diff --git a/sflphone-client-gnome/src/config/audioconf.c b/sflphone-client-gnome/src/config/audioconf.c
index 62abf5a4e247aa7f719f379586c0a66451bead96..29ec95f57ba570700c07a4e68e393b5ac0973426 100644
--- a/sflphone-client-gnome/src/config/audioconf.c
+++ b/sflphone-client-gnome/src/config/audioconf.c
@@ -35,9 +35,11 @@
GtkListStore *pluginlist;
GtkListStore *outputlist;
GtkListStore *inputlist;
+GtkListStore *ringtonelist;
GtkWidget *output;
GtkWidget *input;
+GtkWidget *ringtone;
GtkWidget *plugin;
GtkWidget *codecMoveUpButton;
GtkWidget *codecMoveDownButton;
@@ -103,7 +105,7 @@ void preferences_dialog_fill_codec_list (account_t **a) {
* Fill store with output audio plugins
*/
void
-preferences_dialog_fill_output_audio_plugin_list()
+preferences_dialog_fill_audio_plugin_list()
{
GtkTreeIter iter;
gchar** list;
@@ -112,7 +114,7 @@ preferences_dialog_fill_output_audio_plugin_list()
gtk_list_store_clear(pluginlist);
// Call dbus to retreive list
- list = dbus_get_output_audio_plugin_list();
+ list = dbus_get_audio_plugin_list();
// For each API name included in list
int c = 0;
@@ -127,6 +129,7 @@ preferences_dialog_fill_output_audio_plugin_list()
list = NULL;
}
+
/**
* Fill output audio device store
*/
@@ -155,6 +158,37 @@ preferences_dialog_fill_output_audio_device_list()
}
}
+
+/**
+ * Fill rigntone audio device store
+ */
+
+void
+preferences_dialog_fill_ringtone_audio_device_list()
+{
+
+ GtkTreeIter iter;
+ gchar** list;
+ gchar** audioDevice;
+ int index;
+
+ gtk_list_store_clear(ringtonelist);
+
+ // Call dbus to retreive output device
+ list = dbus_get_audio_output_device_list();
+
+ // For each device name in the list
+ int c = 0;
+ for(audioDevice = list; *list; list++) {
+ index = dbus_get_audio_device_index( *list );
+ gtk_list_store_append(ringtonelist, &iter);
+ gtk_list_store_set(ringtonelist, &iter, 0, *list, 1, index, -1);
+ c++;
+ }
+}
+
+
+
/**
* Select active output audio device
*/
@@ -194,6 +228,46 @@ select_active_output_audio_device()
}
}
+
+/**
+ * Select active output audio device
+ */
+ void
+select_active_ringtone_audio_device()
+{
+ if( SHOW_ALSA_CONF )
+ {
+
+ GtkTreeModel* model;
+ GtkTreeIter iter;
+ gchar** devices;
+ int currentDeviceIndex;
+ int deviceIndex;
+
+ // Select active ringtone device on server
+ devices = dbus_get_current_audio_devices_index();
+ currentDeviceIndex = atoi(devices[2]);
+ DEBUG("audio device index for ringtone = %d", currentDeviceIndex);
+ model = gtk_combo_box_get_model(GTK_COMBO_BOX(ringtone));
+
+ // Find the currently set ringtone device
+ gtk_tree_model_get_iter_first(model, &iter);
+ do {
+ gtk_tree_model_get(model, &iter, 1, &deviceIndex, -1);
+ if(deviceIndex == currentDeviceIndex)
+ {
+ // Set current iteration the active one
+ gtk_combo_box_set_active_iter(GTK_COMBO_BOX(ringtone), &iter);
+ return;
+ }
+ } while(gtk_tree_model_iter_next(model, &iter));
+
+ // No index was found, select first one
+ WARN("Warning : No active ringtone device found");
+ gtk_combo_box_set_active(GTK_COMBO_BOX(ringtone), 0);
+ }
+}
+
/**
* Fill input audio device store
*/
@@ -364,6 +438,29 @@ select_audio_input_device(GtkComboBox* comboBox, gpointer data UNUSED)
}
}
+/**
+ * Set the audio ringtone device on the server with its index
+ */
+static void
+select_audio_ringtone_device(GtkComboBox *comboBox, gpointer data UNUSED)
+{
+ GtkTreeModel *model;
+ GtkTreeIter iter;
+ int comboBoxIndex;
+ int deviceIndex;
+
+ comboBoxIndex = gtk_combo_box_get_active(comboBox);
+
+ if(comboBoxIndex >= 0) {
+ model = gtk_combo_box_get_model(comboBox);
+ gtk_combo_box_get_active_iter(comboBox, &iter);
+
+ gtk_tree_model_get(model, &iter, 1, &deviceIndex, -1);
+
+ dbus_set_audio_ringtone_device(deviceIndex);
+ }
+}
+
/**
* Toggle move buttons on if a codec is selected, off elsewise
*/
@@ -647,8 +744,7 @@ GtkWidget* codecs_box (account_t **a)
void
select_audio_manager( void )
{
-
- DEBUG("audio manager selected\n");
+ DEBUG("audio manager selected");
if( !SHOW_ALSA_CONF && !gtk_toggle_button_get_active( GTK_TOGGLE_BUTTON(pulse) ) )
{
@@ -690,19 +786,19 @@ GtkWidget* alsa_box()
ret = gtk_hbox_new(FALSE, 10);
gtk_widget_show( ret );
- table = gtk_table_new(4, 3, FALSE);
+ table = gtk_table_new(5, 3, FALSE);
gtk_table_set_col_spacing(GTK_TABLE(table), 0, 40);
gtk_box_pack_start( GTK_BOX(ret) , table , TRUE , TRUE , 1);
gtk_widget_show(table);
- DEBUG("plugin");
+ DEBUG("Audio: Configuration plugin");
item = gtk_label_new(_("ALSA plugin"));
gtk_misc_set_alignment(GTK_MISC(item), 0, 0.5);
gtk_table_attach(GTK_TABLE(table), item, 1, 2, 1, 2, GTK_FILL | GTK_EXPAND, GTK_SHRINK, 0, 0);
gtk_widget_show( item );
// Set choices of audio managers
pluginlist = gtk_list_store_new(1, G_TYPE_STRING);
- preferences_dialog_fill_output_audio_plugin_list();
+ preferences_dialog_fill_audio_plugin_list();
plugin = gtk_combo_box_new_with_model(GTK_TREE_MODEL(pluginlist));
select_active_output_audio_plugin();
gtk_label_set_mnemonic_widget(GTK_LABEL(item), plugin);
@@ -717,7 +813,7 @@ GtkWidget* alsa_box()
// Device : Output device
// Create title label
- DEBUG("output");
+ DEBUG("Audio: Configuration output");
item = gtk_label_new(_("Output"));
gtk_misc_set_alignment(GTK_MISC(item), 0, 0.5);
gtk_table_attach(GTK_TABLE(table), item, 1, 2, 2, 3, GTK_FILL | GTK_EXPAND, GTK_SHRINK, 0, 0);
@@ -739,7 +835,7 @@ GtkWidget* alsa_box()
// Device : Input device
// Create title label
- DEBUG("input");
+ DEBUG("Audio: Configuration input");
item = gtk_label_new(_("Input"));
gtk_misc_set_alignment(GTK_MISC(item), 0, 0.5);
gtk_table_attach(GTK_TABLE(table), item, 1, 2, 3, 4, GTK_FILL | GTK_EXPAND, GTK_SHRINK, 0, 0);
@@ -760,6 +856,27 @@ GtkWidget* alsa_box()
gtk_table_attach(GTK_TABLE(table), input, 2, 3, 3, 4, GTK_FILL | GTK_EXPAND, GTK_SHRINK, 0, 0);
gtk_widget_show(input);
+
+ DEBUG("Audio: Configuration rintgtone");
+ item = gtk_label_new(_("Ringtone"));
+ gtk_misc_set_alignment(GTK_MISC(item), 0, 0.5);
+ gtk_table_attach(GTK_TABLE(table), item, 1, 2, 4, 5, GTK_FILL | GTK_EXPAND, GTK_SHRINK, 0, 0);
+ gtk_widget_show(item);
+ // set choices of ringtone devices
+ ringtonelist = gtk_list_store_new(2, G_TYPE_STRING, G_TYPE_INT);
+ preferences_dialog_fill_ringtone_audio_device_list();
+ ringtone = gtk_combo_box_new_with_model(GTK_TREE_MODEL(ringtonelist));
+ select_active_ringtone_audio_device();
+ gtk_label_set_mnemonic_widget(GTK_LABEL(item), output);
+ g_signal_connect(G_OBJECT(ringtone), "changed", G_CALLBACK(select_audio_ringtone_device), output);
+
+ // Set rendering
+ renderer = gtk_cell_renderer_text_new();
+ gtk_cell_layout_pack_start(GTK_CELL_LAYOUT(ringtone), renderer, TRUE);
+ gtk_cell_layout_set_attributes(GTK_CELL_LAYOUT(ringtone), renderer, "text", 0, NULL);
+ gtk_table_attach(GTK_TABLE(table), ringtone, 2, 3, 4, 5, GTK_FILL | GTK_EXPAND, GTK_SHRINK, 0, 0);
+ gtk_widget_show(ringtone);
+
gtk_widget_show_all(ret);
DEBUG("done");
diff --git a/sflphone-client-gnome/src/dbus/configurationmanager-introspec.xml b/sflphone-client-gnome/src/dbus/configurationmanager-introspec.xml
index 8d8dcf9468526693cbaa6b74e668247fb0485919..8a0c59c8b2b1aa69d46519915f26228a9d27584b 100644
--- a/sflphone-client-gnome/src/dbus/configurationmanager-introspec.xml
+++ b/sflphone-client-gnome/src/dbus/configurationmanager-introspec.xml
@@ -105,8 +105,8 @@
<tp:docstring>
</tp:docstring>
<arg type="s" name="accountID" direction="in">
- <tp:docstring>
- </tp:docstring>
+ <tp:docstring>
+ </tp:docstring>
</arg>
<arg type="i" name="number" direction="in">
<tp:docstring>
@@ -118,8 +118,9 @@
<tp:docstring>
</tp:docstring>
<arg type="s" name="accountID" direction="in">
- <tp:docstring>
- </tp:docstring>
+ <tp:docstring>
+ Account ID
+ </tp:docstring>
</arg>
</method>
@@ -206,55 +207,55 @@
<method name="addAccount" tp:name-for-bindings="addAccount">
<tp:docstring>
- Add a new account. When created, the signal <tp:member-ref>accountsChanged</tp:member-ref> is emitted. The clients must then call <tp:member-ref>getAccountList</tp:member-ref> to update their internal data structure.
- <tp:rationale>If no details are specified, the default parameters are used.</tp:rationale>
- <tp:rationale>The core tries to register the account as soon it is created.</tp:rationale>
+ Add a new account. When created, the signal <tp:member-ref>accountsChanged</tp:member-ref> is emitted. The clients must then call <tp:member-ref>getAccountList</tp:member-ref> to update their internal data structure.
+ <tp:rationale>If no details are specified, the default parameters are used.</tp:rationale>
+ <tp:rationale>The core tries to register the account as soon it is created.</tp:rationale>
</tp:docstring>
<annotation name="com.trolltech.QtDBus.QtTypeName.In0" value="MapStringString"/>
<arg type="a{ss}" name="details" direction="in" tp:type="String_String_Map">
- <tp:docstring>
- The new account settings
- </tp:docstring>
+ <tp:docstring>
+ The new account settings
+ </tp:docstring>
</arg>
<arg type="s" name="createdAccountId" direction="out">
- <tp:docstring>
- A new account ID
- </tp:docstring>
+ <tp:docstring>
+ A new account ID
+ </tp:docstring>
</arg>
</method>
<method name="setAccountsOrder" tp:name-for-bindings="setAccountsOrder">
<tp:docstring>
- Update the accounts order.
- <tp:rationale>When placing a call, the first registered account in the list is used.</tp:rationale>
+ Update the accounts order.
