--- last.fm-1.3.0.62.orig/src/output/alsa-playback/alsaaudio.cpp
+++ last.fm-1.3.0.62/src/output/alsa-playback/alsaaudio.cpp
@@ -63,6 +63,15 @@
}
+AlsaAudio::~AlsaAudio()
+{
+ // Close here just to be sure
+ // These are safe to call more than once
+ stopPlayback();
+ alsaClose();
+}
+
+
/******************************************************************************
* Device Detection
@@ -176,17 +185,27 @@
Device Setup
******************************************************************************/
-bool AlsaAudio::alsaSetup( QString device, snd_pcm_uframes_t periodSize, uint periodCount, snd_format *f )
+bool
+AlsaAudio::alsaOpen( QString device, AFormat format, unsigned int rate,
+ unsigned int channels, snd_pcm_uframes_t periodSize,
+ unsigned int periodCount )
{
- Q_DEBUG_BLOCK;
-
int err;
ssize_t hw_period_size;
snd_pcm_hw_params_t *hwparams;
snd_pcm_sw_params_t *swparams;
snd_pcm_uframes_t alsa_buffer_size, alsa_period_size;
snd_pcm_access_mask_t *mask;
-
+
+ Q_DEBUG_BLOCK << "Setting up Device:" << device;
+
+ inputf = snd_format_from_xmms( format, rate, channels );
+
+ // We'll be using this in alsaWrite
+ m_max_buffer_size = inputf->bps;
+
+ convertb = xmms_convert_buffers_new();
+
snd_output_stdio_attach( &logs, stderr, 0 );
alsa_convert_func = NULL;
@@ -194,7 +213,7 @@
alsa_frequency_convert_func = NULL;
free( outputf );
- outputf = snd_format_from_xmms( f->xmms_format, f->rate, f->channels );
+ outputf = snd_format_from_xmms( inputf->xmms_format, inputf->rate, inputf->channels );
qDebug() << "Opening device:" << device;
@@ -232,6 +251,7 @@
{
qDebug() << "No configuration available for playback: "
<< snd_strerror( -err );
+ alsaClose();
return false;
}
@@ -255,6 +275,7 @@
SND_PCM_ACCESS_RW_INTERLEAVED ) ) < 0 )
{
qDebug() << "Cannot set normal write mode: " << snd_strerror( -err );
+ alsaClose();
return false;
}
}
@@ -280,41 +301,44 @@
break;
}
}
- if ( outputf->format != f->format )
+ if ( outputf->format != inputf->format )
{
outputf->xmms_format = (AFormat)format_from_alsa( outputf->format );
- qDebug() << "Converting format from" << f->xmms_format << "to" << outputf->xmms_format;
+ qDebug() << "Converting format from" << inputf->xmms_format << "to" << outputf->xmms_format;
if ( outputf->xmms_format < 0 )
return -1;
- alsa_convert_func = xmms_convert_get_func( outputf->xmms_format, f->xmms_format );
+ alsa_convert_func = xmms_convert_get_func( outputf->xmms_format, inputf->xmms_format );
if ( alsa_convert_func == NULL )
{
- qDebug() << "Format translation needed, but not available. Input: " << f->xmms_format << "; Output: " << outputf->xmms_format ;
+ qDebug() << "Format translation needed, but not available. Input: " << inputf->xmms_format << "; Output: " << outputf->xmms_format ;
+ alsaClose();
return false;
}
}
else
{
qDebug() << "Sample format not available for playback: " << snd_strerror( -err );
+ alsaClose();
return false;
}
}
snd_pcm_hw_params_set_channels_near( alsa_pcm, hwparams, &outputf->channels );
- if ( outputf->channels != f->channels )
+ if ( outputf->channels != inputf->channels )
{
- qDebug() << "Converting channels from" << f->channels << "to" << outputf->channels;
+ qDebug() << "Converting channels from" << inputf->channels << "to" << outputf->channels;
alsa_stereo_convert_func =
xmms_convert_get_channel_func( outputf->xmms_format,
outputf->channels,
- f->channels );
+ inputf->channels );
if ( alsa_stereo_convert_func == NULL )
{
- qDebug() << "No stereo conversion available. Format: " << outputf->xmms_format << "; Input Channels: " << f->channels << "; Output Channels: " << outputf->channels ;
+ qDebug() << "No stereo conversion available. Format: " << outputf->xmms_format << "; Input Channels: " << inputf->channels << "; Output Channels: " << outputf->channels ;
+ alsaClose();
return false;
}
}
@@ -323,15 +347,17 @@
if ( outputf->rate == 0 )
{
qDebug() << "No usable samplerate available." ;
+ alsaClose();
