diff --git a/sflphone-gtk/src/configurationmanager-glue.h b/sflphone-gtk/src/configurationmanager-glue.h
index b2e3c447dabc912dbfcf8b22dcebeca23cb3626a..9e395b9f8782ba75ba14912c7f5d41a17655144b 100644
--- a/sflphone-gtk/src/configurationmanager-glue.h
+++ b/sflphone-gtk/src/configurationmanager-glue.h
@@ -617,22 +617,22 @@ static
inline
#endif
gboolean
-org_sflphone_SFLphone_ConfigurationManager_get_audio_manager_list (DBusGProxy *proxy, char *** OUT_list, GError **error)
+org_sflphone_SFLphone_ConfigurationManager_get_input_audio_plugin_list (DBusGProxy *proxy, char *** OUT_list, GError **error)
{
- return dbus_g_proxy_call (proxy, "getAudioManagerList", error, G_TYPE_INVALID, G_TYPE_STRV, OUT_list, G_TYPE_INVALID);
+ return dbus_g_proxy_call (proxy, "getInputAudioPluginList", error, G_TYPE_INVALID, G_TYPE_STRV, OUT_list, G_TYPE_INVALID);
}
-typedef void (*org_sflphone_SFLphone_ConfigurationManager_get_audio_manager_list_reply) (DBusGProxy *proxy, char * *OUT_list, GError *error, gpointer userdata);
+typedef void (*org_sflphone_SFLphone_ConfigurationManager_get_input_audio_plugin_list_reply) (DBusGProxy *proxy, char * *OUT_list, GError *error, gpointer userdata);
static void
-org_sflphone_SFLphone_ConfigurationManager_get_audio_manager_list_async_callback (DBusGProxy *proxy, DBusGProxyCall *call, void *user_data)
+org_sflphone_SFLphone_ConfigurationManager_get_input_audio_plugin_list_async_callback (DBusGProxy *proxy, DBusGProxyCall *call, void *user_data)
{
DBusGAsyncData *data = (DBusGAsyncData*) user_data;
GError *error = NULL;
char ** OUT_list;
dbus_g_proxy_end_call (proxy, call, &error, G_TYPE_STRV, &OUT_list, G_TYPE_INVALID);
- (*(org_sflphone_SFLphone_ConfigurationManager_get_audio_manager_list_reply)data->cb) (proxy, OUT_list, error, data->userdata);
+ (*(org_sflphone_SFLphone_ConfigurationManager_get_input_audio_plugin_list_reply)data->cb) (proxy, OUT_list, error, data->userdata);
return;
}
@@ -641,35 +641,110 @@ static
inline
#endif
DBusGProxyCall*
-org_sflphone_SFLphone_ConfigurationManager_get_audio_manager_list_async (DBusGProxy *proxy, org_sflphone_SFLphone_ConfigurationManager_get_audio_manager_list_reply callback, gpointer userdata)
+org_sflphone_SFLphone_ConfigurationManager_get_input_audio_plugin_list_async (DBusGProxy *proxy, org_sflphone_SFLphone_ConfigurationManager_get_input_audio_plugin_list_reply callback, gpointer userdata)
{
DBusGAsyncData *stuff;
stuff = g_new (DBusGAsyncData, 1);
stuff->cb = G_CALLBACK (callback);
stuff->userdata = userdata;
- return dbus_g_proxy_begin_call (proxy, "getAudioManagerList", org_sflphone_SFLphone_ConfigurationManager_get_audio_manager_list_async_callback, stuff, g_free, G_TYPE_INVALID);
+ return dbus_g_proxy_begin_call (proxy, "getInputAudioPluginList", org_sflphone_SFLphone_ConfigurationManager_get_input_audio_plugin_list_async_callback, stuff, g_free, G_TYPE_INVALID);
}
static
#ifdef G_HAVE_INLINE
inline
#endif
gboolean
-org_sflphone_SFLphone_ConfigurationManager_set_audio_manager (DBusGProxy *proxy, const char * IN_audioManager, GError **error)
+org_sflphone_SFLphone_ConfigurationManager_get_output_audio_plugin_list (DBusGProxy *proxy, char *** OUT_list, GError **error)
{
- return dbus_g_proxy_call (proxy, "setAudioManager", error, G_TYPE_STRING, IN_audioManager, G_TYPE_INVALID, G_TYPE_INVALID);
+ return dbus_g_proxy_call (proxy, "getOutputAudioPluginList", error, G_TYPE_INVALID, G_TYPE_STRV, OUT_list, G_TYPE_INVALID);
}
-typedef void (*org_sflphone_SFLphone_ConfigurationManager_set_audio_manager_reply) (DBusGProxy *proxy, GError *error, gpointer userdata);
+typedef void (*org_sflphone_SFLphone_ConfigurationManager_get_output_audio_plugin_list_reply) (DBusGProxy *proxy, char * *OUT_list, GError *error, gpointer userdata);
static void
-org_sflphone_SFLphone_ConfigurationManager_set_audio_manager_async_callback (DBusGProxy *proxy, DBusGProxyCall *call, void *user_data)
+org_sflphone_SFLphone_ConfigurationManager_get_output_audio_plugin_list_async_callback (DBusGProxy *proxy, DBusGProxyCall *call, void *user_data)
+{
+ DBusGAsyncData *data = (DBusGAsyncData*) user_data;
+ GError *error = NULL;
+ char ** OUT_list;
+ dbus_g_proxy_end_call (proxy, call, &error, G_TYPE_STRV, &OUT_list, G_TYPE_INVALID);
+ (*(org_sflphone_SFLphone_ConfigurationManager_get_output_audio_plugin_list_reply)data->cb) (proxy, OUT_list, error, data->userdata);
+ return;
+}
+
+static
+#ifdef G_HAVE_INLINE
+inline
+#endif
+DBusGProxyCall*
+org_sflphone_SFLphone_ConfigurationManager_get_output_audio_plugin_list_async (DBusGProxy *proxy, org_sflphone_SFLphone_ConfigurationManager_get_output_audio_plugin_list_reply callback, gpointer userdata)
+
+{
+ DBusGAsyncData *stuff;
+ stuff = g_new (DBusGAsyncData, 1);
+ stuff->cb = G_CALLBACK (callback);
+ stuff->userdata = userdata;
+ return dbus_g_proxy_begin_call (proxy, "getOutputAudioPluginList", org_sflphone_SFLphone_ConfigurationManager_get_output_audio_plugin_list_async_callback, stuff, g_free, G_TYPE_INVALID);
+}
+static
+#ifdef G_HAVE_INLINE
+inline
+#endif
+gboolean
+org_sflphone_SFLphone_ConfigurationManager_set_input_audio_plugin (DBusGProxy *proxy, const char * IN_audioPlugin, GError **error)
+
+{
+ return dbus_g_proxy_call (proxy, "setInputAudioPlugin", error, G_TYPE_STRING, IN_audioPlugin, G_TYPE_INVALID, G_TYPE_INVALID);
+}
+
+typedef void (*org_sflphone_SFLphone_ConfigurationManager_set_input_audio_plugin_reply) (DBusGProxy *proxy, GError *error, gpointer userdata);
+
+static void
+org_sflphone_SFLphone_ConfigurationManager_set_input_audio_plugin_async_callback (DBusGProxy *proxy, DBusGProxyCall *call, void *user_data)
+{
+ DBusGAsyncData *data = (DBusGAsyncData*) user_data;
+ GError *error = NULL;
+ dbus_g_proxy_end_call (proxy, call, &error, G_TYPE_INVALID);
+ (*(org_sflphone_SFLphone_ConfigurationManager_set_input_audio_plugin_reply)data->cb) (proxy, error, data->userdata);
+ return;
+}
+
+static
+#ifdef G_HAVE_INLINE
+inline
+#endif
+DBusGProxyCall*
+org_sflphone_SFLphone_ConfigurationManager_set_input_audio_plugin_async (DBusGProxy *proxy, const char * IN_audioPlugin, org_sflphone_SFLphone_ConfigurationManager_set_input_audio_plugin_reply callback, gpointer userdata)
+
+{
+ DBusGAsyncData *stuff;
+ stuff = g_new (DBusGAsyncData, 1);
+ stuff->cb = G_CALLBACK (callback);
+ stuff->userdata = userdata;
+ return dbus_g_proxy_begin_call (proxy, "setInputAudioPlugin", org_sflphone_SFLphone_ConfigurationManager_set_input_audio_plugin_async_callback, stuff, g_free, G_TYPE_STRING, IN_audioPlugin, G_TYPE_INVALID);
+}
+static
+#ifdef G_HAVE_INLINE
+inline
+#endif
+gboolean