+ <tp:rationale>When placing a call, the first registered account in the list is used.</tp:rationale>
</tp:docstring>
<arg type="s" name="order" direction="in">
- <tp:docstring>
- An ordered list of account IDs, delimited by '/'
- </tp:docstring>
+ <tp:docstring>
+ An ordered list of account IDs, delimited by '/'
+ </tp:docstring>
</arg>
</method>
<method name="removeAccount" tp:name-for-bindings="removeAccount">
- <tp:docstring>
- Remove an existing account. When removed, the signal <tp:member-ref>accountsChanged</tp:member-ref> is emitted. The clients must then call <tp:member-ref>getAccountList</tp:member-ref> to update their internal data structure.
- </tp:docstring>
- <arg type="s" name="accoundID" direction="in">
- <tp:docstring>
- The account to remove, identified by its ID
- </tp:docstring>
- </arg>
+ <tp:docstring>
+ Remove an existing account. When removed, the signal <tp:member-ref>accountsChanged</tp:member-ref> is emitted. The clients must then call <tp:member-ref>getAccountList</tp:member-ref> to update their internal data structure.
+ </tp:docstring>
+ <arg type="s" name="accoundID" direction="in">
+ <tp:docstring>
+ The account to remove, identified by its ID
+ </tp:docstring>
+ </arg>
</method>
<method name="getAccountList" tp:name-for-bindings="getAccountList">
- <tp:docstring>
- Get a list of all created accounts, as stored by the core.
- </tp:docstring>
+ <tp:docstring>
+ Get a list of all created accounts, as stored by the core.
+ </tp:docstring>
<annotation name="com.trolltech.QtDBus.QtTypeName.Out0" value="VectorString"/>
<arg type="as" name="list" direction="out">
- <tp:docstring>
- A list of account IDs
- </tp:docstring>
+ <tp:docstring>
+ A list of account IDs
+ </tp:docstring>
</arg>
</method>
@@ -267,15 +268,15 @@
@param[in] input accountID
-->
<arg type="s" name="accountID" direction="in">
- <tp:docstring>
- The account ID
- </tp:docstring>
+ <tp:docstring>
+ The account ID
+ </tp:docstring>
</arg>
<arg type="i" name="expire" direction="in">
- <tp:docstring>
- <p>To register, expire must be 1.</p>
- <p>To un-register, expire must be 0.</p>
- </tp:docstring>
+ <tp:docstring>
+ <p>To register, expire must be 1.</p>
+ <p>To un-register, expire must be 0.</p>
+ </tp:docstring>
</arg>
</method>
@@ -284,13 +285,14 @@
</tp:docstring>
<annotation name="com.trolltech.QtDBus.QtTypeName.Out0" value="VectorString"/>
<arg type="as" name="list" direction="out">
- <tp:docstring>
- </tp:docstring>
+ <tp:docstring>
+ </tp:docstring>
</arg>
</method>
<method name="getVersion" tp:name-for-bindings="getVersion">
<tp:docstring>
+ Return SFLphone-daemon version
</tp:docstring>
<arg type="s" name="version" direction="out">
<tp:docstring>
@@ -300,16 +302,19 @@
<method name="getRingtoneList" tp:name-for-bindings="getRingtoneList">
<tp:docstring>
+ Return a list of valid Sun's .au sound file used
+ as ringtones.
</tp:docstring>
<annotation name="com.trolltech.QtDBus.QtTypeName.Out0" value="VectorString"/>
<arg type="as" name="list" direction="out">
- <tp:docstring>
- </tp:docstring>
+ <tp:docstring>
+ </tp:docstring>
</arg>
</method>
<method name="getPlaybackDeviceList" tp:name-for-bindings="getPlaybackDeviceList">
<tp:docstring>
+ Provide a list of playback device from ALSA
</tp:docstring>
<annotation name="com.trolltech.QtDBus.QtTypeName.Out0" value="VectorString"/>
<arg type="as" name="list" direction="out">
@@ -320,6 +325,7 @@
<method name="getRecordDeviceList" tp:name-for-bindings="getRecordDeviceList">
<tp:docstring>
+ Provide a list of record device from ALSA
</tp:docstring>
<annotation name="com.trolltech.QtDBus.QtTypeName.Out0" value="VectorString"/>
<arg type="as" name="list" direction="out">
@@ -330,6 +336,7 @@
<method name="isRingtoneEnabled" tp:name-for-bindings="isRingtoneEnabled">
<tp:docstring>
+ Return true if ringtone is enabled, false otherwise
</tp:docstring>
<arg type="i" name="bool" direction="out">
<tp:docstring>
@@ -339,24 +346,28 @@
<method name="ringtoneEnabled" tp:name-for-bindings="ringtoneEnabled">
<tp:docstring>
+ Unused
</tp:docstring>
</method>
<method name="getRingtoneChoice" tp:name-for-bindings="getRingtoneChoice">
<tp:docstring>
+ Get current ringtone .au file selected
</tp:docstring>
<arg type="s" name="tone" direction="out">
- <tp:docstring>
- </tp:docstring>
+ <tp:docstring>
+ </tp:docstring>
</arg>
</method>
<method name="setRingtoneChoice" tp:name-for-bindings="setRingtoneChoice">
<tp:docstring>
+ Set current ringtone .au file from list acquired using <tp:member-ref>getAccountList</tp:member-ref>
</tp:docstring>
<arg type="s" name="tone" direction="in">
- <tp:docstring>
- </tp:docstring>
+ <tp:docstring>
+ A valid .au file path
+ </tp:docstring>
</arg>
</method>
@@ -455,17 +466,7 @@
<!-- Audio devices methods -->
- <method name="getInputAudioPluginList" tp:name-for-bindings="getInputAudioPluginList">
- <tp:docstring>
- </tp:docstring>
- <annotation name="com.trolltech.QtDBus.QtTypeName.Out0" value="VectorString"/>
- <arg type="as" name="list" direction="out">
- <tp:docstring>
- </tp:docstring>
- </arg>
- </method>
-
- <method name="getOutputAudioPluginList" tp:name-for-bindings="getOutputAudioPluginList">
+ <method name="getAudioPluginList" tp:name-for-bindings="getAudioPluginList">
<tp:docstring>
</tp:docstring>
<annotation name="com.trolltech.QtDBus.QtTypeName.Out0" value="VectorString"/>
@@ -512,25 +513,35 @@
</arg>
</method>
- <method name="getAudioInputDeviceList" tp:name-for-bindings="getAudioInputDeviceList">
+ <method name="setAudioInputDevice" tp:name-for-bindings="setAudioInputDevice">
<tp:docstring>
</tp:docstring>
- <annotation name="com.trolltech.QtDBus.QtTypeName.Out0" value="VectorString"/>
- <arg type="as" name="list" direction="out">
+ <arg type="i" name="index" direction="in">
<tp:docstring>
</tp:docstring>
</arg>
</method>
- <method name="setAudioInputDevice" tp:name-for-bindings="setAudioInputDevice">
- <tp:docstring>
+ <method name="setAudioRingtoneDevice" tp:name-for-binding="setAudioRingtoneDevice">
+ <tp:docstring>
</tp:docstring>
<arg type="i" name="index" direction="in">
+ <tp:docstring>
+ </tp:docstring>
+ </arg>
+ </method>
+
+ <method name="getAudioInputDeviceList" tp:name-for-bindings="getAudioInputDeviceList">
+ <tp:docstring>
+ </tp:docstring>
+ <annotation name="com.trolltech.QtDBus.QtTypeName.Out0" value="VectorString"/>
+ <arg type="as" name="list" direction="out">
<tp:docstring>
</tp:docstring>
</arg>
</method>
+
<method name="getCurrentAudioDevicesIndex" tp:name-for-bindings="getCurrentAudioDevicesIndex">
<tp:docstring>
</tp:docstring>
@@ -732,6 +743,7 @@
<method name="getWindowWidth" tp:name-for-bindings="getWindowWidth">
<tp:docstring>
+ Unused
</tp:docstring>
<arg type="i" name="width" direction="out">
<tp:docstring>
@@ -741,6 +753,7 @@
<method name="getWindowHeight" tp:name-for-bindings="getWindowHeight">
<tp:docstring>
+ Unused
</tp:docstring>
<arg type="i" name="height" direction="out">
<tp:docstring>
@@ -750,6 +763,7 @@
<method name="setWindowWidth" tp:name-for-bindings="setWindowWidth">
<tp:docstring>
+ Unused
</tp:docstring>
<arg type="i" name="width" direction="in">
<tp:docstring>
@@ -759,6 +773,7 @@
<method name="setWindowHeight" tp:name-for-bindings="setWindowHeight">
<tp:docstring>
+ Unused
</tp:docstring>
<arg type="i" name="height" direction="in">
<tp:docstring>
@@ -768,6 +783,7 @@
<method name="getWindowPositionX" tp:name-for-bindings="getWindowPositionX">
<tp:docstring>
+ Unused
</tp:docstring>
<arg type="i" name="posX" direction="out">
<tp:docstring>
@@ -777,6 +793,7 @@
<method name="setWindowPositionX" tp:name-for-bindings="setWindowPositionX">
<tp:docstring>
+ Unused
</tp:docstring>
<arg type="i" name="posX" direction="in">
<tp:docstring>
@@ -786,6 +803,7 @@
<method name="getWindowPositionY" tp:name-for-bindings="getWindowPositionY">
<tp:docstring>
+ Unused
</tp:docstring>
<arg type="i" name="posY" direction="out">
<tp:docstring>
@@ -795,28 +813,33 @@
<method name="setWindowPositionY" tp:name-for-bindings="setWindowPositionY">
<tp:docstring>
+ Unused
</tp:docstring>
<arg type="i" name="posY" direction="in">
- <tp:docstring>
- </tp:docstring>
+ <tp:docstring>
+ </tp:docstring>
</arg>
</method>
<method name="enableStatusIcon" tp:name-for-bindings="enableStatusIcon">
<tp:docstring>
+ Allow SFLphone icon to be displayed in system tray
</tp:docstring>
<arg type="s" name="value" direction="in">
- <tp:docstring>
- </tp:docstring>
+ <tp:docstring>
+ true/false
+ </tp:docstring>
</arg>
</method>
<method name="isStatusIconEnabled" tp:name-for-bindings="isStatusIconEnabled">
<tp:docstring>
+ Test if SFLphone icon is displayed in system tray.
