* #27232: jni: added pjproject checkout as regular git content
We will remove it once the next release of pjsip (with Android support)
comes out and is merged into SFLphone.
diff --git a/jni/pjproject-android/pjsip/src/pjsua-lib/pjsua_aud.c b/jni/pjproject-android/pjsip/src/pjsua-lib/pjsua_aud.c
new file mode 100644
index 0000000..0f768b8
--- /dev/null
+++ b/jni/pjproject-android/pjsip/src/pjsua-lib/pjsua_aud.c
@@ -0,0 +1,2151 @@
+/* $Id: pjsua_aud.c 4537 2013-06-19 06:47:43Z riza $ */
+/*
+ * 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 =
+ 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) {
+ 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);
+
+ 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; 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)(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;
+ 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 = pjsua_var.media_cfg.no_rtcp_sdes_bye;
+
+ /* Check if no media is active */
+ if (si->dir != PJMEDIA_DIR_NONE) {
+
+ /* 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*)(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, &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,
+ 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, (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_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, (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<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);
+ }
+
+ /* 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;
+}
+
+/* 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 */