+org_sflphone_SFLphone_ConfigurationManager_set_output_audio_plugin (DBusGProxy *proxy, const char * IN_audioPlugin, GError **error)
+
+{
+ return dbus_g_proxy_call (proxy, "setOutputAudioPlugin", error, G_TYPE_STRING, IN_audioPlugin, G_TYPE_INVALID, G_TYPE_INVALID);
+}
+
+typedef void (*org_sflphone_SFLphone_ConfigurationManager_set_output_audio_plugin_reply) (DBusGProxy *proxy, GError *error, gpointer userdata);
+
+static void
+org_sflphone_SFLphone_ConfigurationManager_set_output_audio_plugin_async_callback (DBusGProxy *proxy, DBusGProxyCall *call, void *user_data)
{
DBusGAsyncData *data = (DBusGAsyncData*) user_data;
GError *error = NULL;
dbus_g_proxy_end_call (proxy, call, &error, G_TYPE_INVALID);
- (*(org_sflphone_SFLphone_ConfigurationManager_set_audio_manager_reply)data->cb) (proxy, error, data->userdata);
+ (*(org_sflphone_SFLphone_ConfigurationManager_set_output_audio_plugin_reply)data->cb) (proxy, error, data->userdata);
return;
}
@@ -678,14 +753,14 @@ static
inline
#endif
DBusGProxyCall*
-org_sflphone_SFLphone_ConfigurationManager_set_audio_manager_async (DBusGProxy *proxy, const char * IN_audioManager, org_sflphone_SFLphone_ConfigurationManager_set_audio_manager_reply callback, gpointer userdata)
+org_sflphone_SFLphone_ConfigurationManager_set_output_audio_plugin_async (DBusGProxy *proxy, const char * IN_audioPlugin, org_sflphone_SFLphone_ConfigurationManager_set_output_audio_plugin_reply callback, gpointer userdata)
{
DBusGAsyncData *stuff;
stuff = g_new (DBusGAsyncData, 1);
stuff->cb = G_CALLBACK (callback);
stuff->userdata = userdata;
- return dbus_g_proxy_begin_call (proxy, "setAudioManager", org_sflphone_SFLphone_ConfigurationManager_set_audio_manager_async_callback, stuff, g_free, G_TYPE_STRING, IN_audioManager, G_TYPE_INVALID);
+ return dbus_g_proxy_begin_call (proxy, "setOutputAudioPlugin", org_sflphone_SFLphone_ConfigurationManager_set_output_audio_plugin_async_callback, stuff, g_free, G_TYPE_STRING, IN_audioPlugin, G_TYPE_INVALID);
}
static
#ifdef G_HAVE_INLINE
diff --git a/sflphone-gtk/src/configwindow.c b/sflphone-gtk/src/configwindow.c
index ebcb0d8dd97ff0bfd8e3971b0777f5f247d3bf82..98ccdfbc1f2f71239bcfc78599ddc2b60dd29758 100644
--- a/sflphone-gtk/src/configwindow.c
+++ b/sflphone-gtk/src/configwindow.c
@@ -39,7 +39,8 @@ gboolean dialogOpen = FALSE;
GtkListStore *accountStore;
GtkWidget *codecTreeView; // View used instead of store to get access to selection
// instead of keeping selected codec as a variable
-GtkListStore *audioManagerStore;
+GtkListStore *inputAudioPluginStore;
+GtkListStore *outputAudioPluginStore;
GtkListStore *outputAudioDeviceManagerStore;
GtkListStore *inputAudioDeviceManagerStore;
@@ -137,30 +138,54 @@ config_window_fill_codec_list()
}
/**
- * Fill store with audio managers
+ * Fill store with input audio plugins
*/
void
-config_window_fill_audio_manager_list()
+config_window_fill_input_audio_plugin_list()
{
GtkTreeIter iter;
gchar** list;
gchar* managerName;
- gtk_list_store_clear(audioManagerStore);
+ gtk_list_store_clear(inputAudioPluginStore);
// Call dbus to retreive list
- list = dbus_get_audio_manager_list();
+ list = dbus_get_input_audio_plugin_list();
// For each API name included in list
int c = 0;
for(managerName = list[c]; managerName != NULL; managerName = list[c])
{
c++;
- gtk_list_store_append(audioManagerStore, &iter);
- gtk_list_store_set(audioManagerStore, &iter, 0 , managerName, -1);
+ gtk_list_store_append(inputAudioPluginStore, &iter);
+ gtk_list_store_set(inputAudioPluginStore, &iter, 0 , managerName, -1);
}
}
+/**
+ * Fill store with output audio plugins
+ */
+void
+config_window_fill_output_audio_plugin_list()
+{
+ GtkTreeIter iter;
+ gchar** list;
+ gchar* managerName;
+
+ gtk_list_store_clear(outputAudioPluginStore);
+
+ // Call dbus to retreive list
+ list = dbus_get_output_audio_plugin_list();
+
+ // For each API name included in list
+ int c = 0;
+ for(managerName = list[c]; managerName != NULL; managerName = list[c])
+ {
+ c++;
+ gtk_list_store_append(outputAudioPluginStore, &iter);
+ gtk_list_store_set(outputAudioPluginStore, &iter, 0 , managerName, -1);
+ }
+}
/**
* Fill output audio device store
*/
@@ -281,15 +306,37 @@ select_active_input_audio_device()
}
/**
- * Select the audio manager by calling the server
- * Not yet used because audio manager is by default ALSA
+ * Select the input audio plugin by calling the server
*/
static void
-select_audio_manager(GtkWidget* widget, gpointer data)
+select_input_audio_plugin(GtkWidget* widget, gpointer data)
{
//dbus_set_audio_manager("");
}
+/**
+ * Select the output audio plugin by calling the server
+ */
+static void
+select_output_audio_plugin(GtkComboBox* widget, gpointer data)
+{
+ GtkTreeModel* model;
+ GtkTreeIter iter;
+ int comboBoxIndex;
+ gchar* pluginName;
+
+ comboBoxIndex = gtk_combo_box_get_active(widget);
+
+ if(comboBoxIndex >= 0)
+ {
+ model = gtk_combo_box_get_model(widget);
+ gtk_combo_box_get_active_iter(widget, &iter);
+ gtk_tree_model_get(model, &iter, 0, &pluginName, -1);
+
+ dbus_set_output_audio_plugin(pluginName);
+ }
+ //dbus_set_output_audio_manager("");
+}
/**
* Set the audio output device on the server with its index
*/
@@ -873,6 +920,7 @@ create_audio_tab ()
gtk_misc_set_alignment(GTK_MISC(deviceLabel), 0, 0.5);
gtk_label_set_justify(GTK_LABEL(deviceLabel), GTK_JUSTIFY_LEFT);
gtk_box_pack_start(GTK_BOX(ret), deviceLabel, FALSE, FALSE, 0);
+
// Main device widget
deviceBox = gtk_hbox_new(FALSE, 10);
@@ -887,17 +935,38 @@ create_audio_tab ()
// Device : Audio manager
// Create title label
- titleLabel = gtk_label_new("Audio manager:");
- gtk_misc_set_alignment(GTK_MISC(titleLabel), 0, 0.5);
+ /*
+ titleLabel = gtk_label_new("Alsa plug-IN:");
+ gtk_misc_set_alignment(GTK_MISC(titleLabel), 0, 0.5);
+ gtk_table_attach(GTK_TABLE(deviceTable), titleLabel, 0, 1, 0, 1, GTK_FILL | GTK_EXPAND, GTK_SHRINK, 0, 0);
+ gtk_widget_show(titleLabel);
+ // Set choices of audio managers
+ inputAudioPluginStore = gtk_list_store_new(1, G_TYPE_STRING);
+ config_window_fill_input_audio_plugin_list();
+ comboBox = gtk_combo_box_new_with_model(GTK_TREE_MODEL(inputAudioPluginStore));
+ gtk_combo_box_set_active(GTK_COMBO_BOX(comboBox), 0);
+ gtk_label_set_mnemonic_widget(GTK_LABEL(titleLabel), comboBox);
+ g_signal_connect(G_OBJECT(comboBox), "changed", G_CALLBACK(select_input_audio_plugin), comboBox);
+
+ // Set rendering
+ renderer = gtk_cell_renderer_text_new();
+ gtk_cell_layout_pack_start(GTK_CELL_LAYOUT(comboBox), renderer, TRUE);
+ gtk_cell_layout_set_attributes(GTK_CELL_LAYOUT(comboBox), renderer, "text", 0, NULL);
+ gtk_table_attach(GTK_TABLE(deviceTable), comboBox, 1, 2, 0, 1, GTK_FILL | GTK_EXPAND, GTK_SHRINK, 0, 0);