</tp:docstring>
<arg type="s" name="value" direction="out">
- <tp:docstring>
- </tp:docstring>
+ <tp:docstring>
+ true/false
+ </tp:docstring>
</arg>
</method>
@@ -899,100 +922,144 @@
</tp:docstring>
<annotation name="com.trolltech.QtDBus.QtTypeName.In0" value="MapStringString"/>
<arg type="a{ss}" name="entries" direction="in">
- <tp:docstring>
- </tp:docstring>
+ <tp:docstring>
+ </tp:docstring>
</arg>
</method>
<signal name="accountsChanged" tp:name-for-bindings="accountsChanged">
+ <tp:docstring>
+ Signal emited on account changes. Clients should update
+ all account status with <tp:member-ref>getAccountDetails</tp:member-ref>
+ iterating over the list provided by <tp:member-ref>getAccountList</tp:member-ref>
+ </tp:docstring>
</signal>
<signal name="errorAlert" tp:name-for-bindings="errorAlert">
<arg type="i" name="code">
- <tp:docstring>
- </tp:docstring>
+ <tp:docstring>
+ </tp:docstring>
</arg>
</signal>
<!-- TLS Methods -->
<method name="getSupportedTlsMethod" tp:name-for-bindings="getSupportedTlsMethod">
<tp:docstring>
+ Provide a list of supported TLS method
</tp:docstring>
<annotation name="com.trolltech.QtDBus.QtTypeName.Out0" value="VectorString"/>
<arg type="as" name="list" direction="out">
- <tp:docstring>
- </tp:docstring>
+ <tp:docstring>
+ A list of TLS method: (TLSv1, SSLv1, SSLv2,
+ SSLv3, SSLv23)
+ </tp:docstring>
</arg>
</method>
<method name="getTlsSettingsDefault" tp:name-for-bindings="getTlsSettingsDefault">
<annotation name="com.trolltech.QtDBus.QtTypeName.Out0" value="MapStringString"/>
<tp:docstring>
+ Get default TLS setting for new accounts
</tp:docstring>
<arg type="a{ss}" name="details" direction="out">
- <tp:docstring>
- </tp:docstring>
+ <tp:docstring>
+ A hash table containing details. Refer
+ to <tp:member-ref>getTlsSettings</tp:member-ref>
+ for possible keys
+ </tp:docstring>
</arg>
</method>
<method name="getTlsSettings" tp:name-for-bindings="getTlsSettings">
<tp:docstring>
+ Get current TLS setting for a specific account
</tp:docstring>
<arg type="s" name="accountID" direction="in">
- <tp:docstring>
- </tp:docstring>
+ <tp:docstring>
+ The account ID
+ </tp:docstring>
</arg>
<annotation name="com.trolltech.QtDBus.QtTypeName.Out0" value="MapStringString"/>
<arg type="a{ss}" name="details" direction="out">
- <tp:docstring>
- </tp:docstring>
+ <tp:docstring>
+ A hash table containing details with key:
+ <ul>
+ <li>"TLS.listenerPort": valid numerical port</li>
+ <li>"TLS.enable": true/false</li>
+ <li>"TLS.certificateListFile": A valid path to
+ a .pem file containing CA certificate</li>
+ <li>"TLS.certificateFile": A valid path to a
+ file containing the public end-point
+ certificate (optional)</li>
+ <li>"TLS.privateKeyFile": A valid path to a
+ file containing the public end-point private
+ key (optional)</li>
+ <li>"TLS.password": Public end-point private
+ key password (optional)</li>
+ <li>"TLS.method": (TLSv1, SSLv1, SSLv2,
+ SSLv3, SSLv23)</li>
+ </ul>
+ </tp:docstring>
</arg>
</method>
<method name="setTlsSettings" tp:name-for-bindings="setTlsSettings">
<tp:docstring>
+ Update TLS setting for a specific account
</tp:docstring>
<annotation name="com.trolltech.QtDBus.QtTypeName.In1" value="MapStringString"/>
<arg type="s" name="accountID" direction="in">
- <tp:docstring>
- </tp:docstring>
+ <tp:docstring>
+ The account ID
+ </tp:docstring>
</arg>
<arg type="a{ss}" name="details" direction="in">
- <tp:docstring>
- </tp:docstring>
+ <tp:docstring>
+ A hash table containing details. Refer
+ to <tp:member-ref>getTlsSettings</tp:member-ref>
+ for possible keys.
+ </tp:docstring>
</arg>
</method>
<method name="getAddrFromInterfaceName" tp:name-for-bindings="getAddrFromInterfaceName">
<tp:docstring>
+ Resolve interface IPv4 address provided its name.
</tp:docstring>
<arg type="s" name="interface" direction="in">
- <tp:docstring>
- </tp:docstring>
+ <tp:docstring>
+ Interface name
+ </tp:docstring>
</arg>
<arg type="s" name="address" direction="out">
- <tp:docstring>
- </tp:docstring>
+ <tp:docstring>
+ Interface IPv4 address
+ </tp:docstring>
</arg>
</method>
<method name="getAllIpInterface" tp:name-for-bindings="getAllIpInterface">
<tp:docstring>
+ Provide a list of IP interface's IPv4 address.
</tp:docstring>
<annotation name="com.trolltech.QtDBus.QtTypeName.Out0" value="VectorString"/>
<arg type="as" name="list" direction="out">
- <tp:docstring>
- </tp:docstring>
+ <tp:docstring>
+ A list of interface's IPv4 address
+ </tp:docstring>
</arg>
</method>
<method name="getAllIpInterfaceByName" tp:name-for-bindings="getAllIpInterfaceByName">
<tp:docstring>
+ Provide a list of IP interface's name: default
+ (0.0.0.0), lo, eth0 ...
</tp:docstring>
<annotation name="com.trolltech.QtDBus.QtTypeName.Out0" value="VectorString"/>
<arg type="as" name="list" direction="out">
- <tp:docstring>
- </tp:docstring>
+ <tp:docstring>
+ A list of interface's name
+ </tp:docstring>
</arg>
</method>
diff --git a/sflphone-client-gnome/src/dbus/dbus.c b/sflphone-client-gnome/src/dbus/dbus.c
index b8c37a8fce6b2df850543205b96448f118c126ea..8e5cd1d18b0116dc581cc3d9ac90b1697cefaa38 100644
--- a/sflphone-client-gnome/src/dbus/dbus.c
+++ b/sflphone-client-gnome/src/dbus/dbus.c
@@ -1308,44 +1308,27 @@ dbus_set_active_codec_list(const gchar** list, const gchar *accountID)
}
}
-/**
- * Get a list of input supported audio plugins
- */
-gchar**
-dbus_get_input_audio_plugin_list()
-{
- gchar** array;
- GError* error = NULL;
- org_sflphone_SFLphone_ConfigurationManager_get_input_audio_plugin_list(
- configurationManagerProxy, &array, &error);
- if (error)
- {
- ERROR("Failed to call get_input_audio_plugin_list() on ConfigurationManager: %s", error->message);
- g_error_free(error);
- }
- return array;
-}
/**
* Get a list of output supported audio plugins
*/
gchar**
-dbus_get_output_audio_plugin_list()
+dbus_get_audio_plugin_list()
{
gchar** array;
GError* error = NULL;
- if (!org_sflphone_SFLphone_ConfigurationManager_get_output_audio_plugin_list(
+ if (!org_sflphone_SFLphone_ConfigurationManager_get_audio_plugin_list(
configurationManagerProxy, &array, &error))
{
if (error->domain == DBUS_GERROR && error->code
== DBUS_GERROR_REMOTE_EXCEPTION)
{
- ERROR ("Caught remote method (get_output_audio_plugin_list) exception %s: %s", dbus_g_error_get_name(error), error->message);
+ ERROR ("Caught remote method (get_output_plugin_list) exception %s: %s", dbus_g_error_get_name(error), error->message);
}
else
{
- ERROR("Error while calling get_out_audio_plugin_list: %s", error->message);
+ ERROR("Error while calling get_out_plugin_list: %s", error->message);
}
g_error_free(error);
return NULL;
@@ -1417,37 +1400,53 @@ dbus_set_audio_output_device(const int index)
}
/**
- * Get all input devices index supported by current audio manager
+ * Set audio input device from its index
*/
-gchar**
-dbus_get_audio_input_device_list()
+void
+dbus_set_audio_input_device(const int index)
{
- gchar** array;
GError* error = NULL;
- org_sflphone_SFLphone_ConfigurationManager_get_audio_input_device_list(
- configurationManagerProxy, &array, &error);
+ org_sflphone_SFLphone_ConfigurationManager_set_audio_input_device(
+ configurationManagerProxy, index, &error);
if (error)
{
- ERROR("Failed to call get_audio_input_device_list() on ConfigurationManager: %s", error->message);
+ ERROR("Failed to call set_audio_input_device() on ConfigurationManager: %s", error->message);
g_error_free(error);
}
- return array;
}
/**
- * Set audio input device from its index
+ * Set adio ringtone device from its index
*/
void
-dbus_set_audio_input_device(const int index)
+dbus_set_audio_ringtone_device(const int index)
{
GError* error = NULL;
- org_sflphone_SFLphone_ConfigurationManager_set_audio_input_device(
+ org_sflphone_SFLphone_ConfigurationManager_set_audio_ringtone_device(
configurationManagerProxy, index, &error);
+ if(error)
+ {
+ ERROR("Failed to call set_audio_ringtone_device() on ConfigurationManager: %s", error->message);
+ g_error_free(error);
+ }
+}
+
+/**
+ * Get all input devices index supported by current audio manager
+ */
+gchar**
+dbus_get_audio_input_device_list()
+{
+ gchar** array;
+ GError* error = NULL;
+ org_sflphone_SFLphone_ConfigurationManager_get_audio_input_device_list(
+ configurationManagerProxy, &array, &error);
if (error)
{
- ERROR("Failed to call set_audio_input_device() on ConfigurationManager: %s", error->message);
+ ERROR("Failed to call get_audio_input_device_list() on ConfigurationManager: %s", error->message);
g_error_free(error);
}
+ return array;
}
/**
diff --git a/sflphone-client-gnome/src/dbus/dbus.h b/sflphone-client-gnome/src/dbus/dbus.h
index 53e4747cf975300324fe4d0135e2516bd80f6d5c..cf75d9ce98095092c525900676d5e20a1a05f606 100644
--- a/sflphone-client-gnome/src/dbus/dbus.h
+++ b/sflphone-client-gnome/src/dbus/dbus.h
@@ -251,17 +251,11 @@ void dbus_set_active_codec_list (const gchar** list, const gchar*);
*/
gchar* dbus_get_current_codec_name(const callable_obj_t * c);
-/**
- * ConfigurationManager - Get the list of available input audio plugins
- * @return gchar** The list of plugins
- */
-gchar** dbus_get_input_audio_plugin_list();
-
/**
* ConfigurationManager - Get the list of available output audio plugins
* @return gchar** The list of plugins
*/
-gchar** dbus_get_output_audio_plugin_list();
+gchar** dbus_get_audio_plugin_list();
/**
* ConfigurationManager - Select an input audio plugin
diff --git a/sflphone-common/src/audio/alsa/alsalayer.cpp b/sflphone-common/src/audio/alsa/alsalayer.cpp
index d20f3d09eafe2e3b3d469b7178657a579bf1cdbe..d5eea5aef0846869e9e64bef3325277e65d039f8 100644
--- a/sflphone-common/src/audio/alsa/alsalayer.cpp
+++ b/sflphone-common/src/audio/alsa/alsalayer.cpp
@@ -1,6 +1,7 @@
/*
* Copyright (C) 2004, 2005, 2006, 2009, 2008, 2009, 2010 Savoir-Faire Linux Inc.
* Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
+ * Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
*
* 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
@@ -38,6 +39,7 @@ int framesPerBufferAlsa = 2048;
AlsaLayer::AlsaLayer (ManagerImpl* manager)
: AudioLayer (manager , ALSA)
, _PlaybackHandle (NULL)
+ , _RingtoneHandle (NULL)
, _CaptureHandle (NULL)
, _periodSize()
, _audioPlugin()
@@ -97,60 +99,47 @@ AlsaLayer::closeLayer()
/* Then close the audio devices */
closeCaptureStream();
-
closePlaybackStream();
- _CaptureHandle = 0;
-
- _PlaybackHandle = 0;
+ _CaptureHandle = NULL;
+ _PlaybackHandle = NULL;
+ _RingtoneHandle = NULL;
return true;
}
bool
-AlsaLayer::openDevice (int indexIn, int indexOut, int sampleRate, int frameSize, int stream , std::string plugin)
+AlsaLayer::openDevice (int indexIn, int indexOut, int indexRing, int sampleRate, int frameSize, int stream , std::string plugin)
{
/* Close the devices before open it */
if (stream == SFL_PCM_BOTH && is_capture_open() == true && is_playback_open() == true) {
closeCaptureStream();
closePlaybackStream();
- } else if ( (stream == SFL_PCM_CAPTURE || stream == SFL_PCM_BOTH) && is_capture_open() == true)
+ } else if((stream == SFL_PCM_CAPTURE || stream == SFL_PCM_BOTH) && is_capture_open() == true)
closeCaptureStream ();
- else if ( (stream == SFL_PCM_PLAYBACK || stream == SFL_PCM_BOTH) && is_playback_open () == true)
+ else if((stream == SFL_PCM_PLAYBACK || stream == SFL_PCM_BOTH) && is_playback_open () == true)
closePlaybackStream ();
-
_indexIn = indexIn;
-
_indexOut = indexOut;
+ _indexRing = indexRing;
_audioSampleRate = sampleRate;
-
_frameSize = frameSize;
_audioPlugin = std::string (plugin);
- _debugAlsa (" Setting AlsaLayer: device in=%2d, out=%2d", _indexIn, _indexOut);
-
+ _debugAlsa (" Setting AlsaLayer: device in=%2d, out=%2d, ring=%2d", _indexIn, _indexOut, _indexRing);
_debugAlsa (" : alsa plugin=%s", _audioPlugin.c_str());
-
_debugAlsa (" : nb channel in=%2d, out=%2d", _inChannel, _outChannel);
-
_debugAlsa (" : sample rate=%5d, format=%s", _audioSampleRate, SFLDataFormatString);
_audioThread = NULL;
- ost::MutexLock lock (_mutex);
-
- std::string pcmp = buildDeviceTopo (plugin, indexOut, 0);
-
- std::string pcmc = buildDeviceTopo (plugin, indexIn, 0);
-
// use 1 sec buffer for resampling
_converter = new SamplerateConverter (_audioSampleRate, 1000);
- // open_device (pcmp, pcmc, stream);
- return true; // open_device (pcmp, pcmc, stream);
+ return true;
}
void
@@ -158,20 +147,39 @@ AlsaLayer::startStream (void)
{
_debug ("AlsaLayer:: startStream");
- std::string pcmp = buildDeviceTopo (_audioPlugin, _indexOut, 0);
- std::string pcmc = buildDeviceTopo (_audioPlugin, _indexIn, 0);
+ if(is_playback_running() && is_capture_running() )
+ return;
+
+ std::string pcmp;
+ std::string pcmr;
+ std::string pcmc;
+
+ if(_audioPlugin == PCM_DMIX_DSNOOP) {
+ pcmp = buildDeviceTopo (PCM_DMIX, _indexOut, 0);
+ pcmr = buildDeviceTopo (PCM_DMIX, _indexRing, 0);
+ pcmc = buildDeviceTopo(PCM_DSNOOP, _indexIn, 0);
+ }
+ else {
+ pcmp = buildDeviceTopo (_audioPlugin, _indexOut, 0);
+ pcmr = buildDeviceTopo (_audioPlugin, _indexRing, 0);
+ pcmc = buildDeviceTopo(_audioPlugin, _indexIn, 0);
+ }
+
+ _debug("pcmp: %s, index %d", pcmp.c_str(), _indexOut);
+ _debug("pcmr: %s, index %d", pcmr.c_str(), _indexRing);
+ _debug("pcmc: %s, index %d", pcmc.c_str(), _indexIn);
if (!is_playback_open()) {
- open_device (pcmp, pcmc, SFL_PCM_PLAYBACK);
+ open_device (pcmp, pcmc, pcmr, SFL_PCM_PLAYBACK);
}
if (!is_capture_open()) {
- open_device (pcmp, pcmc, SFL_PCM_CAPTURE);
+ open_device (pcmp, pcmc, pcmr, SFL_PCM_CAPTURE);
}
prepareCaptureStream ();
-
preparePlaybackStream ();
+
startCaptureStream ();
startPlaybackStream ();
@@ -211,9 +219,12 @@ AlsaLayer::stopStream (void)
}
closeCaptureStream ();
-
closePlaybackStream ();
+ _PlaybackHandle = NULL;
+ _CaptureHandle = NULL;
+ _RingtoneHandle = NULL;
+
/* Flush the ring buffers */
flushUrgent ();
flushMain ();
@@ -299,11 +310,19 @@ void AlsaLayer::stopPlaybackStream (void)
{
int err;
+ if(_RingtoneHandle && is_playback_running()) {
+ _debug("Audio: Stop ALSA ringtone");
+
+ if( (err = snd_pcm_drop(_RingtoneHandle)) < 0) {
+ _debug("Audio: Error: Stop ALSA ringtone: %s", snd_strerror(err));
+ }
+ }
+
if (_PlaybackHandle && is_playback_running()) {
- _debug ("AlsaLayer:: stop ALSA playback");
+ _debug ("Audio: Stop ALSA playback");
if ( (err = snd_pcm_drop (_PlaybackHandle)) < 0)
- _debug ("Error stopping ALSA playback: %s", snd_strerror (err));
+ _debug ("Audio: Error: Stopping ALSA playback: %s", snd_strerror (err));
else
stop_playback ();
}
@@ -317,14 +336,22 @@ void AlsaLayer::closePlaybackStream (void)
if (is_playback_prepared() == true && is_playback_running() == true)
stopPlaybackStream ();
+
if (is_playback_open()) {
- _debug ("AlsaLayer:: close ALSA playback");
+ _debug("Audio: Close ALSA playback");
+
+ if(_RingtoneHandle) {
+ if((err = snd_pcm_close(_RingtoneHandle)) < 0) {
+ _warn("Audio: Error: Closing ALSA ringtone: %s", snd_strerror(err));
+ }
+ }
if ( (err = snd_pcm_close (_PlaybackHandle)) < 0)
- _debug ("Error closing ALSA playback: %s", snd_strerror (err));
+ _warn("Audio: Error: Closing ALSA playback: %s", snd_strerror (err));
else
close_playback ();
}
+
}
void AlsaLayer::startPlaybackStream (void)
@@ -508,46 +535,59 @@ bool AlsaLayer::alsa_set_params (snd_pcm_t *pcm_handle, int type, int rate)
bool
-AlsaLayer::open_device (std::string pcm_p, std::string pcm_c, int flag)
+AlsaLayer::open_device (std::string pcm_p, std::string pcm_c, std::string pcm_r, int flag)
{
int err;
if (flag == SFL_PCM_BOTH || flag == SFL_PCM_PLAYBACK) {
- _debug ("AlsaLayer:: open playback device");
- // if((err = snd_pcm_open(&_PlaybackHandle, pcm_p.c_str(), SND_PCM_STREAM_PLAYBACK, 0 )) < 0){
+ _debug ("Audio: Open playback device (and ringtone)");
- if ( (err = snd_pcm_open (&_PlaybackHandle, pcm_p.c_str(), SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
- _debugAlsa ("Error while opening playback device %s", pcm_p.c_str());
+ if ((err = snd_pcm_open(&_PlaybackHandle, pcm_p.c_str(), SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
+ _warn("Audio: Error while opening playback device %s", pcm_p.c_str());
setErrorMessage (ALSA_PLAYBACK_DEVICE);
close_playback ();
return false;
}
if (!alsa_set_params (_PlaybackHandle, 1, getSampleRate())) {
- _debug ("playback failed");
+ _warn ("Audio: Error: Playback failed");
snd_pcm_close (_PlaybackHandle);
close_playback ();
return false;
}
+ if (getIndexOut() != getIndexRing()) {
+
+ if((err = snd_pcm_open(&_RingtoneHandle, pcm_r.c_str(), SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
+ _warn("Audio: Error: Opening ringtone device %s", pcm_r.c_str());
+ // setErrorMessage(ALSA_RINGTONE_DEVICE);
+ }
+
+ if(!alsa_set_params(_RingtoneHandle, 1, getSampleRate())) {
+ _warn("Audio: Error: Ringtone failed");
+ snd_pcm_close(_RingtoneHandle);
+
+ }
+ }
+
open_playback ();
}
if (flag == SFL_PCM_BOTH || flag == SFL_PCM_CAPTURE) {
- _debug ("AlsaLayer:: open capture device");
+ _debug ("Audio: Open capture device");
if ( (err = snd_pcm_open (&_CaptureHandle, pcm_c.c_str(), SND_PCM_STREAM_CAPTURE, 0)) < 0) {
- _debugAlsa ("Error while opening capture device %s", pcm_c.c_str());
+ _warn("Audio: Error: Opening capture device %s", pcm_c.c_str());
setErrorMessage (ALSA_CAPTURE_DEVICE);
close_capture ();
return false;
}
if (!alsa_set_params (_CaptureHandle, 0, 8000)) {
- _debug ("capture failed");
+ _warn("Audio: Error: Capture failed");
snd_pcm_close (_CaptureHandle);
close_capture ();
return false;
@@ -558,32 +598,25 @@ AlsaLayer::open_device (std::string pcm_p, std::string pcm_c, int flag)
// prepare_capture ();
}
- /* Start the secondary audio thread for callbacks */
- /*
- try {
- _audioThread->start();
- } catch (...) {
- _debugException ("Fail to start audio thread");
- }
- */
+
return true;
}
//TODO first frame causes broken pipe (underrun) because not enough data are send --> make the handle wait to be ready
int
-AlsaLayer::write (void* buffer, int length)
+AlsaLayer::write (void* buffer, int length, snd_pcm_t * handle)
{
if (_trigger_request == true) {
_trigger_request = false;
startPlaybackStream ();
}
- snd_pcm_uframes_t frames = snd_pcm_bytes_to_frames (_PlaybackHandle, length);
+ snd_pcm_uframes_t frames = snd_pcm_bytes_to_frames (handle, length);
int err;
- if ( (err=snd_pcm_writei (_PlaybackHandle , buffer , frames)) <0) {
+ if ( (err=snd_pcm_writei (handle, buffer , frames)) <0) {
switch (err) {
case -EPIPE:
@@ -592,17 +625,17 @@ AlsaLayer::write (void* buffer, int length)
case -EIO:
//_debugAlsa(" XRUN playback ignored (%s)", snd_strerror(err));
- handle_xrun_playback();
+ handle_xrun_playback(handle);
- if (snd_pcm_writei (_PlaybackHandle , buffer , frames) <0)
- _debugAlsa ("XRUN handling failed");
+ if (snd_pcm_writei (handle, buffer , frames) <0)
+ _debugAlsa ("Audio: XRUN handling failed");
_trigger_request = true;
break;
default:
- _debugAlsa ("Write error unknown - dropping frames **********************************: %s", snd_strerror (err));
+ _debugAlsa ("Audio: Write error unknown - dropping frames: %s", snd_strerror (err));
stopPlaybackStream ();
@@ -635,14 +668,14 @@ AlsaLayer::read (void* buffer, int toCopy)
case -ESTRPIPE:
case -EIO:
- _debugAlsa (" XRUN capture ignored (%s)", snd_strerror (samples));
+ _debugAlsa ("Audio: XRUN capture ignored (%s)", snd_strerror (samples));
handle_xrun_capture();
//samples = snd_pcm_readi( _CaptureHandle, buffer, frames);
//if (samples<0) samples=0;
break;
case EPERM:
- _debugAlsa (" Capture EPERM (%s)", snd_strerror (samples));
+ _debugAlsa ("Audio: Capture EPERM (%s)", snd_strerror (samples));
prepareCaptureStream ();
startCaptureStream ();
break;
@@ -662,7 +695,7 @@ AlsaLayer::read (void* buffer, int toCopy)
void
AlsaLayer::handle_xrun_capture (void)
{
- _debugAlsa ("handle_xrun_capture");
+ _debugAlsa ("Audio: Handle xrun capture");
snd_pcm_status_t* status;
snd_pcm_status_alloca (&status);
@@ -676,23 +709,25 @@ AlsaLayer::handle_xrun_capture (void)
startCaptureStream ();
}
} else
- _debugAlsa (" Get status failed");
+ _debugAlsa ("Audio: Get status failed");
}
void
-AlsaLayer::handle_xrun_playback (void)
+AlsaLayer::handle_xrun_playback (snd_pcm_t *handle)
{
- _debugAlsa ("AlsaLayer:: handle_xrun_playback");
+ _debugAlsa ("Audio: Handle xrun playback");
int state;
snd_pcm_status_t* status;
snd_pcm_status_alloca (&status);
- if ( (state = snd_pcm_status (_PlaybackHandle, status)) < 0) _debugAlsa (" Error: Cannot get playback handle status (%s)" , snd_strerror (state));
+ if ( (state = snd_pcm_status (handle, status)) < 0)
+ _debugAlsa ("Audio: Error: Cannot get playback handle status (%s)" , snd_strerror (state));
else {
state = snd_pcm_status_get_state (status);
if (state == SND_PCM_STATE_XRUN) {
+ _debug("Audio: audio device in state SND_PCM_STATE_XRUN, restart device");
stopPlaybackStream ();
preparePlaybackStream ();
@@ -705,10 +740,10 @@ AlsaLayer::handle_xrun_playback (void)
std::string
AlsaLayer::buildDeviceTopo (std::string plugin, int card, int subdevice)
{
- std::string pcm = plugin;
std::stringstream ss,ss1;
+ std::string pcm = plugin;
- if (pcm == "default" || pcm == "pulse")
+ if (pcm == PCM_DEFAULT)
return pcm;
ss << card;
@@ -723,6 +758,8 @@ AlsaLayer::buildDeviceTopo (std::string plugin, int card, int subdevice)
pcm.append (ss1.str());
}
+ _debug("Audio: Device topo: %s", pcm.c_str());
+
return pcm;
}
@@ -732,6 +769,8 @@ AlsaLayer::getSoundCardsInfo (int stream)
std::vector<std::string> cards_id;
HwIDPair p;
+ _debug("Audio: Get sound cards info: ");
+
snd_ctl_t* handle;
snd_ctl_card_info_t *info;
snd_pcm_info_t* pcminfo;
@@ -768,6 +807,7 @@ AlsaLayer::getSoundCardsInfo (int stream)
// The number of the sound card is associated with a string description
p = HwIDPair (numCard , description);
IDSoundCards.push_back (p);
+
}
}
@@ -835,24 +875,32 @@ void AlsaLayer::audioCallback (void)
spkrVolume = _manager->getSpkrVolume();
micVolume = _manager->getMicVolume();
- /*
- int writeableSize = snd_pcm_avail_update(_PlaybackHandle);
- _debug("writeableSize %i", writeableSize);
- */
+ tone = _manager->getTelephoneTone();
+ file_tone = _manager->getTelephoneFile();
// AvailForGet tell the number of chars inside the buffer
// framePerBuffer are the number of data for one channel (left)
urgentAvailBytes = _urgentRingBuffer.AvailForGet();
+ // toGet = framesPerBufferAlsa;
+ // maxBytes = toGet * sizeof (SFLDataFormat);
+
+ if(!_PlaybackHandle)
+ return;
+
+ int playbackAvailSmpl = snd_pcm_avail_update(_PlaybackHandle);
+ int playbackAvailBytes = playbackAvailSmpl*sizeof(SFLDataFormat);
+ // _debug("PLAYBACK: %d", playbackAvailSmpl);
+
if (urgentAvailBytes > 0) {
// Urgent data (dtmf, incoming call signal) come first.
