diff --git a/sflphone-client-gnome/src/config/audioconf.c b/sflphone-client-gnome/src/config/audioconf.c
index 25d8a76118c868a77e69a70df689c6023a3df04a..dcaa7e85ce1c909077a2f0b644adfa327a14b716 100644
--- a/sflphone-client-gnome/src/config/audioconf.c
+++ b/sflphone-client-gnome/src/config/audioconf.c
@@ -94,7 +94,7 @@ 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;
@@ -103,7 +103,7 @@ 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;
@@ -235,7 +235,7 @@ select_active_ringtone_audio_device()
// Select active ringtone device on server
devices = dbus_get_current_audio_devices_index();
- currentDeviceIndex = atoi(devices[0]);
+ currentDeviceIndex = atoi(devices[2]);
DEBUG("audio device index for ringtone = %d", currentDeviceIndex);
model = gtk_combo_box_get_model(GTK_COMBO_BOX(ringtone));
@@ -787,7 +787,7 @@ GtkWidget* alsa_box()
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);
diff --git a/sflphone-client-gnome/src/dbus/configurationmanager-introspec.xml b/sflphone-client-gnome/src/dbus/configurationmanager-introspec.xml
index f38682c65e70350aae1bb21333c528ba46056d04..8a0c59c8b2b1aa69d46519915f26228a9d27584b 100644
--- a/sflphone-client-gnome/src/dbus/configurationmanager-introspec.xml
+++ b/sflphone-client-gnome/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"/>
diff --git a/sflphone-client-gnome/src/dbus/dbus.c b/sflphone-client-gnome/src/dbus/dbus.c
index 1c6749099269ca661ab2322af019d2edffa05512..08cf832c4cc3f4856ea5b334896303215cc5a9d8 100644
--- a/sflphone-client-gnome/src/dbus/dbus.c
+++ b/sflphone-client-gnome/src/dbus/dbus.c
@@ -1297,44 +1297,27 @@ 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;
diff --git a/sflphone-client-gnome/src/dbus/dbus.h b/sflphone-client-gnome/src/dbus/dbus.h
index 394b1505695bf84b6c5cfa6e3e44cc5cd14461b8..80d99f19619f7d3ce4705dc897119fffadb550e4 100644
--- a/sflphone-client-gnome/src/dbus/dbus.h
+++ b/sflphone-client-gnome/src/dbus/dbus.h
@@ -241,17 +241,11 @@ 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 6cf89046a85f6085a4427da70e4032721f8e8a9d..ec08bac656fb0135a0d436c8959d42250ae56805 100644
--- a/sflphone-common/src/audio/alsa/alsalayer.cpp
+++ b/sflphone-common/src/audio/alsa/alsalayer.cpp
@@ -1,6 +1,7 @@
/*
* Copyright (C) 2008 2009 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
@@ -147,9 +148,27 @@ AlsaLayer::startStream (void)
if(_echoCanceller)
_echoCanceller->resetAlgorithm();
- std::string pcmp = buildDeviceTopo (_audioPlugin, _indexOut, 0);
- std::string pcmc = buildDeviceTopo (_audioPlugin, _indexIn, 0);
- std::string pcmr = buildDeviceTopo (_audioPlugin, _indexRing, 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, pcmr, SFL_PCM_PLAYBACK);
@@ -323,9 +342,11 @@ void AlsaLayer::closePlaybackStream (void)
_debug("Audio: Close ALSA playback");
- if((err = snd_pcm_close(_RingtoneHandle)) < 0) {
- _warn("Audio: Error: Closing ALSA ringtone: %s", snd_strerror(err));
- }
+ 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)
_warn("Audio: Error: Closing ALSA playback: %s", snd_strerror (err));
@@ -539,15 +560,18 @@ AlsaLayer::open_device (std::string pcm_p, std::string pcm_c, std::string pcm_r,
return false;
}
- 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 (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);
+ if(!alsa_set_params(_RingtoneHandle, 1, getSampleRate())) {
+ _warn("Audio: Error: Ringtone failed");
+ snd_pcm_close(_RingtoneHandle);
+ }
}
open_playback ();
@@ -604,9 +628,9 @@ AlsaLayer::write (void* buffer, int length, snd_pcm_t * handle)
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)
+ if (snd_pcm_writei (handle, buffer , frames) <0)
_debugAlsa ("Audio: XRUN handling failed");
_trigger_request = true;
@@ -674,6 +698,7 @@ AlsaLayer::read (void* buffer, int toCopy)
void
AlsaLayer::handle_xrun_capture (void)
{
+ _debugAlsa ("Audio: Handle xrun capture");