+ gtk_widget_show(comboBox);
+ */
+ // Create title label
+ titleLabel = gtk_label_new("Alsa plug-OUT:");
+ gtk_misc_set_alignment(GTK_MISC(titleLabel), 0, 0.5);
gtk_table_attach(GTK_TABLE(deviceTable), titleLabel, 1, 2, 0, 1, GTK_FILL | GTK_EXPAND, GTK_SHRINK, 0, 0);
gtk_widget_show(titleLabel);
// Set choices of audio managers
- audioManagerStore = gtk_list_store_new(1, G_TYPE_STRING);
- config_window_fill_audio_manager_list();
- comboBox = gtk_combo_box_new_with_model(GTK_TREE_MODEL(audioManagerStore));
+ outputAudioPluginStore = gtk_list_store_new(1, G_TYPE_STRING);
+ config_window_fill_output_audio_plugin_list();
+ comboBox = gtk_combo_box_new_with_model(GTK_TREE_MODEL(outputAudioPluginStore));
gtk_combo_box_set_active(GTK_COMBO_BOX(comboBox), 0);
gtk_label_set_mnemonic_widget(GTK_LABEL(titleLabel), comboBox);
- g_signal_connect(G_OBJECT(comboBox), "changed", G_CALLBACK(select_audio_manager), comboBox);
+ g_signal_connect(G_OBJECT(comboBox), "changed", G_CALLBACK(select_output_audio_plugin), comboBox);
// Set rendering
renderer = gtk_cell_renderer_text_new();
@@ -951,7 +1020,7 @@ create_audio_tab ()
// Create detect button
refreshButton = gtk_button_new_with_label("Detect all");
gtk_button_set_image(GTK_BUTTON(refreshButton), gtk_image_new_from_stock(GTK_STOCK_REFRESH, GTK_ICON_SIZE_BUTTON));
- gtk_table_attach(GTK_TABLE(deviceTable), refreshButton, 3, 4, 0, 3, GTK_EXPAND, GTK_EXPAND, 0, 0);
+ gtk_table_attach(GTK_TABLE(deviceTable), refreshButton, 3, 4, 3, 4, GTK_EXPAND, GTK_EXPAND, 0, 0);
// Set event on selection
g_signal_connect(G_OBJECT(refreshButton), "clicked", G_CALLBACK(detect_all_audio_settings), NULL);
diff --git a/sflphone-gtk/src/configwindow.h b/sflphone-gtk/src/configwindow.h
index f22909a2fcd64b003e4fe2ae454402746e1fc22c..783dc677a337ce514d37ef03e4e8203fc04a6e3a 100644
--- a/sflphone-gtk/src/configwindow.h
+++ b/sflphone-gtk/src/configwindow.h
@@ -28,7 +28,8 @@
*/
void config_window_fill_account_list();
void config_window_fill_codec_list();
-void config_window_fill_audio_manager_list();
+void config_window_fill_input_audio_plugin_list();
+void config_window_fill_output_audio_plugin_list();
void config_window_fill_output_audio_device_list();
void select_active_output_audio_device();
void config_window_fill_input_audio_device_list();
diff --git a/sflphone-gtk/src/dbus.c b/sflphone-gtk/src/dbus.c
index d50823ab61a1d1e0335f4d36aee208456d5d4066..4400c347c40f3efd0663aa4b8f167a921b3e2ed9 100644
--- a/sflphone-gtk/src/dbus.c
+++ b/sflphone-gtk/src/dbus.c
@@ -772,50 +772,95 @@ dbus_set_active_codec_list(const gchar** list)
}
/**
- * Get a list of all supported audio managers
+ * Get a list of input supported audio plugins
*/
gchar**
-dbus_get_audio_manager_list()
+dbus_get_input_audio_plugin_list()
{
- g_print("Before get audio manager list");
+ g_print("Before get input audio plugin list");
gchar** array;
GError* error = NULL;
- org_sflphone_SFLphone_ConfigurationManager_get_audio_manager_list(
+ org_sflphone_SFLphone_ConfigurationManager_get_input_audio_plugin_list(
configurationManagerProxy,
&array,
&error);
g_print("After");
if(error)
{
- g_printerr("Failed to call get_audio_manager_list() on ConfigurationManager: %s\n", error->message);
+ g_printerr("Failed to call get_input_audio_plugin_list() on ConfigurationManager: %s\n", error->message);
g_error_free(error);
}
else
- g_print("DBus called get_audio_manager_list() on ConfigurationManager\n");
+ g_print("DBus called get_input_audio_plugin_list() on ConfigurationManager\n");
return array;
}
/**
- * Sets the audio manager from its name
- * Still not used because ALSA is used by default
+ * Get a list of output supported audio plugins
+ */
+gchar**
+dbus_get_output_audio_plugin_list()
+{
+ g_print("Before get output audio plugin list");
+ gchar** array;
+ GError* error = NULL;
+ org_sflphone_SFLphone_ConfigurationManager_get_output_audio_plugin_list(
+ configurationManagerProxy,
+ &array,
+ &error);
+ g_print("After");
+ if(error)
+ {
+ g_printerr("Failed to call get_output_audio_plugin_list() on ConfigurationManager: %s\n", error->message);
+ g_error_free(error);
+ }
+ else
+ g_print("DBus called get_output_audio_plugin_list() on ConfigurationManager\n");
+ return array;
+}
+
+/**
+ * Sets the input audio plugin from its name
+ */
+void
+dbus_set_input_audio_plugin(gchar* audioPlugin)
+{
+ g_print("Before set input audio plugin");
+ GError* error = NULL;
+ org_sflphone_SFLphone_ConfigurationManager_set_input_audio_plugin(
+ configurationManagerProxy,
+ audioPlugin,
+ &error);
+ g_print("After");
+ if(error)
+ {
+ g_printerr("Failed to call set_input_audio_plugin() on ConfigurationManager: %s\n", error->message);
+ g_error_free(error);
+ }
+ else
+ g_print("DBus called set_input_audio_plugin() on ConfigurationManager\n");
+}
+
+/**
+ * Sets the output audio plugin from its name
*/
void
-dbus_set_audio_manager(gchar* audioManager)
+dbus_set_output_audio_plugin(gchar* audioPlugin)
{
- g_print("Before set audio manager");
+ g_print("Before set output audio plugin");
GError* error = NULL;
- org_sflphone_SFLphone_ConfigurationManager_set_audio_manager(
+ org_sflphone_SFLphone_ConfigurationManager_set_output_audio_plugin(
configurationManagerProxy,
- audioManager,
+ audioPlugin,
&error);
g_print("After");
if(error)
{
- g_printerr("Failed to call set_audio_manager() on ConfigurationManager: %s\n", error->message);
+ g_printerr("Failed to call set_output_audio_plugin() on ConfigurationManager: %s\n", error->message);
g_error_free(error);
}
else
- g_print("DBus called set_audio_manager() on ConfigurationManager\n");
+ g_print("DBus called set_output_audio_plugin() on ConfigurationManager\n");
}
/**
diff --git a/sflphone-gtk/src/dbus.h b/sflphone-gtk/src/dbus.h
index c954087674e42b02ae4415fc9e8dcfa913bb0e1d..2282fe38a60f7e77331cda1e8039f7df4532bf50 100644
--- a/sflphone-gtk/src/dbus.h
+++ b/sflphone-gtk/src/dbus.h
@@ -61,8 +61,10 @@ gchar** dbus_get_active_codec_list( void );
void dbus_set_active_codec_list( const gchar** list );
// Audio devices related methods
-gchar** dbus_get_audio_manager_list();
-void dbus_set_audio_manager(gchar* audioManager);
+gchar** dbus_get_input_audio_plugin_list();
+gchar** dbus_get_output_audio_plugin_list();
+void dbus_set_input_audio_plugin(gchar* audioPlugin);
+void dbus_set_output_audio_plugin(gchar* audioPlugin);
gchar** dbus_get_audio_output_device_list();
void dbus_set_audio_output_device(const int index);
gchar** dbus_get_audio_input_device_list();
diff --git a/src/audio/audiolayer.cpp b/src/audio/audiolayer.cpp
index 91575dd030f049c0b253a557f01105f25ed810c0..937faabdfcad4fa2a97f285e8796cbf8dc54590e 100644
