Changes between Version 3 and Version 4 of Nokia_APS_VAS_Direct

Timestamp:

Feb 12, 2009 2:48:55 PM (15 years ago)

Author:

nanang

Comment:

Added instructions for using APS-Direct (initial version).

Nokia_APS_VAS_Direct

@@ -18 +18 @@
 
 Due to these benefits, the ability to use these codecs in PJSIP applications is very desirable.
+
+Note that non-APS codecs can still be used as usual, e.g: GSM, Speex/8000.
 
 [[BR]]
@@ -64 +66 @@
 
 The APS/VAS based sound device backends will support additional APIs:
- - to query the list of supported codecs/formats,
+ - to query the list of supported codecs/formats (for APS, the list is currently hardcoded),
  - to set which format to use when opening the sound device,
  - audio routing to loudspeaker or earpiece (this API is already available)
@@ -103 +105 @@
 == Using APS-Direct or VAS-Direct ==
 
-TBD.
+Currently, it's only APS-Direct that has been implemented, here are the steps to build an application with APS-Direct feature.
+
+ 1. Enable APS sound device implementation as described [http://trac.pjsip.org/repos/wiki/APS here].
+ 1. Enable audio switch board, i.e. in config_site.h:
+ {{{
+#define PJMEDIA_CONF_USE_SWITCH_BOARD   1
+ }}}
+ 1. Enable passthrough codecs, i.e. in config_site.h:
+ {{{
+#define PJMEDIA_HAS_PASSTHROUGH_CODECS  1
+ }}}
+
+For building sample application {{{symbian_ua}}}, those steps are enough since it's already prepared to use APS-Direct. 
+
+For general application, there are few more things to be handled:
+ - Reopening sound device when it needs to change the active format/codec, e.g: when a call is confirmed and stream has been created, the sound device format should be matched to the SDP negotiation result. Here is the sample code for application that using pjsua-lib, reopening sound device is done in {{{on_stream_created()}}} pjsua callback, this will replace the precreated pjsua-lib sound device instance:
+ {{{
+/* Global sound port. */
+static pjmedia_snd_port *snd_port;
+
+/* Reopen sound device on on_stream_created() pjsua callback. */
+static void on_stream_created(pjsua_call_id call_id, 
+                              pjmedia_session *sess,
+                              unsigned stream_idx, 
+                              pjmedia_port **)
+{
+    pjmedia_port *conf;
+    pjmedia_session_info sess_info;
+    pjmedia_stream_info *strm_info;
+    pjmedia_snd_setting setting;
+    unsigned samples_per_frame;
+
+    /* Get active format for this stream, based on SDP negotiation result. */    
+    pjmedia_session_get_info(sess, &sess_info);
+    strm_info = &sess_info.stream_info[stream_idx];
+
+    /* Init sound device setting based on stream info. */
+    pj_bzero(&setting, sizeof(setting));
+    setting.format = strm_info->param->info.format;
+    setting.bitrate = strm_info->param->info.avg_bps;
+    setting.cng = strm_info->param->setting.cng;
+    setting.vad = strm_info->param->setting.vad;
+    setting.plc = strm_info->param->setting.plc;
+
+    /* Close sound device and get the conference port. */
+    conf = pjsua_set_no_snd_dev();
+    
+    samples_per_frame = strm_info->param->info.clock_rate *
+                        strm_info->param->info.frm_ptime *
+                        strm_info->param->info.channel_cnt /
+                        1000;
+
+    /* Reset conference port attributes. */
+    conf->info.samples_per_frame = samples_per_frame;
+    conf->info.clock_rate = 8000;
+    conf->info.channel_count = 1;
+    conf->info.bits_per_sample = 16;
+
+    /* Reopen sound device. */
+    pjmedia_snd_port_create2(app_pool, 
+                             PJMEDIA_DIR_CAPTURE_PLAYBACK,
+                             0,
+                             0,
+                             8000,
+                             1,
+                             samples_per_frame,
+                             16,
+                             &setting,
+                             &snd_port);
+
+    /* Connect sound to conference port. */
+    pjmedia_snd_port_connect(snd_port, conf);
+}
+ }}}
+ - Note that sound device instance is now owned and managed by application, so {{{pjsua_media_config.snd_auto_close_time}}} will not work. Here is a very simple sample code to close the sound device immediately when a call get disconnected:
+ {{{
+/* Callback called by the pjsua-lib when call's state has changed. */
+static void on_call_state(pjsua_call_id call_id, pjsip_event *)
+{
+    pjsua_call_info ci;
+
+    pjsua_call_get_info(call_id, &ci);
+    
+    if (ci.state == PJSIP_INV_STATE_DISCONNECTED) {
+        if (snd_port) {
+            pjmedia_snd_port_destroy(snd_port);
+            snd_port = NULL;
+        }
+    }
+}
+ }}}