snd_pcm_status_t* status;
snd_pcm_status_alloca (&status);
@@ -687,22 +712,25 @@ 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 ("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 ("Audio: 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 ();
@@ -715,10 +743,10 @@ 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;
@@ -850,16 +878,25 @@ void AlsaLayer::audioCallback(void)
spkrVolume = _manager->getSpkrVolume();
micVolume = _manager->getMicVolume();
+ 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();
- //
+ 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 */
@@ -873,58 +910,44 @@ void AlsaLayer::audioCallback(void)
} else {
- tone = _manager->getTelephoneTone();
- file_tone = _manager->getTelephoneFile();
-
- toGet = framesPerBufferAlsa;
- maxBytes = toGet * sizeof (SFLDataFormat);
+ if (tone) {
- if (tone != 0) {
-
- out = (SFLDataFormat*) malloc (maxBytes * sizeof (SFLDataFormat));
- tone->getNext (out, toGet, spkrVolume);
- write (out , maxBytes, _PlaybackHandle);
+ 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 (maxBytes * sizeof (SFLDataFormat));
- file_tone->getNext (out, toGet, spkrVolume);
- write (out , maxBytes, _PlaybackHandle);
+ out = (SFLDataFormat *) malloc (playbackAvailBytes);
+ file_tone->getNext (out, playbackAvailSmpl, spkrVolume);
+ write (out, playbackAvailBytes, _PlaybackHandle);
- 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;
@@ -945,7 +968,7 @@ 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);
@@ -967,12 +990,12 @@ void AlsaLayer::audioCallback(void)
} 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), _PlaybackHandle);
+ bzero (zeros, playbackAvailBytes);
+ write (zeros, playbackAvailBytes, _PlaybackHandle);
free (zeros);
}
@@ -987,26 +1010,28 @@ void AlsaLayer::audioCallback(void)
}
- // Playback ringtone
- file_tone = _manager->getTelephoneFile();
+ if (file_tone && _RingtoneHandle) {
- toGet = framesPerBufferAlsa;
- maxBytes = toGet * sizeof (SFLDataFormat);
+ int ringtoneAvailSmpl = snd_pcm_avail_update(_RingtoneHandle);
+ int ringtoneAvailBytes = ringtoneAvailSmpl*sizeof(SFLDataFormat);
- if (file_tone != NULL) {
+ // _debug("RINGTONE: %d", ringtoneAvailSmpl);
- out = (SFLDataFormat*) malloc (maxBytes * sizeof (SFLDataFormat));
- file_tone->getNext (out, toGet, spkrVolume);
- write (out, maxBytes, _RingtoneHandle);
+ out = (SFLDataFormat *) malloc(ringtoneAvailBytes);
+ file_tone->getNext (out, ringtoneAvailSmpl, spkrVolume);
+ write (out, ringtoneAvailBytes, _RingtoneHandle);
free (out);
out = NULL;
- } else {
+ } else if (_RingtoneHandle) {
+
+ int ringtoneAvailSmpl = snd_pcm_avail_update(_RingtoneHandle);
+ int ringtoneAvailBytes = ringtoneAvailSmpl*sizeof(SFLDataFormat);
- out = (SFLDataFormat*) malloc ( maxBytes * sizeof (SFLDataFormat));
- memset(out, 0, maxBytes * sizeof (SFLDataFormat));
- write(out, maxBytes, _RingtoneHandle);
+ out = (SFLDataFormat *) malloc(ringtoneAvailBytes);
+ memset(out, 0, ringtoneAvailBytes);
+ write(out, ringtoneAvailBytes, _RingtoneHandle);
free(out);
out = NULL;
diff --git a/sflphone-common/src/audio/alsa/alsalayer.h b/sflphone-common/src/audio/alsa/alsalayer.h
index 6a4623a8a5c8d2be14aac5a5dc8feab567552b17..29a94711c89b71f1e27722022079f31bdd9008cd 100644
--- a/sflphone-common/src/audio/alsa/alsalayer.h
+++ b/sflphone-common/src/audio/alsa/alsalayer.h
@@ -1,6 +1,7 @@
/*
* Copyright (C) 2007 - 2008 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
@@ -214,8 +215,6 @@ class AlsaLayer : public AudioLayer {
* @return int The number of frames actually read
*/
int read( void* buffer, int toCopy);
-
-
/**
* Recover from XRUN state for capture
@@ -227,8 +226,8 @@ 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 );
/**
@@ -262,7 +261,7 @@ 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/pulseaudio/audiostream.cpp b/sflphone-common/src/audio/pulseaudio/audiostream.cpp