--- a/src/audio/audiolayer.cpp
+++ b/src/audio/audiolayer.cpp
@@ -41,7 +41,9 @@
, _manager(manager)
, _playback_handle( NULL )
, _capture_handle( NULL )
- , device_closed( true )
+ , deviceClosed( true )
+ , _mainBuffer( SIZEBUF )
+ , _urgentBuffer( SIZEBUF )
{
_inChannel = 1; // don't put in stereo
@@ -53,7 +55,7 @@
// Destructor
AudioLayer::~AudioLayer (void)
{
- device_closed = true;
+ deviceClosed = true;
if(_capture_handle){
snd_pcm_drop( _capture_handle );
snd_pcm_close( _capture_handle );
@@ -68,12 +70,12 @@ AudioLayer::~AudioLayer (void)
bool
-AudioLayer::openDevice (int indexIn, int indexOut, int sampleRate, int frameSize, int flag)
+AudioLayer::openDevice (int indexIn, int indexOut, int sampleRate, int frameSize, int stream , std::string plugin)
{
- if(device_closed == false)
+ if(deviceClosed == false)
{
- if( flag == SFL_PCM_CAPTURE)
+ if( stream == SFL_PCM_CAPTURE ){
if(_capture_handle)
{
_debugAlsa(" Close the current capture device\n");
@@ -81,30 +83,35 @@ AudioLayer::openDevice (int indexIn, int indexOut, int sampleRate, int frameSize
snd_pcm_close( _capture_handle );
_capture_handle = 0;
}
- else if( flag == SFL_PCM_PLAYBACK)
- if(_playback_handle){
- _debugAlsa(" Close the current playback device\n");
- snd_pcm_drop( _playback_handle );
- snd_pcm_close( _playback_handle );
- _playback_handle = 0;
- }
+ }
+ else if( stream == SFL_PCM_PLAYBACK){
+ if(_playback_handle){
+ _debugAlsa(" Close the current playback device\n");
+ snd_pcm_drop( _playback_handle );
+ snd_pcm_close( _playback_handle );
+ _playback_handle = 0;
+ }
+ }
}
_indexIn = indexIn;
_indexOut = indexOut;
_sampleRate = sampleRate;
_frameSize = frameSize;
+ _audioPlugin = plugin;
_debug(" Setting audiolayer: device in=%2d, out=%2d\n", _indexIn, _indexOut);
_debug(" : nb channel in=%2d, out=%2d\n", _inChannel, _outChannel);
- _debug(" : sample rate=%5d, format=%s\n", _sampleRate, SFLPortaudioFormatString);
+ _debug(" : sample rate=%5d, format=%s\n", _sampleRate, SFLDataFormatString);
_debug(" : frame per buffer=%d\n", FRAME_PER_BUFFER);
ost::MutexLock guard( _mutex );
- std::string pcmp = buildDeviceTopo(PCM_DMIX, indexOut , 0);
- std::string pcmc = buildDeviceTopo(PCM_SURROUND40, indexIn , 0);
- return open_device( pcmp , pcmc , flag);
+ //std::string plugin = PCM_DEFAULT;
+ //std::string plugout = PCM_DEFAULT;
+ std::string pcmp = buildDeviceTopo( plugin , indexOut , 0);
+ std::string pcmc = buildDeviceTopo(PCM_SURROUND40 , indexIn , 0);
+ return open_device( pcmp , pcmc , stream);
}
int
@@ -114,14 +121,54 @@ AudioLayer::getDeviceCount()
return 1;
}
+void AudioLayer::AlsaCallBack( snd_async_handler_t* pcm_callback )
+{
+ ( ( AudioLayer *)snd_async_handler_get_callback_private( pcm_callback )) -> updateBuffers();
+}
+
+ void
+AudioLayer::fillHWBuffer( void)
+{
+ unsigned char* data;
+ int pcmreturn, l1, l2;
+ short s1, s2;
+ int frames;
+ int periodSize = 1024 * 2;
+
+ data = (unsigned char*)malloc(periodSize);
+ frames = periodSize >> 2;
+ for(l1 = 0; l1 < 100; l1++) {
+ for(l2 = 0; l2 < frames; l2++) {
+ s1 = 0;
+ s2 = 0;
+ data[4*l2] = (unsigned char)s1;
+ data[4*l2+1] = s1 >> 8;
+ data[4*l2+2] = (unsigned char)s2;
+ data[4*l2+3] = s2 >> 8;
+ }
+ /* Write num_frames frames from buffer data to the PCM device pointed to by pcm_handle */
+ /* writei for interleaved */
+ /* writen for non-interleaved */
+ while ((pcmreturn = snd_pcm_writei(_playback_handle, data, frames)) < 0) {
+ snd_pcm_prepare(_playback_handle);
+ fprintf(stderr, "<<<<<<<<<<<<<<< Buffer Underrun >>>>>>>>>>>>>>>\n");
+ }
+ }
+
+}
+
void
AudioLayer::startStream(void)
{
_debug(" Start stream\n");
+ int err;
ost::MutexLock guard( _mutex );
snd_pcm_prepare( _capture_handle );
snd_pcm_start( _capture_handle ) ;
- //snd_pcm_start( _playback_handle ) ;
+
+ snd_pcm_prepare( _playback_handle );
+
+ //if( err = snd_pcm_start( _playback_handle) < 0 ) _debugAlsa(" Cannot start (%s)\n", snd_strerror(err));
}
@@ -131,13 +178,15 @@ AudioLayer::stopStream(void)
ost::MutexLock guard( _mutex );
snd_pcm_drop( _capture_handle );
snd_pcm_prepare( _capture_handle );
+ snd_pcm_drop( _playback_handle );
+ snd_pcm_prepare( _playback_handle );
}
void
AudioLayer::sleep(int msec)
{
- snd_pcm_wait(_playback_handle, msec);
+ //snd_pcm_wait(_playback_handle, msec);
}
bool
@@ -151,33 +200,46 @@ AudioLayer::isStreamActive (void)
int
AudioLayer::playSamples(void* buffer, int toCopy)
{
+ //_debug(" Write %d bytes in the main buffer \n ", toCopy);
ost::MutexLock guard( _mutex );
if ( _playback_handle ){
- //_debug("Play samples\n");
write(buffer, toCopy);
}
+ /* int a = _mainBuffer.AvailForPut();
+ if( a >= toCopy ){
+ return _mainBuffer.Put( buffer , toCopy , _defaultVolume );
+ } else {
+ return _mainBuffer.Put( buffer , a , _defaultVolume ) ;
+ }*/
return 0;
}
-
int
-AudioLayer::playRingTone( void* buffer, int toCopy)
+AudioLayer::putUrgent(void* buffer, int toCopy)
{
- _debug(" %d\n", toCopy);
+ fillHWBuffer();
ost::MutexLock guard( _mutex );
- if( _playback_handle )
- snd_pcm_start( _playback_handle );
- write(buffer, toCopy);
+ if ( _playback_handle )
+ {
+ int a = _urgentBuffer.AvailForPut();
+ if( a >= toCopy ){
+ return _urgentBuffer.Put( buffer , toCopy , _defaultVolume );
+ } else {
+ return _urgentBuffer.Put( buffer , a , _defaultVolume ) ;
+ }
+ }
return 0;
}
- int
-AudioLayer::putUrgent(void* buffer, int toCopy)
+ void
+AudioLayer::stopPlaybackStream( void )
{
ost::MutexLock guard( _mutex );
- if ( _playback_handle )
- write(buffer, toCopy);
- return 0;
+ if( _playback_handle )
+ {
+ snd_pcm_drop( _playback_handle );
+ snd_pcm_prepare( _playback_handle );
+ }
}
int
@@ -224,6 +286,29 @@ AudioLayer::toggleEchoTesting() {
///////////////// ALSA PRIVATE FUNCTIONS ////////////////////////////////////////////////
////////////////////////////////////////////////////////////////////////////////////////////
+
+
+ void
+AudioLayer::updateBuffers( void )
+{
+
+ int toGet;
+ int urgentAvail;
+ int bytes = 882 * sizeof(SFLDataFormat);
+ SFLDataFormat* toWrite[bytes];
+ snd_pcm_uframes_t avail = 0;
+ avail = snd_pcm_avail_update( _playback_handle );
+ if( avail ) {
+ //_debug("%i\n", avail);
+ urgentAvail = _urgentBuffer.AvailForGet();
+ if( urgentAvail > 0 ){
+ toGet = ( urgentAvail < bytes ) ? urgentAvail : bytes;
+ _urgentBuffer.Get( toWrite , toGet , 100 );
+ //write( toWrite , toGet );
+ }
+ }
+}
+
bool
AudioLayer::isPlaybackActive(void) {
ost::MutexLock guard( _mutex );
@@ -254,7 +339,7 @@ AudioLayer::open_device(std::string pcm_p, std::string pcm_c, int flag)
snd_pcm_uframes_t period_size_in = getFrameSize() * getSampleRate() / 1000 ;
snd_pcm_uframes_t buffer_size_in = 4096;
