(); + return status; +} + + +/* + * 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; + 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; + + /* Build array of listeners */ + info->listener_cnt = cinfo.listener_cnt; + for (i=0; i<cinfo.listener_cnt; ++i) { + info->listeners[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) +{ + 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 != NULL_SND_DEV_ID) { + 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) { + status = pjmedia_conf_connect_port(pjsua_var.mconf, source, sink, 0); + } + + 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_id<PJ_ARRAY_SIZE(pjsua_var.player); ++file_id) { + if (pjsua_var.player[file_id].port == NULL) + break; + } + + if (file_id == PJ_ARRAY_SIZE(pjsua_var.player)) { + /* This is unexpected */ + pj_assert(0); + status = PJ_EBUG; + goto on_error; + } + + pj_memcpy(path, filename->ptr, filename->slen); + path[filename->slen] = '\0'; + + pool = pjsua_pool_create(get_basename(path, 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_id<PJ_ARRAY_SIZE(pjsua_var.player); ++file_id) { + if (pjsua_var.player[file_id].port == NULL) + break; + } + + if (file_id == PJ_ARRAY_SIZE(pjsua_var.player)) { + /* This is unexpected */ + pj_assert(0); + status = PJ_EBUG; + goto on_error; + } + + + ptime = pjsua_var.mconf_cfg.samples_per_frame * 1000 / + pjsua_var.media_cfg.clock_rate; + + pool = pjsua_pool_create("playlist", 1000, 1000); + if (!pool) { + status = PJ_ENOMEM; + goto on_error; + } + + status = pjmedia_wav_playlist_create(pool, label, + file_names, file_count, + ptime, options, 0, &port); + if (status != PJ_SUCCESS) { + pjsua_perror(THIS_FILE, "Unable to create playlist", status); + goto on_error; + } + + status = pjmedia_conf_add_port(pjsua_var.mconf, pool, + port, &port->info.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; +} + +/* + * Set playback position. + */ +PJ_DEF(pj_status_t) pjsua_player_set_pos( pjsua_player_id id, + pj_uint32_t samples) +{ + 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_port_set_pos(pjsua_var.player[id].port, samples); +} + + +/* + * 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_id<PJ_ARRAY_SIZE(pjsua_var.recorder); ++file_id) { + if (pjsua_var.recorder[file_id].port == NULL) + break; + } + + if (file_id == PJ_ARRAY_SIZE(pjsua_var.recorder)) { + /* This is unexpected */ + pj_assert(0); + status = PJ_EBUG; + goto on_return; + } + + pj_memcpy(path, filename->ptr, filename->slen); + path[filename->slen] = '\0'; + + pool = pjsua_pool_create(get_basename(path, 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<dev_count; ++i) { + pj_status_t status; + + status = pjmedia_aud_dev_get_info(i, &info[i]); + if (status != PJ_SUCCESS) + return status; + } + + *count = dev_count; + + return PJ_SUCCESS; +} + + +PJ_DEF(pj_status_t) pjsua_enum_snd_devs( pjmedia_snd_dev_info info[], + unsigned *count) +{ + unsigned i, dev_count; + + dev_count = pjmedia_aud_dev_count(); + + if (dev_count > *count) dev_count = *count; + pj_bzero(info, dev_count * sizeof(pjmedia_snd_dev_info)); + + for (i=0; i<dev_count; ++i) { + pjmedia_aud_dev_info ai; + pj_status_t status; + + status = pjmedia_aud_dev_get_info(i, &ai); + if (status != PJ_SUCCESS) + return status; + + strncpy(info[i].name, ai.name, sizeof(info[i].name)); + info[i].name[sizeof(info[i].name)-1] = '\0'; + info[i].input_count = ai.input_count; + info[i].output_count = ai.output_count; + info[i].default_samples_per_sec = ai.default_samples_per_sec; + } + + *count = dev_count; + + return PJ_SUCCESS; +} + +/* Create audio device parameter to open the device */ +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) +{ + pj_status_t status; + + /* Normalize device ID with new convention about default device ID */ + if (playback_dev == PJMEDIA_AUD_DEFAULT_CAPTURE_DEV) + playback_dev = PJMEDIA_AUD_DEFAULT_PLAYBACK_DEV; + + /* Create default parameters for the device */ + status = pjmedia_aud_dev_default_param(capture_dev, param); + if (status != PJ_SUCCESS) { + pjsua_perror(THIS_FILE, "Error retrieving default audio " + "device parameters", status); + return status; + } + param->dir = 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); + } + + 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; +} + +/* 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_ASSERT_RETURN(param, PJ_EINVAL); + + /* Check if NULL sound device is used */ + if (NULL_SND_DEV_ID==param->base.rec_id || + NULL_SND_DEV_ID==param->base.play_id) + { + return pjsua_set_null_snd_dev(); + } + + /* Close existing sound port */ + close_snd_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 sound device %s@%d/%d/%dms", + 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(); + + 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; + } + + /* Save the device IDs */ + pjsua_var.cap_dev = param->base.rec_id; + pjsua_var.play_dev = param->base.play_id; + + /* 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 = sizeof(tmp_buf); + + tmp_buf_len = pj_ansi_snprintf(tmp_buf, sizeof(tmp_buf)-1, + "%s (%dKHz)", + rec_info.name, + param->base.clock_rate/1000); + 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; + + 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)); + + 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(); +} + + +/* + * 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_dev( int capture_dev, + int playback_dev) +{ + unsigned alt_cr_cnt = 1; + unsigned alt_cr[] = {0, 44100, 48000, 32000, 16000, 8000}; + unsigned i; + pj_status_t status = -1; + + PJ_LOG(4,(THIS_FILE, "Set sound device: capture=%d, playback=%d", + capture_dev, playback_dev)); + pj_log_push_indent(); + + PJSUA_LOCK(); + + /* Null-sound */ + if (capture_dev==NULL_SND_DEV_ID && playback_dev==NULL_SND_DEV_ID) { + PJSUA_UNLOCK(); + status = pjsua_set_null_snd_dev(); + pj_log_pop_indent(); + return status; + } + + /* 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; i<alt_cr_cnt; ++i) { + pjmedia_snd_port_param param; + unsigned samples_per_frame; + + /* Create the default audio param */ + samples_per_frame = alt_cr[i] * + pjsua_var.media_cfg.audio_frame_ptime * + pjsua_var.media_cfg.channel_count / 1000; + pjmedia_snd_port_param_default(¶m); + param.ec_options = pjsua_var.media_cfg.ec_options; + status = create_aud_param(¶m.base, capture_dev, 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_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(); + + /* 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.cap_dev = NULL_SND_DEV_ID; + pjsua_var.play_dev = NULL_SND_DEV_ID; + + 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_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; +} + + +/* + * 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; +} + +#endif /* PJSUA_MEDIA_HAS_PJMEDIA */ |