- toGet = (urgentAvailBytes < (int) (framesPerBufferAlsa * sizeof (SFLDataFormat))) ? urgentAvailBytes : framesPerBufferAlsa * sizeof (SFLDataFormat);
- out = (SFLDataFormat*) malloc (toGet * sizeof (SFLDataFormat));
+ toGet = (urgentAvailBytes < (int) (playbackAvailBytes)) ? urgentAvailBytes : playbackAvailBytes;
+ out = (SFLDataFormat*) malloc (toGet);
_urgentRingBuffer.Get (out, toGet, spkrVolume);
/* Play the sound */
- write (out, toGet);
+ write (out, toGet, _PlaybackHandle);
free (out);
out=0;
@@ -862,55 +910,44 @@ void AlsaLayer::audioCallback (void)
} else {
- tone = _manager->getTelephoneTone();
- file_tone = _manager->getTelephoneFile();
+ if (tone) {
- toGet = framesPerBufferAlsa;
- maxBytes = toGet * sizeof (SFLDataFormat);
-
- if (tone != 0) {
-
- out = (SFLDataFormat*) malloc (maxBytes * sizeof (SFLDataFormat));
- tone->getNext (out, toGet, spkrVolume);
- write (out , maxBytes);
+ out = (SFLDataFormat *) malloc (playbackAvailBytes);
+ tone->getNext (out, playbackAvailSmpl, spkrVolume);
+ write (out , playbackAvailBytes, _PlaybackHandle);
free (out);
out = 0;
+
+ }
+ else if (file_tone && !_RingtoneHandle) {
- } else if (file_tone != 0) {
+ out = (SFLDataFormat *) malloc (playbackAvailBytes);
+ file_tone->getNext (out, playbackAvailSmpl, spkrVolume);
+ write (out, playbackAvailBytes, _PlaybackHandle);
- out = (SFLDataFormat*) malloc (maxBytes * sizeof (SFLDataFormat));
- file_tone->getNext (out, toGet, spkrVolume);
- write (out , maxBytes);
-
- free (out);
- out = 0;
+ free (out);
+ out = NULL;
- } else {
+ } else {
// If nothing urgent, play the regular sound samples
int _mainBufferSampleRate = getMainBuffer()->getInternalSamplingRate();
- int maxNbSamplesToGet = 0;
- int maxNbBytesToGet = 0;
+ int maxNbSamplesToGet = playbackAvailSmpl;
+ int maxNbBytesToGet = playbackAvailBytes;
// Compute maximal value to get into the ring buffer
if (_mainBufferSampleRate && ( (int) _audioSampleRate != _mainBufferSampleRate)) {
double upsampleFactor = (double) _audioSampleRate / _mainBufferSampleRate;
+ maxNbSamplesToGet = (int) ( (double) playbackAvailSmpl / upsampleFactor);
+ maxNbBytesToGet = maxNbSamplesToGet * sizeof (SFLDataFormat);
- maxNbSamplesToGet = (int) ( (double) framesPerBufferAlsa / upsampleFactor);
-
-
- } else {
+ }
- maxNbSamplesToGet = framesPerBufferAlsa;
-
- }
-
- maxNbBytesToGet = maxNbSamplesToGet * sizeof (SFLDataFormat);
normalAvailBytes = getMainBuffer()->availForGet();
toGet = (normalAvailBytes < (int) maxNbBytesToGet) ? normalAvailBytes : maxNbBytesToGet;
@@ -924,7 +961,7 @@ void AlsaLayer::audioCallback (void)
if (_mainBufferSampleRate && ( (int) _audioSampleRate != _mainBufferSampleRate)) {
- rsmpl_out = (SFLDataFormat*) malloc (framesPerBufferAlsa * sizeof (SFLDataFormat));
+ rsmpl_out = (SFLDataFormat*) malloc (playbackAvailBytes*2);
// Do sample rate conversion
int nb_sample_down = toGet / sizeof (SFLDataFormat);
@@ -933,25 +970,25 @@ void AlsaLayer::audioCallback (void)
- write (rsmpl_out, nbSample*sizeof (SFLDataFormat));
+ write (rsmpl_out, nbSample*sizeof (SFLDataFormat), _PlaybackHandle);
free (rsmpl_out);
rsmpl_out = 0;
} else {
- write (out, toGet);
+ write (out, toGet, _PlaybackHandle);
}
} else {
- if ( (tone == 0) && (file_tone == 0)) {
+ if (!tone && !file_tone) {
- SFLDataFormat* zeros = (SFLDataFormat*) malloc (framesPerBufferAlsa * sizeof (SFLDataFormat));
+ SFLDataFormat *zeros = (SFLDataFormat*)malloc(playbackAvailBytes);
- bzero (zeros, framesPerBufferAlsa * sizeof (SFLDataFormat));
- write (zeros, framesPerBufferAlsa * sizeof (SFLDataFormat));
+ bzero (zeros, playbackAvailBytes);
+ write (zeros, playbackAvailBytes, _PlaybackHandle);
free (zeros);
}
@@ -966,16 +1003,40 @@ void AlsaLayer::audioCallback (void)
}
+ if (file_tone && _RingtoneHandle) {
+
+ int ringtoneAvailSmpl = snd_pcm_avail_update(_RingtoneHandle);
+ int ringtoneAvailBytes = ringtoneAvailSmpl*sizeof(SFLDataFormat);
+
+ // _debug("RINGTONE: %d", ringtoneAvailSmpl);
+
+ out = (SFLDataFormat *) malloc(ringtoneAvailBytes);
+ file_tone->getNext (out, ringtoneAvailSmpl, spkrVolume);
+ write (out, ringtoneAvailBytes, _RingtoneHandle);
+
+ free (out);
+ out = NULL;
+
+ } else if (_RingtoneHandle) {
+
+ int ringtoneAvailSmpl = snd_pcm_avail_update(_RingtoneHandle);
+ int ringtoneAvailBytes = ringtoneAvailSmpl*sizeof(SFLDataFormat);
+
+ out = (SFLDataFormat *) malloc(ringtoneAvailBytes);
+ memset(out, 0, ringtoneAvailBytes);
+ write(out, ringtoneAvailBytes, _RingtoneHandle);
+
+ free(out);
+ out = NULL;
+ }
+
// Additionally handle the mic's audio stream
int micAvailBytes;
-
int micAvailPut;
-
int toPut;
SFLDataFormat* in;
-
// snd_pcm_sframes_t micAvailAlsa;
in = 0;
diff --git a/sflphone-common/src/audio/alsa/alsalayer.h b/sflphone-common/src/audio/alsa/alsalayer.h
index f2736a800a141558df2403f5af13e2a1f3f76f36..0346e6b98adb46813ca3e98794e9816b784a1086 100644
--- a/sflphone-common/src/audio/alsa/alsalayer.h
+++ b/sflphone-common/src/audio/alsa/alsalayer.h
@@ -1,6 +1,7 @@
/*
* Copyright (C) 2004, 2005, 2006, 2009, 2008, 2009, 2010 Savoir-Faire Linux Inc.
* Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
+ * Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
*
* 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
@@ -76,7 +77,7 @@ class AlsaLayer : public AudioLayer {
* SFL_PCM_BOTH
* @param plugin The alsa plugin ( dmix , default , front , surround , ...)
*/
- bool openDevice(int indexIn, int indexOut, int sampleRate, int frameSize, int stream, std::string plugin);
+ bool openDevice(int indexIn, int indexOut, int indexRing, int sampleRate, int frameSize, int stream, std::string plugin);
/**
* Start the capture stream and prepare the playback stream.
@@ -205,7 +206,7 @@ class AlsaLayer : public AudioLayer {
* @return true if successful
* false otherwise
*/
- bool open_device( std::string pcm_p, std::string pcm_c, int flag);
+ bool open_device( std::string pcm_p, std::string pcm_c, std::string pcm_r, int flag);
bool alsa_set_params( snd_pcm_t *pcm_handle, int type, int rate );
@@ -216,7 +217,7 @@ class AlsaLayer : public AudioLayer {
* @param length The size of the buffer
* @return int The number of frames actually copied
*/
- int write( void* buffer, int length);
+ int write( void* buffer, int length, snd_pcm_t *handle);
/**
* Read data from the internal ring buffer
@@ -226,8 +227,6 @@ class AlsaLayer : public AudioLayer {
* @return int The number of frames actually read
*/
int read( void* buffer, int toCopy);
-
-
/**
* Recover from XRUN state for capture
@@ -239,16 +238,22 @@ class AlsaLayer : public AudioLayer {
* Recover from XRUN state for playback
* ALSA Library API
*/
- void handle_xrun_playback( void );
-
+ void handle_xrun_playback( snd_pcm_t *handle );
+
void* adjustVolume( void* buffer , int len, int stream );
-/**
+ /**
* Handles to manipulate playback stream
* ALSA Library API
*/
snd_pcm_t* _PlaybackHandle;
+ /**
+ * Handles to manipulate ringtone stream
+ *
+ */
+ snd_pcm_t *_RingtoneHandle;
+
/**
* Handles to manipulate capture stream
* ALSA Library API
@@ -268,7 +273,7 @@ class AlsaLayer : public AudioLayer {
/** Vector to manage all soundcard index - description association of the system */
std::vector<HwIDPair> IDSoundCards;
- bool _is_prepared_playback;
+ bool _is_prepared_playback;
bool _is_prepared_capture;
bool _is_running_playback;
bool _is_running_capture;
diff --git a/sflphone-common/src/audio/audiolayer.h b/sflphone-common/src/audio/audiolayer.h
index 336e29f078ff2954161a81ed4f0d1937c5a31ebf..821e1ccde36d2d5e271bfe8df505b7c7975d783a 100644
--- a/sflphone-common/src/audio/audiolayer.h
+++ b/sflphone-common/src/audio/audiolayer.h
@@ -49,11 +49,9 @@
* @brief Main sound class. Manages the data transfers between the application and the hardware.