snd_pcm_uframes_t threshold = getFrameSize() * getSampleRate() / 1000 ;
- snd_pcm_uframes_t period_size_out = 2048 ;
+ snd_pcm_uframes_t period_size_out = 1024 ;
snd_pcm_uframes_t buffer_size_out = 4096 ;
snd_pcm_sw_params_t *swparams = NULL;
@@ -280,20 +365,21 @@ AudioLayer::open_device(std::string pcm_p, std::string pcm_c, int flag)
if( err = snd_pcm_hw_params( _capture_handle, hwparams ) < 0) _debug(" Cannot set hw parameters (%s)\n", snd_strerror(err));
snd_pcm_hw_params_free( hwparams );
- snd_pcm_sw_params_alloca( &swparams );
- snd_pcm_sw_params_current( _capture_handle, swparams );
+ //snd_pcm_sw_params_malloc( &swparams );
+ //snd_pcm_sw_params_current( _capture_handle, swparams );
- if( err = snd_pcm_sw_params_set_start_threshold( _capture_handle, swparams, 1 ) < 0 ) _debug(" Cannot set start threshold (%s)\n", snd_strerror(err));
- if( err = snd_pcm_sw_params_set_stop_threshold( _capture_handle, swparams, buffer_size_in ) < 0 ) _debug(" Cannot get stop threshold (%s)\n", snd_strerror(err));
- if( err = snd_pcm_sw_params( _capture_handle, swparams ) < 0 ) _debug(" Cannot set sw parameters (%s)\n", snd_strerror(err));
- device_closed = false;
+ //if( err = snd_pcm_sw_params_set_start_threshold( _capture_handle, swparams, 1 ) < 0 ) _debug(" Cannot set start threshold (%s)\n", snd_strerror(err));
+ //if( err = snd_pcm_sw_params_set_stop_threshold( _capture_handle, swparams, buffer_size_in ) < 0 ) _debug(" Cannot get stop threshold (%s)\n", snd_strerror(err));
+ //if( err = snd_pcm_sw_params( _capture_handle, swparams ) < 0 ) _debug(" Cannot set sw parameters (%s)\n", snd_strerror(err));
+ deviceClosed = false;
+ //snd_pcm_sw_params_free( swparams );
}
if(flag == SFL_PCM_BOTH || flag == SFL_PCM_PLAYBACK)
{
_debug(" Opening playback device %s\n", pcm_p.c_str());
- if(err = snd_pcm_open(&_playback_handle, pcm_p.c_str(), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK) < 0){
+ if(err = snd_pcm_open(&_playback_handle, pcm_p.c_str(), SND_PCM_STREAM_PLAYBACK, 0 ) < 0){
_debug(" Error while opening playback device %s (%s)\n", pcm_p.c_str(), snd_strerror(err));
return false;
}
@@ -312,71 +398,91 @@ AudioLayer::open_device(std::string pcm_p, std::string pcm_c, int flag)
snd_pcm_hw_params_free( hwparams );
snd_pcm_uframes_t val = 1024 ;
- snd_pcm_sw_params_alloca( &swparams );
+ snd_pcm_sw_params_malloc( &swparams );
snd_pcm_sw_params_current( _playback_handle, swparams );
- if( err = snd_pcm_sw_params_set_start_threshold( _playback_handle, swparams, val ) < 0 ) _debug(" Cannot set start threshold (%s)\n", snd_strerror(err));
- if( err = snd_pcm_sw_params_set_stop_threshold( _playback_handle, swparams, buffer_size_out ) < 0 ) _debug(" Cannot set stop threshold (%s)\n", snd_strerror(err));
+ if( err = snd_pcm_sw_params_set_start_threshold( _playback_handle, swparams, period_size_out ) < 0 ) _debug(" Cannot set start threshold (%s)\n", snd_strerror(err));
+ //if( err = snd_pcm_sw_params_set_stop_threshold( _playback_handle, swparams, buffer_size_out ) < 0 ) _debug(" Cannot set stop threshold (%s)\n", snd_strerror(err));
+ if( err = snd_pcm_sw_params_set_avail_min( _playback_handle, swparams, period_size_out) < 0) _debug(" Cannot set min avail (%s)\n" , snd_strerror(err));
+ if( err = snd_pcm_sw_params_set_xfer_align( _playback_handle , swparams , 1 ) < 0) _debug( "Cannot align transfer (%s)\n", snd_strerror(err));
if( err = snd_pcm_sw_params( _playback_handle, swparams ) < 0 ) _debug(" Cannot set sw parameters (%s)\n", snd_strerror(err));
- device_closed = false;
+ snd_pcm_sw_params_free( swparams );
+
+ if ( err = snd_async_add_pcm_handler( &_AsyncHandler, _playback_handle , AlsaCallBack, this ) < 0) _debugAlsa(" Unable to install the async callback handler (%s)\n", snd_strerror(err));
+ deviceClosed = false;
+
+ //fillHWBuffer();
+
+ //if( err = snd_pcm_start(_playback_handle ) < 0 ) _debug(" Cannot start\n");
+
}
return true;
}
+//TODO EAGAIN error case
+//TODO first frame causes broken pipe (underrun) because not enough data are send --> make the handle wait to be ready
int
AudioLayer::write(void* buffer, int length)
{
-
- if(device_closed || _playback_handle == NULL)
- return 0;
-
int bytes;
- short* buff = (short*) buffer;
- int result = 0 ;
-
-
+ /* if(deviceClosed || _playback_handle == NULL)
+ return 0;
+ int framesPerBuffer = 882;
+ bool playUrgent = false;
+ SFLDataFormat ringtone[ framesPerBuffer ];
+ AudioLoop* tone = _manager->getTelephoneTone();
+ if( tone != 0){
+ _debug("Play tone \n");
+ tone->getNext(ringtone, framesPerBuffer , _manager->getSpkrVolume());
+ playUrgent = true;
+ }
+ else if( (tone=_manager->getTelephoneFile()) != 0 ){
+ _debug("Play tone also \n");
+ tone->getNext(ringtone, framesPerBuffer , _manager->getSpkrVolume());
+ playUrgent = true;
+ }
+
+ int bytes;*/
snd_pcm_uframes_t frames = snd_pcm_bytes_to_frames( _playback_handle, length);
-
- while( frames > 0 )
- {
- bytes = snd_pcm_writei( _playback_handle, buff, frames);
- if( bytes == -EAGAIN || (bytes >=0 && bytes < frames))
+ bytes = snd_pcm_writei( _playback_handle, buffer, frames);
+
+ if( bytes == -EAGAIN)
{
- //snd_pcm_wait( _playback_handle, 20 );
- break;
+ _debugAlsa(" (%s)\n", snd_strerror( bytes ));
+ snd_pcm_resume( _playback_handle );
}
+ else if(bytes >=0 && bytes < frames)
+ {
+ _debugAlsa("Short write - Frames remaining = %d\n", frames);
+ }
else if( bytes == -EPIPE )
{
- _debugAlsa(" %d Alsa error from writei (%s)\n", bytes, snd_strerror(bytes));
- snd_pcm_prepare( _playback_handle );
- snd_pcm_writei( _playback_handle , buff , frames );
- break;
+ _debugAlsa(" %d Alsa error from writei (%s)\n", bytes, snd_strerror(bytes));
+ snd_pcm_prepare( _playback_handle );
+ snd_pcm_writei( _playback_handle , buffer , frames );
}
else if( bytes == -ESTRPIPE )
{
- _debug(" Playback suspend \n");
+ _debugAlsa(" Playback suspend (%s)\n", snd_strerror(bytes));
snd_pcm_resume( _playback_handle );
- break;
}
- if( bytes != frames)
- {
- _debug(" Short write\n");
- frames -= bytes;
- buff += bytes;
- }
-
+ else if( bytes == -EBADFD)
+ {
+ _debugAlsa(" PCM is not in the right state (%s)\n", snd_strerror( bytes ));
}
+ snd_pcm_sframes_t delay;
+ //snd_pcm_delay( _playback_handle, &delay);
+ //snd_pcm_avail_update( _playback_handle );
return 0;
-
}
int
AudioLayer::read( void* buffer, int toCopy)
{
- if(device_closed || _capture_handle == NULL)
+ if(deviceClosed || _capture_handle == NULL)
return 0;
int err;
if(snd_pcm_state( _capture_handle ) == SND_PCM_STATE_XRUN)
diff --git a/src/audio/audiolayer.h b/src/audio/audiolayer.h
index 78f1f9dd4a5a9e1f9e66654c274741c6accec130..d60e4fe95fb2b701bc08c47f79a37a9933224bbd 100644