index be8606958f722016876ba1689f3da7aec36c1f7a..1e10ed4ec0aa06b7dece2c8f126a080223aeefb3 100644
--- a/sflphone-common/src/audio/pulseaudio/audiostream.cpp
+++ b/sflphone-common/src/audio/pulseaudio/audiostream.cpp
@@ -228,7 +228,7 @@ 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 8025a125bbf1f1393993a4f7cc8ea114a5c8951c..5f9b6cb19366de3c500a0b8416b9cc8cf1a2359d 100644
--- a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
+++ b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
@@ -770,7 +770,6 @@ void PulseLayer::writeToSpeaker (void)
} else {
AudioLoop* tone = _manager->getTelephoneTone();
- AudioLoop* file_tone = _manager->getTelephoneFile();
// flush remaining samples in _urgentRingBuffer
flushUrgent();
@@ -787,22 +786,6 @@ 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 {
@@ -876,7 +859,7 @@ void PulseLayer::writeToSpeaker (void)
} else {
- if ( (tone == 0) && (file_tone == 0)) {
+ if (tone == 0) {
SFLDataFormat* zeros = (SFLDataFormat*) pa_xmalloc (writeableSize);
@@ -977,27 +960,29 @@ 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/dbus/configurationmanager-introspec.xml b/sflphone-common/src/dbus/configurationmanager-introspec.xml
index f38682c65e70350aae1bb21333c528ba46056d04..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"/>
diff --git a/sflphone-common/src/dbus/configurationmanager.cpp b/sflphone-common/src/dbus/configurationmanager.cpp
index 7c2928349c6f248a72437bfb4d3de185ad951add..c7d309b3a7b90de5d5a181ebabda8c02a4f0eb0c 100644
--- a/sflphone-common/src/dbus/configurationmanager.cpp
+++ b/sflphone-common/src/dbus/configurationmanager.cpp
@@ -505,28 +505,13 @@ 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;
}
diff --git a/sflphone-common/src/dbus/configurationmanager.h b/sflphone-common/src/dbus/configurationmanager.h
index 27442ced3c363bc951d7f8969f628c86fdfa6e60..17ea89be6dbfbcea45c4233884ace5ec92e292ff 100644
--- a/sflphone-common/src/dbus/configurationmanager.h
+++ b/sflphone-common/src/dbus/configurationmanager.h
@@ -64,8 +64,7 @@ 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();
diff --git a/sflphone-common/src/eventthread.cpp b/sflphone-common/src/eventthread.cpp
index 3c548e5f7140daa6461497d9a47eaa5ea24a39c6..9ed09633fc2598db03f64fa9f1c3d5f885f7b1bd 100644
--- a/sflphone-common/src/eventthread.cpp
+++ b/sflphone-common/src/eventthread.cpp
@@ -54,7 +54,7 @@ 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 312f8407c5fb61f952d1d0e4239b5187d01e7f1d..b9812b725551257ea7754c4e5cf61ce75ea8dca2 100644
--- a/sflphone-common/src/global.h
+++ b/sflphone-common/src/global.h
@@ -86,6 +86,8 @@ 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 */
diff --git a/sflphone-common/src/managerimpl.cpp b/sflphone-common/src/managerimpl.cpp
index 53c1332c6ea439fb43114e1622d63d370de1a5ba..f32562d0d155221329780890632ca4b04bfdb8f3 100644
--- a/sflphone-common/src/managerimpl.cpp
+++ b/sflphone-common/src/managerimpl.cpp
@@ -2440,11 +2440,13 @@ std::vector<std::string> ManagerImpl::getAudioInputDeviceList (void) {
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;
}
diff --git a/sflphone-common/src/sip/sipvoiplink.cpp b/sflphone-common/src/sip/sipvoiplink.cpp
index 3a3df884beff8615dbf7dfc1f25dc24048138d23..a5642a3b2281936a58b970abe8eb06240dda9171 100644
--- a/sflphone-common/src/sip/sipvoiplink.cpp
+++ b/sflphone-common/src/sip/sipvoiplink.cpp
@@ -269,6 +269,7 @@ SIPVoIPLink::~SIPVoIPLink()
_debug("UserAgent: SIPVoIPLink destructor called");
terminate();
+
}
SIPVoIPLink* SIPVoIPLink::instance (const AccountID& id)
@@ -333,7 +334,6 @@ SIPVoIPLink::terminate()
initDone (false);
- _debug("Terminating");
}
void
@@ -1248,6 +1248,8 @@ SIPVoIPLink::refuse (const CallID& id)
call->getInvSession()->mod_data[getModId() ] = NULL;
+ removeCall(id);
+
terminateOneCall (id);
_debug("UserAgent: Refuse call completed");