*/
-
class Recordable;
class ManagerImpl;
-
class AudioLayer {
private:
@@ -106,7 +104,7 @@ class AudioLayer {
* SFL_PCM_BOTH
* @param plugin The alsa plugin ( dmix , default , front , surround , ...)
*/
- virtual bool openDevice(int indexIn, int indexOut, int sampleRate, int frameSize, int stream , std::string plugin) = 0;
+ virtual bool openDevice(int indexIn, int indexOut, int indexRing, int sampleRate, int frameSize, int stream , std::string plugin) = 0;
/**
* Start the capture stream and prepare the playback stream.
@@ -174,6 +172,13 @@ class AudioLayer {
*/
int getIndexOut() { return _indexOut; }
+ /**
+ * Get the index of the audio card for ringtone (could be differnet from playback)
+ * @return int The index of the card used for ringtone
+ * 0 for the first available card on the system, 1 ...
+ */
+ int getIndexRing() {return _indexRing; }
+
/**
* Get the sample rate of the audio layer
* @return unsigned int The sample rate
@@ -278,6 +283,11 @@ class AudioLayer {
*/
int _indexOut;
+ /**
+ * Number of audio cards on which ringtone stream has been opened
+ */
+ int _indexRing;
+
/**
* Sample Rate SFLphone should send sound data to the sound card
* The value can be set in the user config file- now: 44100HZ
diff --git a/sflphone-common/src/audio/pulseaudio/audiostream.cpp b/sflphone-common/src/audio/pulseaudio/audiostream.cpp
index 8a246e5b38a46be146741ab3efa9b8182dc47c28..0b59c2e968ef5eb70a4d574fed00d6ac61af04e9 100644
--- a/sflphone-common/src/audio/pulseaudio/audiostream.cpp
+++ b/sflphone-common/src/audio/pulseaudio/audiostream.cpp
@@ -237,7 +237,7 @@ AudioStream::createStream (pa_context* c, std::string *deviceName)
attributes->minreq = (uint32_t) -1;
pa_threaded_mainloop_lock(_mainloop);
- pa_stream_connect_playback(s, NULL, attributes, (pa_stream_flags_t) (PA_STREAM_NOFLAGS), NULL, NULL);
+ pa_stream_connect_playback(s, NULL, attributes, (pa_stream_flags_t) (PA_STREAM_ADJUST_LATENCY|PA_STREAM_AUTO_TIMING_UPDATE), NULL, NULL);
pa_threaded_mainloop_unlock(_mainloop);
} else if (_streamType == UPLOAD_STREAM) {
diff --git a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
index f3decdac51166b2481bbfc264f2f5c18e1cfc892..c6ecae78ad178d3354b063b46365b5d8750aee9b 100644
--- a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
+++ b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
@@ -262,42 +262,15 @@ PulseLayer::~PulseLayer (void)
void
PulseLayer::openLayer (void)
{
- if (!is_started) {
+ if (!is_started) {
- _info("Audio: Open layer");
+ _info("Audio: Open layer");
- if (!m) {
+ connectPulseAudioServer();
- _info("Audio: Creating PulseAudio mainloop");
- if (!(m = pa_threaded_mainloop_new()))
- _warn ("Audio: Error: while creating pulseaudio mainloop");
+ is_started = true;
- if (pa_threaded_mainloop_start (m) < 0) {
- _warn("Audio: Error: Failed to start pulseaudio mainloop");
- }
-
- assert(m);
- }
-
- if (!context) {
-
- _info("Audio: Creating new PulseAudio Context");
- pa_threaded_mainloop_lock (m);
-
- // Instanciate a context
- if (! (context = pa_context_new (pa_threaded_mainloop_get_api (m) , "SFLphone")))
- _warn ("Audio: Error: while creating pulseaudio context");
-
- pa_threaded_mainloop_unlock (m);
-
- assert (context);
- }
-
- // Create Streams
- connectPulseAudioServer();
-
- is_started = true;
- }
+ }
}
@@ -336,12 +309,39 @@ PulseLayer::connectPulseAudioServer (void)
pa_context_flags_t flag = PA_CONTEXT_NOAUTOSPAWN ;
- pa_threaded_mainloop_lock (m);
+ if (!m) {
+
+ // Instantiate a mainloop
+ _info("Audio: Creating PulseAudio mainloop");
+ if (!(m = pa_threaded_mainloop_new()))
+ _warn ("Audio: Error: while creating pulseaudio mainloop");
+
+ assert(m);
+ }
- _info("Audio: Connect the context to the server");
- pa_context_connect (context, NULL , flag , NULL);
+ if(!context) {
+
+ // Instantiate a context
+ if (! (context = pa_context_new (pa_threaded_mainloop_get_api (m) , "SFLphone")))
+ _warn ("Audio: Error: while creating pulseaudio context");
+
+ assert(context);
+ }
+ // set context state callback before starting the mainloop
pa_context_set_state_callback (context, context_state_callback, this);
+
+ _info("Audio: Connect the context to the server");
+ if (pa_context_connect (context, NULL , flag , NULL) < 0) {
+ _warn("Audio: Error: Could not connect context to the server");
+ }
+
+ // Lock the loop before starting it
+ pa_threaded_mainloop_lock (m);
+
+ if (pa_threaded_mainloop_start (m) < 0)
+ _warn("Audio: Error: Failed to start pulseaudio mainloop");
+
pa_threaded_mainloop_wait(m);
@@ -395,7 +395,7 @@ void PulseLayer::context_state_callback (pa_context* c, void* user_data)
}
}
-bool PulseLayer::openDevice (int indexIn UNUSED, int indexOut UNUSED, int sampleRate, int frameSize , int stream UNUSED, std::string plugin UNUSED)
+bool PulseLayer::openDevice (int indexIn UNUSED, int indexOut UNUSED, int indexRing UNUSED, int sampleRate, int frameSize , int stream UNUSED, std::string plugin UNUSED)
{
_audioSampleRate = sampleRate;
_frameSize = frameSize;
@@ -743,7 +743,6 @@ void PulseLayer::writeToSpeaker (void)
} else {
AudioLoop* tone = _manager->getTelephoneTone();
- AudioLoop* file_tone = _manager->getTelephoneFile();
// flush remaining samples in _urgentRingBuffer
flushUrgent();
@@ -760,22 +759,6 @@ void PulseLayer::writeToSpeaker (void)
pa_xfree (out);
}
- //}
-
- // else if (file_tone != 0) {
-
- /*
- if (playback->getStreamState() == PA_STREAM_READY) {
-
- out = (SFLDataFormat*) pa_xmalloc (writeableSize);
- int copied = file_tone->getNext (out, writeableSize / sizeof (SFLDataFormat), 100);
-
- pa_stream_write (playback->pulseStream(), out, copied * sizeof (SFLDataFormat), NULL, 0, PA_SEEK_RELATIVE);
-
- pa_xfree (out);
-
- }
- */
} else {
@@ -842,7 +825,7 @@ void PulseLayer::writeToSpeaker (void)
} else {
- if ( (tone == 0) && (file_tone == 0)) {
+ if (tone == 0) {
SFLDataFormat* zeros = (SFLDataFormat*) pa_xmalloc (writeableSize);
@@ -927,27 +910,29 @@ void PulseLayer::ringtoneToSpeaker(void)
int writableSize = pa_stream_writable_size(ringtone->pulseStream());
- // _debug("writable size: %d", writableSize);
+ _debug("writable size: %d", writableSize);
if (file_tone) {
if(ringtone->getStreamState() == PA_STREAM_READY) {
- out = (SFLDataFormat*)pa_xmalloc(writableSize);
+ out = (SFLDataFormat *)pa_xmalloc(writableSize);
int copied = file_tone->getNext(out, writableSize/sizeof(SFLDataFormat), 100);
pa_stream_write(ringtone->pulseStream(), out, copied*sizeof(SFLDataFormat), NULL, 0, PA_SEEK_RELATIVE);
pa_xfree(out);
-
}
}
else {
- out = (SFLDataFormat*)pa_xmalloc(writableSize);
- memset(out, 0, writableSize);
- pa_stream_write(ringtone->pulseStream(), out, writableSize, NULL, 0, PA_SEEK_RELATIVE);
+ if(ringtone->getStreamState() == PA_STREAM_READY) {
+
+ out = (SFLDataFormat*)pa_xmalloc(writableSize);
+ memset(out, 0, writableSize);
+ pa_stream_write(ringtone->pulseStream(), out, writableSize, NULL, 0, PA_SEEK_RELATIVE);
- pa_xfree(out);
+ pa_xfree(out);
+ }
}
diff --git a/sflphone-common/src/audio/pulseaudio/pulselayer.h b/sflphone-common/src/audio/pulseaudio/pulselayer.h
index fd92103879cff11f1d12c4f24b901e3d9babe2a5..59ee2284d486c364ecfb3d363f5ea618214231da 100644
--- a/sflphone-common/src/audio/pulseaudio/pulselayer.h
+++ b/sflphone-common/src/audio/pulseaudio/pulselayer.h
@@ -76,7 +76,7 @@ class PulseLayer : public AudioLayer {
* SFL_PCM_BOTH
* @param plugin The alsa plugin ( dmix , default , front , surround , ...)