--- a/src/audio/audiolayer.h
+++ b/src/audio/audiolayer.h
@@ -28,6 +28,7 @@
#include "../global.h"
#include "audiodevice.h"
+#include "ringbuffer.h"
#include <vector>
#include <alsa/asoundlib.h>
@@ -37,6 +38,7 @@
#include <sstream>
#define FRAME_PER_BUFFER 160
+class RingBuffer;
class ManagerImpl;
class AudioLayer {
@@ -56,7 +58,7 @@ class AudioLayer {
* SFL_PCM_PLAYBACK
* SFL_PCM_BOTH
*/
- bool openDevice(int, int, int, int, int);
+ bool openDevice(int, int, int, int, int, std::string);
/*
* Start the capture stream. The playback starts according th its threshold
@@ -70,6 +72,8 @@ class AudioLayer {
* ALSA Library API
*/
void stopStream(void);
+
+ void fillHWBuffer( void) ;
void sleep(int);
@@ -108,9 +112,10 @@ class AudioLayer {
* @return int The number of bytes played
*/
int playSamples(void* buffer, int toCopy);
- int playRingTone( void* buffer, int toCopy);
int putUrgent(void* buffer, int toCopy);
+ void stopPlaybackStream( void );
+
/*
* Query the audio devices for number of bytes available in the hardware ring buffer
* @return int The number of bytes available
@@ -177,6 +182,8 @@ class AudioLayer {
*/
unsigned int getFrameSize() { return _frameSize; }
+ std::string getAudioPlugin( void ) { return _audioPlugin; }
+
int getDeviceCount();
/**
@@ -184,8 +191,17 @@ class AudioLayer {
*/
void toggleEchoTesting();
+
private:
+ static void AlsaCallBack( snd_async_handler_t* );
+ snd_async_handler_t *_AsyncHandler;
+ snd_pcm_t* _playback_handle;
+ snd_pcm_t* _capture_handle;
+ RingBuffer _urgentBuffer;
+ RingBuffer _mainBuffer;
+ void updateBuffers( void );
+
/*
* Open the specified device.
* ALSA Library API
@@ -231,15 +247,14 @@ class AudioLayer {
* Handle to manipulate capture and playback streams
* ALSA Library API
*/
- snd_pcm_t* _playback_handle;
- snd_pcm_t* _capture_handle;
-
+
+
/*
* Enable to determine if the devices are opened or not
* true if the devices are closed
* false otherwise
*/
- bool device_closed;
+ bool deviceClosed;
/**
* Number of audio cards on which stream has been opened
@@ -259,6 +274,8 @@ class AudioLayer {
*/
unsigned int _frameSize;
+ std::string _audioPlugin;
+
/**
* Input channel (mic) should be 1 mono
*/
diff --git a/src/dbus/configurationmanager-glue.h b/src/dbus/configurationmanager-glue.h
index 196851725b3a852582c6fb29e1cfc10a950e815e..f11da96a601fc933068d7544c795d2421504e9b0 100644
--- a/src/dbus/configurationmanager-glue.h
+++ b/src/dbus/configurationmanager-glue.h
@@ -36,8 +36,10 @@ public:
register_method(ConfigurationManager, getCodecDetails, _getCodecDetails_stub);
register_method(ConfigurationManager, getActiveCodecList, _getActiveCodecList_stub);
register_method(ConfigurationManager, setActiveCodecList, _setActiveCodecList_stub);
- register_method(ConfigurationManager, getAudioManagerList, _getAudioManagerList_stub);
- register_method(ConfigurationManager, setAudioManager, _setAudioManager_stub);
+ register_method(ConfigurationManager, getInputAudioPluginList, _getInputAudioPluginList_stub);
+ register_method(ConfigurationManager, getOutputAudioPluginList, _getOutputAudioPluginList_stub);
+ register_method(ConfigurationManager, setInputAudioPlugin, _setInputAudioPlugin_stub);
+ register_method(ConfigurationManager, setOutputAudioPlugin, _setOutputAudioPlugin_stub);
register_method(ConfigurationManager, getAudioOutputDeviceList, _getAudioOutputDeviceList_stub);
register_method(ConfigurationManager, setAudioOutputDevice, _setAudioOutputDevice_stub);
register_method(ConfigurationManager, getAudioInputDeviceList, _getAudioInputDeviceList_stub);
@@ -131,14 +133,24 @@ public:
{ "list", "as", true },
{ 0, 0, 0 }
};
- static ::DBus::IntrospectedArgument getAudioManagerList_args[] =
+ static ::DBus::IntrospectedArgument getInputAudioPluginList_args[] =
{
{ "list", "as", false },
{ 0, 0, 0 }
};
- static ::DBus::IntrospectedArgument setAudioManager_args[] =
+ static ::DBus::IntrospectedArgument getOutputAudioPluginList_args[] =
{
- { "audioManager", "s", true },
+ { "list", "as", false },
+ { 0, 0, 0 }
+ };
+ static ::DBus::IntrospectedArgument setInputAudioPlugin_args[] =
+ {
+ { "audioPlugin", "s", true },
+ { 0, 0, 0 }
+ };
+ static ::DBus::IntrospectedArgument setOutputAudioPlugin_args[] =
+ {
+ { "audioPlugin", "s", true },
{ 0, 0, 0 }
};
static ::DBus::IntrospectedArgument getAudioOutputDeviceList_args[] =
@@ -199,8 +211,10 @@ public:
{ "getCodecDetails", getCodecDetails_args },
{ "getActiveCodecList", getActiveCodecList_args },
{ "setActiveCodecList", setActiveCodecList_args },
- { "getAudioManagerList", getAudioManagerList_args },
- { "setAudioManager", setAudioManager_args },
+ { "getInputAudioPluginList", getInputAudioPluginList_args },
+ { "getOutputAudioPluginList", getOutputAudioPluginList_args },
+ { "setInputAudioPlugin", setInputAudioPlugin_args },
+ { "setOutputAudioPlugin", setOutputAudioPlugin_args },
{ "getAudioOutputDeviceList", getAudioOutputDeviceList_args },
{ "setAudioOutputDevice", setAudioOutputDevice_args },
{ "getAudioInputDeviceList", getAudioInputDeviceList_args },
@@ -256,8 +270,10 @@ public:
virtual std::vector< ::DBus::String > getCodecDetails( const ::DBus::Int32& payload ) = 0;
virtual std::vector< ::DBus::String > getActiveCodecList( ) = 0;
virtual void setActiveCodecList( const std::vector< ::DBus::String >& list ) = 0;
- virtual std::vector< ::DBus::String > getAudioManagerList( ) = 0;
- virtual void setAudioManager( const ::DBus::String& audioManager ) = 0;
+ virtual std::vector< ::DBus::String > getInputAudioPluginList( ) = 0;
+ virtual std::vector< ::DBus::String > getOutputAudioPluginList( ) = 0;
+ virtual void setInputAudioPlugin( const ::DBus::String& audioPlugin ) = 0;
+ virtual void setOutputAudioPlugin( const ::DBus::String& audioPlugin ) = 0;
virtual std::vector< ::DBus::String > getAudioOutputDeviceList( ) = 0;
virtual void setAudioOutputDevice( const ::DBus::Int32& index ) = 0;
virtual std::vector< ::DBus::String > getAudioInputDeviceList( ) = 0;
@@ -444,22 +460,41 @@ private:
::DBus::ReturnMessage reply(call);
return reply;
}
- ::DBus::Message _getAudioManagerList_stub( const ::DBus::CallMessage& call )
+ ::DBus::Message _getInputAudioPluginList_stub( const ::DBus::CallMessage& call )
+ {
+ ::DBus::MessageIter ri = call.reader();
+
+ std::vector< ::DBus::String > argout1 = getInputAudioPluginList();
+ ::DBus::ReturnMessage reply(call);
+ ::DBus::MessageIter wi = reply.writer();
+ wi << argout1;
+ return reply;
+ }
+ ::DBus::Message _getOutputAudioPluginList_stub( const ::DBus::CallMessage& call )
{
::DBus::MessageIter ri = call.reader();
- std::vector< ::DBus::String > argout1 = getAudioManagerList();