return false;
}
- if ( outputf->rate != f->rate )
+ if ( outputf->rate != inputf->rate )
{
- qDebug() << "Converting samplerate from " << f->rate << " to " << outputf->rate ;
+ qDebug() << "Converting samplerate from " << inputf->rate << " to " << outputf->rate ;
if ( outputf->channels < 1 || outputf->channels > 2 )
{
qDebug() << "Unsupported number of channels: " << outputf->channels << ". Resample function not available" ;
alsa_frequency_convert_func = NULL;
+ alsaClose();
return false;
}
alsa_frequency_convert_func =
@@ -340,6 +366,7 @@
if ( alsa_frequency_convert_func == NULL )
{
qDebug() << "Resample function not available. Format " << outputf->xmms_format ;
+ alsaClose();
return false;
}
}
@@ -351,6 +378,7 @@
&periodSize, NULL ) ) < 0 )
{
qDebug() << "Set period size failed: " << snd_strerror( -err );
+ alsaClose();
return false;
}
@@ -358,6 +386,7 @@
&periodCount, 0 ) ) < 0 )
{
qDebug() << "Set period count failed: " << snd_strerror( -err );
+ alsaClose();
return false;
}
@@ -367,18 +396,21 @@
snd_pcm_hw_params_dump( hwparams, logs );
qDebug() << "Unable to install hw params" ;
+ alsaClose();
return false;
}
if ( ( err = snd_pcm_hw_params_get_buffer_size( hwparams, &alsa_buffer_size ) ) < 0 )
{
qDebug() << "snd_pcm_hw_params_get_buffer_size() failed: " << snd_strerror( -err );
+ alsaClose();
return false;
}
if ( ( err = snd_pcm_hw_params_get_period_size( hwparams, &alsa_period_size, 0 ) ) < 0 )
{
qDebug() << "snd_pcm_hw_params_get_period_size() failed: " << snd_strerror( -err );
+ alsaClose();
return false;
}
snd_pcm_sw_params_alloca( &swparams );
@@ -390,6 +422,7 @@
if ( snd_pcm_sw_params( alsa_pcm, swparams ) < 0 )
{
qDebug() << "Unable to install sw params" ;
+ alsaClose();
return false;
}
@@ -420,38 +453,32 @@
}
-bool
-AlsaAudio::alsaOpen( QString device, AFormat format, unsigned int rate, unsigned int channels, unsigned int buffer_time, unsigned int period_time )
+int
+AlsaAudio::startPlayback()
{
- Q_DEBUG_BLOCK << "Opening device:" << device;
+ int pthreadError = 0;
+
+ // We should double check this here. AlsaPlayback::initAudio
+ // isn't having its emitted error caught.
+ // So double check here to avoid a potential assert.
+ if ( !alsa_pcm )
+ return 1;
//HACK because we should zero pad only after we have started
m_zero_pad = false;
- inputf = snd_format_from_xmms( format, rate, channels );
-
- // We'll be using this in alsaWrite
- m_max_buffer_size = inputf->bps;
-
// And clear the buffer, just in case
clearBuffer();
- if (alsaSetup( device, buffer_time, period_time, inputf ) == false)
- {
- alsaClose();
- return false;
- }
-
going = true;
- convertb = xmms_convert_buffers_new();
AlsaAudio* aaThread = new AlsaAudio();
qDebug() << "Starting thread";
- pthread_create( &audio_thread, NULL, &alsa_loop, (void*)aaThread );
+ pthreadError = pthread_create( &audio_thread, NULL, &alsa_loop, (void*)aaThread );
- return true;
+ return pthreadError;
}
void AlsaAudio::clearBuffer( void )
@@ -503,24 +530,44 @@
void
-AlsaAudio::alsaClose()
+AlsaAudio::stopPlayback()
{
if (going)
{
Q_DEBUG_BLOCK;
-
+
going = false;
-
+
pthread_join( audio_thread, NULL );
-
- xmms_convert_buffers_destroy( convertb );
- convertb = NULL;
+ }
+}
+
+
+
+void
+AlsaAudio::alsaClose()
+{
+ Q_DEBUG_BLOCK;
+
+ alsa_close_pcm();
+
+ xmms_convert_buffers_destroy( convertb );
+ convertb = NULL;
+
+ if ( inputf )
+ {
free( inputf );
inputf = NULL;
+ }
+ if (outputf )
+ {
free( outputf );
outputf = NULL;
-
+ }
+ if ( logs )
+ {
snd_output_close( logs );
+ logs = NULL;
}
}
@@ -541,6 +588,7 @@
AlsaAudio::run()
{
int npfds = snd_pcm_poll_descriptors_count( alsa_pcm );
+ int err;
struct pollfd *pfds;
unsigned short *revents;
@@ -548,6 +596,10 @@
goto _error;
pfds = (struct pollfd*)malloc( sizeof( *pfds ) * npfds );
revents = (unsigned short*)malloc( sizeof( *revents ) * npfds );
+ err = snd_pcm_prepare( alsa_pcm );
+ if ( err < 0 )
+ qDebug() << "snd_pcm_prepare error:" << snd_strerror( err );