*/
- bool openDevice(int indexIn, int indexOut, int sampleRate, int frameSize , int stream, std::string plugin) ;
+ bool openDevice(int indexIn, int indexOut, int indexRing, int sampleRate, int frameSize , int stream, std::string plugin) ;
DeviceList* getSinkList(void) { return &_sinkList; }
diff --git a/sflphone-common/src/dbus/configurationmanager-introspec.xml b/sflphone-common/src/dbus/configurationmanager-introspec.xml
index 9c0b6f1c4fb35ef45d58924dc885ae0ca9acbf48..8a0c59c8b2b1aa69d46519915f26228a9d27584b 100644
--- a/sflphone-common/src/dbus/configurationmanager-introspec.xml
+++ b/sflphone-common/src/dbus/configurationmanager-introspec.xml
@@ -466,17 +466,7 @@
<!-- Audio devices methods -->
- <method name="getInputAudioPluginList" tp:name-for-bindings="getInputAudioPluginList">
- <tp:docstring>
- </tp:docstring>
- <annotation name="com.trolltech.QtDBus.QtTypeName.Out0" value="VectorString"/>
- <arg type="as" name="list" direction="out">
- <tp:docstring>
- </tp:docstring>
- </arg>
- </method>
-
- <method name="getOutputAudioPluginList" tp:name-for-bindings="getOutputAudioPluginList">
+ <method name="getAudioPluginList" tp:name-for-bindings="getAudioPluginList">
<tp:docstring>
</tp:docstring>
<annotation name="com.trolltech.QtDBus.QtTypeName.Out0" value="VectorString"/>
@@ -523,25 +513,35 @@
</arg>
</method>
- <method name="getAudioInputDeviceList" tp:name-for-bindings="getAudioInputDeviceList">
+ <method name="setAudioInputDevice" tp:name-for-bindings="setAudioInputDevice">
<tp:docstring>
</tp:docstring>
- <annotation name="com.trolltech.QtDBus.QtTypeName.Out0" value="VectorString"/>
- <arg type="as" name="list" direction="out">
+ <arg type="i" name="index" direction="in">
<tp:docstring>
</tp:docstring>
</arg>
</method>
- <method name="setAudioInputDevice" tp:name-for-bindings="setAudioInputDevice">
- <tp:docstring>
+ <method name="setAudioRingtoneDevice" tp:name-for-binding="setAudioRingtoneDevice">
+ <tp:docstring>
</tp:docstring>
<arg type="i" name="index" direction="in">
+ <tp:docstring>
+ </tp:docstring>
+ </arg>
+ </method>
+
+ <method name="getAudioInputDeviceList" tp:name-for-bindings="getAudioInputDeviceList">
+ <tp:docstring>
+ </tp:docstring>
+ <annotation name="com.trolltech.QtDBus.QtTypeName.Out0" value="VectorString"/>
+ <arg type="as" name="list" direction="out">
<tp:docstring>
</tp:docstring>
</arg>
</method>
+
<method name="getCurrentAudioDevicesIndex" tp:name-for-bindings="getCurrentAudioDevicesIndex">
<tp:docstring>
</tp:docstring>
diff --git a/sflphone-common/src/dbus/configurationmanager.cpp b/sflphone-common/src/dbus/configurationmanager.cpp
index 798bdaf58c204cee82011f22c7049c981333df24..b928dd36554fbd15000f639d1b3a93364716c58f 100644
--- a/sflphone-common/src/dbus/configurationmanager.cpp
+++ b/sflphone-common/src/dbus/configurationmanager.cpp
@@ -516,28 +516,13 @@ void ConfigurationManager::setActiveCodecList(
}
-// Audio devices related methods
-std::vector<std::string> ConfigurationManager::getInputAudioPluginList() {
-
- _debug("ConfigurationManager: Active codec list received");
-
- std::vector<std::string> v;
-
- v.push_back("default");
- v.push_back("surround40");
- v.push_back("plug:hw");
-
- return v;
-}
-
-
-
-std::vector<std::string> ConfigurationManager::getOutputAudioPluginList() {
+std::vector<std::string> ConfigurationManager::getAudioPluginList() {
std::vector<std::string> v;
v.push_back(PCM_DEFAULT);
- v.push_back(PCM_DMIX);
+ // v.push_back(PCM_DMIX);
+ v.push_back(PCM_DMIX_DSNOOP);
return v;
}
@@ -555,16 +540,20 @@ std::vector<std::string> ConfigurationManager::getAudioOutputDeviceList() {
return Manager::instance().getAudioOutputDeviceList();
}
-void ConfigurationManager::setAudioOutputDevice(const int32_t& index) {
- return Manager::instance().setAudioOutputDevice(index);
-}
-
std::vector<std::string> ConfigurationManager::getAudioInputDeviceList() {
return Manager::instance().getAudioInputDeviceList();
}
+void ConfigurationManager::setAudioOutputDevice(const int32_t& index) {
+ return Manager::instance().setAudioDevice(index, SFL_PCM_PLAYBACK);
+}
+
void ConfigurationManager::setAudioInputDevice(const int32_t& index) {
- return Manager::instance().setAudioInputDevice(index);
+ return Manager::instance().setAudioDevice(index, SFL_PCM_CAPTURE);
+}
+
+void ConfigurationManager::setAudioRingtoneDevice(const int32_t& index) {
+ return Manager::instance().setAudioDevice(index, SFL_PCM_RINGTONE);
}
std::vector<std::string> ConfigurationManager::getCurrentAudioDevicesIndex() {
diff --git a/sflphone-common/src/dbus/configurationmanager.h b/sflphone-common/src/dbus/configurationmanager.h
index c21a2f85b8149c458043d1c215e530ffa975017e..cf711cf907c06e13daf27adc75e2429871b8c5c6 100644
--- a/sflphone-common/src/dbus/configurationmanager.h
+++ b/sflphone-common/src/dbus/configurationmanager.h
@@ -75,14 +75,14 @@ public:
std::vector< std::string > getActiveCodecList (const std::string& accountID);
void setActiveCodecList (const std::vector< std::string >& list, const std::string& accountID);
- std::vector< std::string > getInputAudioPluginList();
- std::vector< std::string > getOutputAudioPluginList();
+ std::vector< std::string > getAudioPluginList();
void setInputAudioPlugin(const std::string& audioPlugin);
void setOutputAudioPlugin(const std::string& audioPlugin);
std::vector< std::string > getAudioOutputDeviceList();
void setAudioOutputDevice(const int32_t& index);
- std::vector< std::string > getAudioInputDeviceList();
void setAudioInputDevice(const int32_t& index);
+ void setAudioRingtoneDevice(const int32_t& index);
+ std::vector< std::string > getAudioInputDeviceList();
std::vector< std::string > getCurrentAudioDevicesIndex();
int32_t getAudioDeviceIndex(const std::string& name);
std::string getCurrentAudioOutputPlugin( void );
diff --git a/sflphone-common/src/eventthread.cpp b/sflphone-common/src/eventthread.cpp
index 93bca48010e8bbc022e74b81f4fca8acc3423fe5..cc16b5e77f45e448e13455b5ed99853c9b17fcda 100644
--- a/sflphone-common/src/eventthread.cpp
+++ b/sflphone-common/src/eventthread.cpp
@@ -65,7 +65,7 @@ void AudioThread::run (void)
{
while (!testCancel()) {
_alsa->audioCallback();
- Thread::sleep (3);
+ Thread::sleep (20);
}
}
diff --git a/sflphone-common/src/global.h b/sflphone-common/src/global.h
index 3ed01299249a17df11fc469a886af17e12780893..552616d7851be88197c105e84c062d3c0b2710fc 100644
--- a/sflphone-common/src/global.h
+++ b/sflphone-common/src/global.h
@@ -97,6 +97,8 @@ static const SOUND_FORMAT INT32 = 0x8;
#define PCM_PLUGHW "plughw" /** Alsa plugin */
#define PCM_DEFAULT "default" /** Default ALSA plugin */
#define PCM_DMIX "plug:dmix" /** Alsa plugin for software mixing */
+#define PCM_DSNOOP "plug:dsnoop" /** Alsa plugin for microphone sharing */
+#define PCM_DMIX_DSNOOP "dmix/dsnoop" /** Audio profile using Alsa dmix/dsnoop */
#define SFL_CODEC_VALID_PREFIX "libcodec_" /** Valid prefix for codecs shared library */
#define SFL_CODEC_VALID_EXTEN ".so" /** Valid extension for codecs shared library */
@@ -106,6 +108,7 @@ static const SOUND_FORMAT INT32 = 0x8;
#define SFL_PCM_BOTH 0x0021 /** To open both playback and capture devices */
#define SFL_PCM_PLAYBACK 0x0022 /** To open playback device only */
#define SFL_PCM_CAPTURE 0x0023 /** To open capture device only */
+#define SFL_PCM_RINGTONE 0x0024
#ifdef USE_IAX
#define IAX2_ENABLED true /** IAX2 support */
diff --git a/sflphone-common/src/managerimpl.cpp b/sflphone-common/src/managerimpl.cpp
index c6057e93ec5f0d4c972aa9c5dec0b4134cf91348..34a8e7494c33a87aabb09e3529803fe687827e84 100644
--- a/sflphone-common/src/managerimpl.cpp
+++ b/sflphone-common/src/managerimpl.cpp
@@ -2112,6 +2112,8 @@ void ManagerImpl::initConfigFile (bool load_user_value, std::string alternate) {
ALSA_CARD_ID_IN, ALSA_DFT_CARD), AUDIO);
_config.addDefaultValue(std::pair<std::string, std::string>(
ALSA_CARD_ID_OUT, ALSA_DFT_CARD), AUDIO);
+ _config.addDefaultValue(std::pair<std::string, std::string>(
+ ALSA_CARD_ID_RING, ALSA_DFT_CARD), AUDIO);
_config.addDefaultValue(std::pair<std::string, std::string>(
AUDIO_SAMPLE_RATE, DFT_SAMPLE_RATE), AUDIO);
_config.addDefaultValue(std::pair<std::string, std::string>(
@@ -2327,8 +2329,8 @@ void ManagerImpl::setInputAudioPlugin (const std::string& audioPlugin) {
if (CHECK_INTERFACE (layer , ALSA)) {
_debug ("Set input audio plugin");
_audiodriver -> setErrorMessage(-1);
- _audiodriver -> openDevice(_audiodriver -> getIndexIn(),
- _audiodriver -> getIndexOut(), _audiodriver -> getSampleRate(),
+ _audiodriver -> openDevice(_audiodriver->getIndexIn(), _audiodriver->getIndexOut(),
+ _audiodriver->getIndexRing(), _audiodriver -> getSampleRate(),
_audiodriver -> getFrameSize(), SFL_PCM_CAPTURE, audioPlugin);
if (_audiodriver -> getErrorMessage() != -1)
@@ -2347,9 +2349,9 @@ void ManagerImpl::setOutputAudioPlugin (const std::string& audioPlugin) {
_debug ("Manager: Set output audio plugin");
_audiodriver -> setErrorMessage(-1);
- res = _audiodriver -> openDevice(_audiodriver -> getIndexIn(),
- _audiodriver -> getIndexOut(), _audiodriver -> getSampleRate(),
- _audiodriver -> getFrameSize(), SFL_PCM_BOTH, audioPlugin);
+ res = _audiodriver -> openDevice(_audiodriver->getIndexIn(), _audiodriver->getIndexOut(),
+ _audiodriver->getIndexRing(), _audiodriver -> getSampleRate(),
+ _audiodriver -> getFrameSize(), SFL_PCM_BOTH, audioPlugin);
if (_audiodriver -> getErrorMessage() != -1)
notifyErrClient(_audiodriver -> getErrorMessage());
@@ -2378,27 +2380,53 @@ std::vector<std::string> ManagerImpl::getAudioOutputDeviceList (void) {
/**
* Set audio output device
*/
-void ManagerImpl::setAudioOutputDevice (const int index) {
- AlsaLayer *alsalayer;
- std::string alsaplugin;
- _debug ("Set audio output device: %i", index);
-
- _audiodriver -> setErrorMessage(-1);
-
- alsalayer = dynamic_cast<AlsaLayer*> (getAudioDriver());
- alsaplugin = alsalayer->getAudioPlugin();
+void ManagerImpl::setAudioDevice (const int index, int streamType) {
+
+ AlsaLayer *alsalayer = NULL;
+ std::string alsaplugin;
+ _debug ("Manager: Set audio device: %i", index);
- _debug (" set output plugin: %s", alsaplugin.c_str());
+ _audiodriver -> setErrorMessage(-1);
- _audiodriver->openDevice(_audiodriver->getIndexIn(), index,
- _audiodriver->getSampleRate(), _audiodriver->getFrameSize(),
- SFL_PCM_PLAYBACK, alsaplugin);
+ if(!(alsalayer = dynamic_cast<AlsaLayer*> (getAudioDriver()))) {