+ std::vector< ::DBus::String > argout1 = getOutputAudioPluginList();
::DBus::ReturnMessage reply(call);
::DBus::MessageIter wi = reply.writer();
wi << argout1;
return reply;
}
- ::DBus::Message _setAudioManager_stub( const ::DBus::CallMessage& call )
+ ::DBus::Message _setInputAudioPlugin_stub( const ::DBus::CallMessage& call )
+ {
+ ::DBus::MessageIter ri = call.reader();
+
+ ::DBus::String argin1; ri >> argin1;
+ setInputAudioPlugin(argin1);
+ ::DBus::ReturnMessage reply(call);
+ return reply;
+ }
+ ::DBus::Message _setOutputAudioPlugin_stub( const ::DBus::CallMessage& call )
{
::DBus::MessageIter ri = call.reader();
::DBus::String argin1; ri >> argin1;
- setAudioManager(argin1);
+ setOutputAudioPlugin(argin1);
::DBus::ReturnMessage reply(call);
return reply;
}
diff --git a/src/dbus/configurationmanager-introspec.xml b/src/dbus/configurationmanager-introspec.xml
index b0b82baa010452ab6309bb666234b08ba6c09a89..09a92e921ccc9172168fcf9c43315cc3e38fc5b2 100644
--- a/src/dbus/configurationmanager-introspec.xml
+++ b/src/dbus/configurationmanager-introspec.xml
@@ -81,11 +81,17 @@
<!-- Audio devices methods -->
- <method name="getAudioManagerList">
+ <method name="getInputAudioPluginList">
<arg type="as" name="list" direction="out"/>
</method>
- <method name="setAudioManager">
- <arg type="s" name="audioManager" direction="in"/>
+ <method name="getOutputAudioPluginList">
+ <arg type="as" name="list" direction="out"/>
+ </method>
+ <method name="setInputAudioPlugin">
+ <arg type="s" name="audioPlugin" direction="in"/>
+ </method>
+ <method name="setOutputAudioPlugin">
+ <arg type="s" name="audioPlugin" direction="in"/>
</method>
<method name="getAudioOutputDeviceList">
diff --git a/src/dbus/configurationmanager.cpp b/src/dbus/configurationmanager.cpp
index 980d1815f577feb40ca03188d0f0a4666f9ef0f6..0f80ed1d27f6d2b13245fce1913f57a5b9f709ea 100644
--- a/src/dbus/configurationmanager.cpp
+++ b/src/dbus/configurationmanager.cpp
@@ -126,19 +126,35 @@ ConfigurationManager::setActiveCodecList( const std::vector< ::DBus::String >& l
}
// Audio devices related methods
-std::vector< ::DBus::String >
-ConfigurationManager::getAudioManagerList()
+ std::vector< ::DBus::String >
+ConfigurationManager::getInputAudioPluginList()
{
- _debug("ConfigurationManager::getAudioManagerList received\n");
- return Manager::instance().getAudioManagerList();
+ _debug("ConfigurationManager::getInputAudioPluginList received\n");
+ return Manager::instance().getInputAudioPluginList();
}
-void
-ConfigurationManager::setAudioManager(const ::DBus::String& audioManager)
+
+ std::vector< ::DBus::String >
+ConfigurationManager::getOutputAudioPluginList()
{
- _debug("ConfigurationManager::setAudioManager received\n");
- return Manager::instance().setAudioManager(audioManager);
+ _debug("ConfigurationManager::getOutputAudioPluginList received\n");
+ return Manager::instance().getOutputAudioPluginList();
}
-std::vector< ::DBus::String >
+
+ void
+ConfigurationManager::setInputAudioPlugin(const ::DBus::String& audioPlugin)
+{
+ _debug("ConfigurationManager::setInputAudioPlugin received\n");
+ return Manager::instance().setInputAudioPlugin(audioPlugin);
+}
+
+ void
+ConfigurationManager::setOutputAudioPlugin(const ::DBus::String& audioPlugin)
+{
+ _debug("ConfigurationManager::setOutputAudioPlugin received\n");
+ return Manager::instance().setOutputAudioPlugin(audioPlugin);
+}
+
+ std::vector< ::DBus::String >
ConfigurationManager::getAudioOutputDeviceList()
{
_debug("ConfigurationManager::getAudioOutputDeviceList received\n");
diff --git a/src/dbus/configurationmanager.h b/src/dbus/configurationmanager.h
index 9e800b04aa1ae24972465b6014efb5019a048cf2..6c13b642ab0b6bdd2757c6a1f5f1dcdfc4e4b9fe 100644
--- a/src/dbus/configurationmanager.h
+++ b/src/dbus/configurationmanager.h
@@ -52,8 +52,10 @@ public:
std::vector< ::DBus::String > getActiveCodecList( );
void setActiveCodecList( const std::vector< ::DBus::String >& list );
- std::vector< ::DBus::String > getAudioManagerList();
- void setAudioManager(const ::DBus::String& audioManager);
+ std::vector< ::DBus::String > getInputAudioPluginList();
+ std::vector< ::DBus::String > getOutputAudioPluginList();
+ void setInputAudioPlugin(const ::DBus::String& audioPlugin);
+ void setOutputAudioPlugin(const ::DBus::String& audioPlugin);
std::vector< ::DBus::String > getAudioOutputDeviceList();
void setAudioOutputDevice(const ::DBus::Int32& index);
std::vector< ::DBus::String > getAudioInputDeviceList();
diff --git a/src/global.h b/src/global.h
index 53735727161d55eba06e0976b73edaa18a1d99ae..f410aa3888ad504e17fe60fa5c1066471ab1ed50 100644
--- a/src/global.h
+++ b/src/global.h
@@ -27,17 +27,14 @@
typedef float float32;
typedef short int16;
-//#define DATAFORMAT_IS_FLOAT
#ifdef DATAFORMAT_IS_FLOAT
#define SFLDataFormat float32
-#define SFLPortaudioFormat portaudio::FLOAT32
-#define SFLPortaudioFormatString "Float32"
+#define SFLDataFormatString "Float32"
#define SFLDataAmplitude 0.05
#define SFLConvertInt16(s) ((float)(s)-16384.0)/16384.0
#else
#define SFLDataFormat int16
-#define SFLPortaudioFormat portaudio::INT16
-#define SFLPortaudioFormatString "Int16"
+#define SFLDataFormatString "Int16"
#define SFLDataAmplitude (32767 >> 4)
#define SFLConvertInt16(s) (s)
#endif
@@ -74,6 +71,7 @@ typedef short int16;
#define CHANNELS 2
#define SIZEBUF 1024*1024
+#define PCM_HW "hw"
#define PCM_PLUGHW "plughw"
#define PCM_FRONT "plug:front"
#define PCM_DEFAULT "default"
diff --git a/src/managerimpl.cpp b/src/managerimpl.cpp
index 171586e51a2359cee7cfb9888dc3c67dd72b2640..e98012be65a914ffcc36f9ccc599c104d9950e1b 100644
--- a/src/managerimpl.cpp
+++ b/src/managerimpl.cpp
@@ -350,7 +350,7 @@ ManagerImpl::offHoldCall(const CallID& id)
switchCall(id);
if (returnValue) {
try {
- getAudioDriver()->startStream();
+ //getAudioDriver()->startStream();
} catch(...) {
_debugException("! Manager Off hold could not start audio stream");
}
@@ -550,24 +550,22 @@ ManagerImpl::playDtmf(char code)
// Put buffer to urgentRingBuffer
// put the size in bytes...
// so size * 1 channel (mono) * sizeof (bytes for the data)
- audiolayer->putUrgent(_buf, size * sizeof(SFLDataFormat));
+ audiolayer->playSamples(_buf, size * sizeof(SFLDataFormat));
+ //audiolayer->putUrgent(_buf, size * sizeof(SFLDataFormat));
- try {
- // We activate the stream if it's not active yet.
- if (!audiolayer->isStreamActive()) {
- //audiolayer->startStream();
- } else {
- audiolayer->sleep(pulselen); // in milliseconds
- }
- } catch(...) {
- _debugException("Portaudio exception when playing a dtmf");
+ // We activate the stream if it's not active yet.