+
while ( going && alsa_pcm )
{
if (audioData.size() < hw_period_size_in)
@@ -598,7 +650,9 @@
free( revents );
_error:
- alsa_close_pcm();
+ err = snd_pcm_drop( alsa_pcm );
+ if ( err < 0 )
+ qDebug() << "snd_pcm_drop error:" << snd_strerror( err );
QMutexLocker locker( &mutex );
audioData.clear();
locker.unlock();
@@ -784,7 +838,7 @@
int err;
snd_pcm_drop( alsa_pcm );
if ( ( err = snd_pcm_close( alsa_pcm ) ) < 0 )
- qDebug() << "alsa_pcm_close() failed: " << snd_strerror( -err );
+ qDebug() << "alsa_close_pcm() failed: " << snd_strerror( -err );
alsa_pcm = NULL;
}
}
--- last.fm-1.3.0.62.orig/src/output/alsa-playback/alsaaudio.h
+++ last.fm-1.3.0.62/src/output/alsa-playback/alsaaudio.h
@@ -65,12 +65,17 @@
{
public:
AlsaAudio();
+ ~AlsaAudio();
int getCards();
AlsaDeviceInfo getDeviceInfo( int device );
- bool alsaOpen( QString device, AFormat format, uint rate, uint channels, uint buffer_time, uint period_time );
+ bool alsaOpen( QString device, AFormat format, unsigned int rate,
+ unsigned int channels, snd_pcm_uframes_t periodSize,
+ unsigned int periodCount );
+ int startPlayback();
void alsaWrite( const QByteArray* inputData );
+ void stopPlayback();
void alsaClose();
void setVolume ( float vol );
--- last.fm-1.3.0.62.orig/src/output/alsa-playback/alsaplayback.cpp
+++ last.fm-1.3.0.62/src/output/alsa-playback/alsaplayback.cpp
@@ -103,27 +103,17 @@
void
AlsaPlayback::startPlayback()
{
- int channels = 2;
- int sampleRate = 44100;
- int periodSize = 1024; // According to mplayer, these two are good defaults.
- int periodCount = 16; // They create a buffer size of 16384 frames.
- QString cardDevice;
-
if (!m_audio)
{
Q_DEBUG_BLOCK << "No AlsaAudio instance available.";
goto _error;
}
- cardDevice = internalSoundCardID();
-
-// We assume host byte order
-#ifdef WORDS_BIGENDIAN
- if (!m_audio->alsaOpen( cardDevice, FMT_S16_BE, sampleRate, channels, periodSize, periodCount ))
-#else
- if (!m_audio->alsaOpen( cardDevice, FMT_S16_LE, sampleRate, channels, periodSize, periodCount ))
-#endif
+ if ( m_audio->startPlayback() )
+ {
+ Q_DEBUG_BLOCK << "Error starting playback.";
goto _error;
+ }
return;
@@ -131,7 +121,7 @@
// We need to send a stop signal to m_iInput here, otherwise
// it will keep running and filling up the buffers even though
// there is no available device.
-
+
emit error( Radio_NoSoundcard, tr("The ALSA soundsystem is either busy or not present.") );
}
@@ -139,16 +129,41 @@
void
AlsaPlayback::stopPlayback()
{
- m_audio->alsaClose();
+ m_audio->stopPlayback();
}
void
AlsaPlayback::initAudio( long /*sampleRate*/, int /*channels*/ )
{
+ int channels = 2;
+ int sampleRate = 44100;
+ int periodSize = 1024; // According to mplayer, these two are good defaults.
+ int periodCount = 16; // They create a buffer size of 16384 frames.
+ QString cardDevice;
+
delete m_audio;
m_audio = new AlsaAudio;
m_audio->clearBuffer();
+
+ cardDevice = internalSoundCardID();
+
+ // We assume host byte order
+#ifdef WORDS_BIGENDIAN
+ if (!m_audio->alsaOpen( cardDevice, FMT_S16_BE, sampleRate, channels, periodSize, periodCount ))
+#else
+ if (!m_audio->alsaOpen( cardDevice, FMT_S16_LE, sampleRate, channels, periodSize, periodCount ))
+#endif
+ goto _error;
+
+ return;
+
+_error:
+ // We need to send a stop signal to m_iInput here, otherwise
+ // it will keep running and filling up the buffers even though
+ // there is no available device.
+
+ emit error( Radio_NoSoundcard, tr("The ALSA soundsystem is either busy or not present.") );
}
--- last.fm-1.3.0.62.orig/src/settingsdialog.cpp
+++ last.fm-1.3.0.62/src/settingsdialog.cpp
@@ -209,6 +209,8 @@
originalProxyPassword = The::settings().getProxyPassword();
originalProxyPort = The::settings().getProxyPort();
originalProxyUsage = The::settings().isUseProxy();
+ originalSoundCard = The::settings().soundCard();
+ originalSoundSystem = The::settings().soundSystem();
populateAccount();
populateRadio();