+ _warn("Manager: Error: No audio driver");
+ return;
+ }
+
+ alsaplugin = alsalayer->getAudioPlugin();
+
+ _debug ("Manager: Set ALSA plugin: %s", alsaplugin.c_str());
+
+
+ switch(streamType) {
+ case SFL_PCM_PLAYBACK:
+ _debug("Manager: Set output device");
+ _audiodriver->openDevice(_audiodriver->getIndexIn(), index, _audiodriver->getIndexRing(),
+ _audiodriver->getSampleRate(), _audiodriver->getFrameSize(),
+ SFL_PCM_PLAYBACK, alsaplugin);
+ setConfig(AUDIO, ALSA_CARD_ID_OUT, index);
+ break;
+ case SFL_PCM_CAPTURE:
+ _debug("Manager: Set input device");
+ _audiodriver->openDevice(index, _audiodriver->getIndexOut(), _audiodriver->getIndexRing(),
+ _audiodriver->getSampleRate(), _audiodriver->getFrameSize(),
+ SFL_PCM_CAPTURE, alsaplugin);
+ setConfig(AUDIO, ALSA_CARD_ID_IN, index);
+ break;
+ case SFL_PCM_RINGTONE:
+ _debug("Manager: Set ringtone device");
+ _audiodriver->openDevice(_audiodriver->getIndexOut(), _audiodriver->getIndexOut(), index,
+ _audiodriver->getSampleRate(), _audiodriver->getFrameSize(),
+ SFL_PCM_RINGTONE, alsaplugin);
+ setConfig(AUDIO, ALSA_CARD_ID_RING, index);
+ break;
+ default:
+ _warn("Unknown stream type");
+ }
- if (_audiodriver -> getErrorMessage() != -1)
- notifyErrClient(_audiodriver -> getErrorMessage());
+ if (_audiodriver -> getErrorMessage() != -1)
+ notifyErrClient(_audiodriver -> getErrorMessage());
- // set config
- setConfig(AUDIO, ALSA_CARD_ID_OUT, index);
}
/**
@@ -2416,44 +2444,19 @@ std::vector<std::string> ManagerImpl::getAudioInputDeviceList (void) {
return devices;
}
-/**
- * Set audio input device
- */
-void ManagerImpl::setAudioInputDevice (const int index) {
- AlsaLayer *alsalayer;
- std::string alsaplugin;
-
- _debug ("Set audio input device %i", index);
-
- _audiodriver -> setErrorMessage(-1);
-
- alsalayer = dynamic_cast<AlsaLayer*> (getAudioDriver());
- alsaplugin = alsalayer->getAudioPlugin();
-
- _debug (" set input plugin: %s", alsaplugin.c_str());
-
- _audiodriver->openDevice(index, _audiodriver->getIndexOut(),
- _audiodriver->getSampleRate(), _audiodriver->getFrameSize(),
- SFL_PCM_CAPTURE, alsaplugin);
-
- if (_audiodriver -> getErrorMessage() != -1)
- notifyErrClient(_audiodriver -> getErrorMessage());
-
- // set config
- setConfig(AUDIO, ALSA_CARD_ID_IN, index);
-}
-
/**
* Get string array representing integer indexes of output and input device
*/
std::vector<std::string> ManagerImpl::getCurrentAudioDevicesIndex () {
_debug ("Get current audio devices index");
std::vector<std::string> v;
- std::stringstream ssi, sso;
+ std::stringstream ssi, sso, ssr;
sso << _audiodriver->getIndexOut();
v.push_back(sso.str());
ssi << _audiodriver->getIndexIn();
v.push_back(ssi.str());
+ ssr << _audiodriver->getIndexRing();
+ v.push_back(ssr.str());
return v;
}
@@ -2761,7 +2764,7 @@ bool ManagerImpl::initAudioDriver (void) {
* Initialization: Main Thread and gui
*/
void ManagerImpl::selectAudioDriver (void) {
- int layer, numCardIn, numCardOut, sampleRate, frameSize;
+ int layer, numCardIn, numCardOut, numCardRing, sampleRate, frameSize;
std::string alsaPlugin;
AlsaLayer *alsalayer;
@@ -2772,6 +2775,8 @@ void ManagerImpl::selectAudioDriver (void) {
alsaPlugin = getConfigString(AUDIO, ALSA_PLUGIN);
numCardIn = getConfigInt(AUDIO, ALSA_CARD_ID_IN);
numCardOut = getConfigInt(AUDIO, ALSA_CARD_ID_OUT);
+ numCardRing = getConfigInt(AUDIO, ALSA_CARD_ID_RING);
+
sampleRate = getConfigInt(AUDIO, AUDIO_SAMPLE_RATE);
if (sampleRate <= 0 || sampleRate > 48000) {
@@ -2783,25 +2788,31 @@ void ManagerImpl::selectAudioDriver (void) {
/* Only for the ALSA layer, we check the sound card information */
if (layer == ALSA) {
- alsalayer = dynamic_cast<AlsaLayer*> (getAudioDriver());
+ alsalayer = dynamic_cast<AlsaLayer*> (getAudioDriver());
+
+ if (!alsalayer -> soundCardIndexExist(numCardIn, SFL_PCM_CAPTURE)) {
+ _debug (" Card with index %i doesn't exist or cannot capture. Switch to 0.", numCardIn);
+ numCardIn = ALSA_DFT_CARD_ID;
+ setConfig(AUDIO, ALSA_CARD_ID_IN, ALSA_DFT_CARD_ID);
+ }
- if (!alsalayer -> soundCardIndexExist(numCardIn, SFL_PCM_CAPTURE)) {
- _debug (" Card with index %i doesn't exist or cannot capture. Switch to 0.", numCardIn);
- numCardIn = ALSA_DFT_CARD_ID;
- setConfig(AUDIO, ALSA_CARD_ID_IN, ALSA_DFT_CARD_ID);
- }
+ if (!alsalayer -> soundCardIndexExist(numCardOut, SFL_PCM_PLAYBACK)) {
+ _debug (" Card with index %i doesn't exist or cannot playback. Switch to 0.", numCardOut);
+ numCardOut = ALSA_DFT_CARD_ID;
+ setConfig(AUDIO, ALSA_CARD_ID_OUT, ALSA_DFT_CARD_ID);
+ }
- if (!alsalayer -> soundCardIndexExist(numCardOut, SFL_PCM_PLAYBACK)) {
- _debug (" Card with index %i doesn't exist or cannot playback . Switch to 0.", numCardOut);
- numCardOut = ALSA_DFT_CARD_ID;
- setConfig(AUDIO, ALSA_CARD_ID_OUT, ALSA_DFT_CARD_ID);
- }
+ if (!alsalayer->soundCardIndexExist(numCardRing, SFL_PCM_RINGTONE)) {
+ _debug(" Card with index %i doesn't exist or cannot ringtone. Switch to 0.", numCardRing);
+ numCardRing = ALSA_DFT_CARD_ID;
+ setConfig(AUDIO, ALSA_CARD_ID_RING, ALSA_DFT_CARD_ID);
+ }
}
_audiodriver->setErrorMessage(-1);
/* Open the audio devices */
- _audiodriver->openDevice(numCardIn, numCardOut, sampleRate, frameSize,
+ _audiodriver->openDevice(numCardIn, numCardOut, numCardRing, sampleRate, frameSize,
SFL_PCM_BOTH, alsaPlugin);
/* Notify the error if there is one */
@@ -2811,10 +2822,10 @@ void ManagerImpl::selectAudioDriver (void) {
}
void ManagerImpl::switchAudioManager (void) {
- int type, samplerate, framesize, numCardIn, numCardOut;
+ int type, samplerate, framesize, numCardIn, numCardOut, numCardRing;
std::string alsaPlugin;
- _debug ("Switching audio manager ");
+ _debug ("Manager: Switching audio manager ");
if (!_audiodriver)
return;
@@ -2822,18 +2833,17 @@ void ManagerImpl::switchAudioManager (void) {
type = _audiodriver->getLayerType();
samplerate = getConfigInt(AUDIO, AUDIO_SAMPLE_RATE);
-
framesize = getConfigInt(AUDIO, ALSA_FRAME_SIZE);
- _debug ("samplerate: %i, framesize %i\n", samplerate, framesize);
+ _debug ("Mnager: samplerate: %i, framesize %i\n", samplerate, framesize);
alsaPlugin = getConfigString(AUDIO, ALSA_PLUGIN);
numCardIn = getConfigInt(AUDIO, ALSA_CARD_ID_IN);
-
numCardOut = getConfigInt(AUDIO, ALSA_CARD_ID_OUT);
+ numCardRing = getConfigInt(AUDIO, ALSA_CARD_ID_RING);
- _debug ("Deleting current layer... ");
+ _debug ("Manager: Deleting current layer... ");
// ost::MutexLock lock (*getAudioLayerMutex());
getAudioLayerMutex()->enter();
@@ -2846,36 +2856,35 @@ void ManagerImpl::switchAudioManager (void) {
switch (type) {
case ALSA:
- _debug ("Creating Pulseaudio layer...");
+ _debug ("Manager: Creating Pulseaudio layer...");
_audiodriver = new PulseLayer(this);
_audiodriver->setMainBuffer(&_mainBuffer);
break;
case PULSEAUDIO:
- _debug ("Creating ALSA layer...");
+ _debug ("Manager: Creating ALSA layer...");
_audiodriver = new AlsaLayer(this);
_audiodriver->setMainBuffer(&_mainBuffer);
break;
default:
- _debug ("Error: audio layer unknown");
+ _warn("Manager: Error: audio layer unknown");
break;
}
_audiodriver->setErrorMessage(-1);
- _audiodriver->openDevice(numCardIn, numCardOut, samplerate, framesize,
+ _audiodriver->openDevice(numCardIn, numCardOut, numCardRing, samplerate, framesize,
SFL_PCM_BOTH, alsaPlugin);
if (_audiodriver -> getErrorMessage() != -1)
- notifyErrClient(_audiodriver -> getErrorMessage());
-
- _debug ("Current device: %i ", type);
+ notifyErrClient(_audiodriver -> getErrorMessage());
- _debug ("has current call: %i ", hasCurrentCall());
+ _debug ("Manager: Current device: %i ", type);
+ _debug ("Manager: Has current call: %i ", hasCurrentCall());
if (hasCurrentCall())
- _audiodriver->startStream();
+ _audiodriver->startStream();
// ost::MutexLock unlock (*getAudioLayerMutex());
getAudioLayerMutex()->leave();
diff --git a/sflphone-common/src/managerimpl.h b/sflphone-common/src/managerimpl.h
index 35e236c4fd4a18648bc3bd7f954257f120a9ce76..09acea4fd8a40c63d2c2c7f9a472a04b5c67455b 100644
--- a/sflphone-common/src/managerimpl.h
+++ b/sflphone-common/src/managerimpl.h
@@ -538,10 +538,11 @@ class ManagerImpl {
std::vector<std::string> getAudioOutputDeviceList(void);
/**
- * Set audio output device
+ * Set audio device
* @param index The index of the soundcard
+ * @param the type of stream, either SFL_PCM_PLAYBACK, SFL_PCM_CAPTURE, SFL_PCM_RINGTONE
*/
- void setAudioOutputDevice(const int index);
+ void setAudioDevice(const int index, const int streamType);
/**
* Get list of supported audio input device
@@ -549,12 +550,6 @@ class ManagerImpl {
*/
std::vector<std::string> getAudioInputDeviceList(void);
- /**
- * Set audio input device
- * @param index The index of the soundcard
- */
- void setAudioInputDevice(const int index);
-
/**
* Get string array representing integer indexes of output and input device
* @return std::vector<std::string> A list of the current audio devices
diff --git a/sflphone-common/src/sip/sipvoiplink.cpp b/sflphone-common/src/sip/sipvoiplink.cpp
index 11d91ca9469118c6c6a476a11b68122291f8e7c4..a005ab44408ff388ebe41076eb77fd0ee2de9444 100644
--- a/sflphone-common/src/sip/sipvoiplink.cpp
+++ b/sflphone-common/src/sip/sipvoiplink.cpp
@@ -280,6 +280,7 @@ SIPVoIPLink::~SIPVoIPLink()
_debug("UserAgent: SIPVoIPLink destructor called");
terminate();
+
}
SIPVoIPLink* SIPVoIPLink::instance (const AccountID& id)
@@ -344,7 +345,6 @@ SIPVoIPLink::terminate()
initDone (false);
- _debug("Terminating");
}
void
@@ -1259,6 +1259,8 @@ SIPVoIPLink::refuse (const CallID& id)
call->getInvSession()->mod_data[getModId() ] = NULL;
+ removeCall(id);
+
terminateOneCall (id);
_debug("UserAgent: Refuse call completed");
diff --git a/sflphone-common/src/user_cfg.h b/sflphone-common/src/user_cfg.h
index 6d45d53c6661785516c2d64ce7e02bc1cdff6239..910e3f9e15691e625918a4eab7920034ca078ca7 100644
--- a/sflphone-common/src/user_cfg.h
+++ b/sflphone-common/src/user_cfg.h
@@ -46,6 +46,7 @@
#define CODECS "ActiveCodecs" /** List of active codecs */
#define ALSA_CARD_ID_IN "Alsa.cardID_In" /** Soundcard index to use for capture */
#define ALSA_CARD_ID_OUT "Alsa.cardID_Out" /** Soundcard index to use for playback */
+#define ALSA_CARD_ID_RING "Alsa.cardID_Ring" /** Soundcard index to use for ringtone */
#define ALSA_FRAME_SIZE "Alsa.framesize" /** Audio layer frame size */
#define ALSA_PLUGIN "Alsa.plugin" /** Alsa plugin */
#define AUDIO_SAMPLE_RATE "Alsa.sampleRate" /** Audio layer sample rate */