+ if (!audiolayer->isStreamActive()) {
+ //audiolayer->startStream();
+ } else {
+ _debugAlsa("send dtmf - sleep\n");
+ audiolayer->sleep(pulselen); // in milliseconds
}
- returnValue = true;
}
+ returnValue = true;
- // TODO: add caching
- delete[] _buf; _buf = 0;
- return returnValue;
+// TODO: add caching
+delete[] _buf; _buf = 0;
+return returnValue;
}
// Multi-thread
@@ -617,8 +615,8 @@ ManagerImpl::incomingCall(Call* call, const AccountID& accountId)
associateCallToAccount(call->getCallId(), accountId);
if ( !hasCurrentCall() ) {
- _debug("INCOMING CALL!!!!!!\n");
call->setConnectionState(Call::Ringing);
+ _debugAlsa(" call ringtone() method\n ");
ringtone();
switchCall(call->getCallId());
} else {
@@ -806,7 +804,6 @@ ManagerImpl::registrationFailed(const AccountID& accountid)
bool
ManagerImpl::playATone(Tone::TONEID toneId) {
int hasToPlayTone = getConfigInt(SIGNALISATION, PLAY_TONES);
-
if (!hasToPlayTone) return false;
if (_telephoneTone != 0) {
@@ -816,14 +813,11 @@ ManagerImpl::playATone(Tone::TONEID toneId) {
AudioLoop* audioloop = getTelephoneTone();
unsigned int nbSampling = audioloop->getSize();
- _debug("Telephone tone size = %d\n", nbSampling);
AudioLayer* audiolayer = getAudioDriver();
- //int chunks = 2000;
SFLDataFormat buf[nbSampling];
- audioloop->getNext(buf, audioloop->getSize());
- if (audiolayer) {
- audiolayer->putUrgent( buf, sizeof(SFLDataFormat)*nbSampling);
- ///buf += chunks;
+ audioloop->getNext(buf, (int) nbSampling);
+ if ( audiolayer ) {
+ //audiolayer->putUrgent( buf, sizeof(SFLDataFormat)*nbSampling );
}
else
return false;
@@ -840,12 +834,8 @@ ManagerImpl::stopTone(bool stopAudio=true) {
if (!hasToPlayTone) return;
if (stopAudio) {
- try {
- AudioLayer* audiolayer = getAudioDriver();
- if (audiolayer) { audiolayer->stopStream(); }
- } catch(...) {
- _debugException("Stop tone and stop stream");
- }
+ AudioLayer* audiolayer = getAudioDriver();
+ if (audiolayer) { audiolayer->stopStream(); }
}
_toneMutex.enterMutex();
@@ -892,8 +882,6 @@ ManagerImpl::ringback () {
void
ManagerImpl::ringtone()
{
-
-
int hasToPlayTone = getConfigInt(SIGNALISATION, PLAY_TONES);
if (!hasToPlayTone) { return; }
@@ -917,14 +905,9 @@ ManagerImpl::ringtone()
_toneMutex.leaveMutex();
int size = _audiofile.getSize();
SFLDataFormat output[ size ];
- //_debugAlsa("Size = %i\n", size);
- //_audiofile.getNext(output, size , 100);
- try {
- //audiolayer->putUrgent( output, size);
- _debug(" Ringtone if everything would have worked fine\n");
- } catch(...) {
- _debugException("Audio file couldn't start audio stream");
- }
+ _debugAlsa("Size = %i\n", size);
+ _audiofile.getNext(output, size , 100);
+ audiolayer->playSamples( output , size );
} else {
ringback();
}
@@ -971,7 +954,7 @@ ManagerImpl::notificationIncomingCall(void) {
unsigned int nbSampling = tone.getSize();
SFLDataFormat buf[nbSampling];
tone.getNext(buf, tone.getSize());
- audiolayer->putUrgent(buf, sizeof(SFLDataFormat)*nbSampling);
+ audiolayer->playSamples(buf, sizeof(SFLDataFormat)*nbSampling);
}
}
@@ -1222,27 +1205,66 @@ ManagerImpl::getCodecDetails( const ::DBus::Int32& payload )
}
/**
- * Get list of supported audio manager
+ * Get list of supported input audio plugin
*/
std::vector<std::string>
-ManagerImpl::getAudioManagerList(void)
+ManagerImpl::getInputAudioPluginList(void)
{
std::vector<std::string> v;
- _debug("Get audio manager list");
+ _debug("Get input audio plugin list");
+
+ v.push_back("default");
+ v.push_back("surround40");
+ v.push_back("plug:hw");
- // Return only ALSA for now
- v.push_back("ALSA");
return v;
}
/**
- * Set audio manager (always put ALSA)
+ * Get list of supported output audio plugin
+ */
+ std::vector<std::string>
+ManagerImpl::getOutputAudioPluginList(void)
+{
+ std::vector<std::string> v;
+ _debug("Get output audio plugin list");
+
+ v.push_back("default");
+ v.push_back("plug:hw");
+ v.push_back("plug:dmix");
+ v.push_back("plug:surround40");
+
+ return v;
+}
+
+/**
+ * Set input audio plugin
+ */
+ void
+ManagerImpl::setInputAudioPlugin(const std::string& audioPlugin)
+{
+ _debug("Set input audio plugin\n");
+ _audiodriver -> openDevice( _audiodriver -> getIndexIn(),
+ _audiodriver -> getIndexOut(),
+ _audiodriver -> getSampleRate(),
+ _audiodriver -> getFrameSize(),
+ SFL_PCM_CAPTURE,
+ audioPlugin);
+}
+
+/**
+ * Set output audio plugin
*/
void
-ManagerImpl::setAudioManager(const std::string& audioManager)
+ManagerImpl::setOutputAudioPlugin(const std::string& audioPlugin)
{
- _debug("Set audio manager\n");
- // Do nothing for now
+ _debug("Set output audio plugin\n");
+ _audiodriver -> openDevice( _audiodriver -> getIndexIn(),
+ _audiodriver -> getIndexOut(),
+ _audiodriver -> getSampleRate(),
+ _audiodriver -> getFrameSize(),
+ SFL_PCM_PLAYBACK,
+ audioPlugin);
}
/**
@@ -1266,7 +1288,8 @@ ManagerImpl::setAudioOutputDevice(const int index)
index,
_audiodriver->getSampleRate(),
_audiodriver->getFrameSize(),
- SFL_PCM_PLAYBACK);
+ SFL_PCM_PLAYBACK,
+ _audiodriver->getAudioPlugin());
}
/**
@@ -1290,7 +1313,8 @@ ManagerImpl::setAudioInputDevice(const int index)
_audiodriver->getIndexOut(),
_audiodriver->getSampleRate(),
_audiodriver->getFrameSize(),
- SFL_PCM_CAPTURE);
+ SFL_PCM_CAPTURE,
+ _audiodriver->getAudioPlugin());
}
/**
@@ -1363,7 +1387,7 @@ ManagerImpl::selectAudioDriver (void)
//}
_debugInit(" AudioLayer Opening Device");
_audiodriver->setErrorMessage("");
- _audiodriver->openDevice(noDeviceIn, noDeviceOut, sampleRate, frameSize, SFL_PCM_BOTH);
+ _audiodriver->openDevice(noDeviceIn, noDeviceOut, sampleRate, frameSize, SFL_PCM_BOTH, PCM_PLUGHW);
}
/**
diff --git a/src/managerimpl.h b/src/managerimpl.h
index 9aae25f52807e77a85158f3bc5dea7c5703243d7..9f5068b304b9cd48947a6811a321b1c20d9d7063 100644
--- a/src/managerimpl.h
+++ b/src/managerimpl.h
@@ -278,15 +278,26 @@ public:
std::vector< ::DBus::String > getCodecDetails( const ::DBus::Int32& payload);
/**
- * Get a list of supported audio managers
+ * Get a list of supported input audio plugin
* @return List of names
*/
- std::vector< std::string> getAudioManagerList(void);
+ std::vector< std::string> getInputAudioPluginList(void);
/**
- * Set audio manager (always put ALSA)
+ * Get a list of supported output audio plugin
+ * @return List of names
+ */
+ std::vector< std::string> getOutputAudioPluginList(void);
+
+ /**
+ * Set input audio plugin
+ */
+ void setInputAudioPlugin(const std::string& audioPlugin);
+
+ /**
+ * Set output audio plugin
*/
- void setAudioManager(const std::string& audioManager);
+ void setOutputAudioPlugin(const std::string& audioPlugin);
/**
* Get list of supported audio output device