diff --git a/src/channel.cpp b/src/channel.cpp
index 11c35271..a9778c76 100755
--- a/src/channel.cpp
+++ b/src/channel.cpp
@@ -166,7 +166,7 @@ void CChannel::SetAudioStreamProperties ( const EAudComprType eNewAudComprType,
}
else
{
- iAudioFrameSizeSamples = SYSTEM_FRAME_SIZE_SAMPLES_SMALL;
+ iAudioFrameSizeSamples = SYSTEM_FRAME_SIZE_SAMPLES;
}
MutexSocketBuf.lock();
@@ -387,7 +387,7 @@ void CChannel::OnNetTranspPropsReceived ( CNetworkTransportProps NetworkTranspor
else
{
iFadeInCntMax = FADE_IN_NUM_FRAMES / iNetwFrameSizeFact;
- iAudioFrameSizeSamples = SYSTEM_FRAME_SIZE_SAMPLES_SMALL;
+ iAudioFrameSizeSamples = SYSTEM_FRAME_SIZE_SAMPLES;
}
MutexSocketBuf.lock();
diff --git a/src/client.cpp b/src/client.cpp
index 8e452374..2ae65647 100755
--- a/src/client.cpp
+++ b/src/client.cpp
@@ -83,7 +83,7 @@ CClient::CClient ( const quint16 iPortNumber,
&iOpusError );
Opus64Mode = opus_custom_mode_create ( SYSTEM_SAMPLE_RATE_HZ,
- SYSTEM_FRAME_SIZE_SAMPLES_SMALL,
+ SYSTEM_FRAME_SIZE_SAMPLES,
&iOpusError );
// init audio encoders and decoders
@@ -669,16 +669,16 @@ void CClient::Stop()
void CClient::Init()
{
// check if possible frame size factors are supported
- const int iFraSizePreffered = SYSTEM_FRAME_SIZE_SAMPLES_SMALL * FRAME_SIZE_FACTOR_PREFERRED;
- const int iFraSizeDefault = SYSTEM_FRAME_SIZE_SAMPLES_SMALL * FRAME_SIZE_FACTOR_DEFAULT;
- const int iFraSizeSafe = SYSTEM_FRAME_SIZE_SAMPLES_SMALL * FRAME_SIZE_FACTOR_SAFE;
+ const int iFraSizePreffered = SYSTEM_FRAME_SIZE_SAMPLES * FRAME_SIZE_FACTOR_PREFERRED;
+ const int iFraSizeDefault = SYSTEM_FRAME_SIZE_SAMPLES * FRAME_SIZE_FACTOR_DEFAULT;
+ const int iFraSizeSafe = SYSTEM_FRAME_SIZE_SAMPLES * FRAME_SIZE_FACTOR_SAFE;
bFraSiFactPrefSupported = ( Sound.Init ( iFraSizePreffered ) == iFraSizePreffered );
bFraSiFactDefSupported = ( Sound.Init ( iFraSizeDefault ) == iFraSizeDefault );
bFraSiFactSafeSupported = ( Sound.Init ( iFraSizeSafe ) == iFraSizeSafe );
// translate block size index in actual block size
- const int iPrefMonoFrameSize = iSndCrdPrefFrameSizeFactor * SYSTEM_FRAME_SIZE_SAMPLES_SMALL;
+ const int iPrefMonoFrameSize = iSndCrdPrefFrameSizeFactor * SYSTEM_FRAME_SIZE_SAMPLES;
// get actual sound card buffer size using preferred size
iMonoBlockSizeSam = Sound.Init ( iPrefMonoFrameSize );
@@ -686,12 +686,12 @@ void CClient::Init()
// Calculate the current sound card frame size factor. In case
// the current mono block size is not a multiple of the system
// frame size, we have to use a sound card conversion buffer.
- if ( ( iMonoBlockSizeSam == ( SYSTEM_FRAME_SIZE_SAMPLES_SMALL * FRAME_SIZE_FACTOR_PREFERRED ) ) ||
- ( iMonoBlockSizeSam == ( SYSTEM_FRAME_SIZE_SAMPLES_SMALL * FRAME_SIZE_FACTOR_DEFAULT ) ) ||
- ( iMonoBlockSizeSam == ( SYSTEM_FRAME_SIZE_SAMPLES_SMALL * FRAME_SIZE_FACTOR_SAFE ) ) )
+ if ( ( iMonoBlockSizeSam == ( SYSTEM_FRAME_SIZE_SAMPLES * FRAME_SIZE_FACTOR_PREFERRED ) ) ||
+ ( iMonoBlockSizeSam == ( SYSTEM_FRAME_SIZE_SAMPLES * FRAME_SIZE_FACTOR_DEFAULT ) ) ||
+ ( iMonoBlockSizeSam == ( SYSTEM_FRAME_SIZE_SAMPLES * FRAME_SIZE_FACTOR_SAFE ) ) )
{
// regular case: one of our predefined buffer sizes is available
- iSndCrdFrameSizeFactor = iMonoBlockSizeSam / SYSTEM_FRAME_SIZE_SAMPLES_SMALL;
+ iSndCrdFrameSizeFactor = iMonoBlockSizeSam / SYSTEM_FRAME_SIZE_SAMPLES;
// no sound card conversion buffer required
bSndCrdConversionBufferRequired = false;
@@ -715,7 +715,7 @@ void CClient::Init()
{
if ( iSndCardMonoBlockSizeSamConvBuff < DOUBLE_SYSTEM_FRAME_SIZE_SAMPLES )
{
- iMonoBlockSizeSam = SYSTEM_FRAME_SIZE_SAMPLES_SMALL;
+ iMonoBlockSizeSam = SYSTEM_FRAME_SIZE_SAMPLES;
eAudioCompressionType = CT_OPUS64;
}
else
@@ -773,7 +773,7 @@ void CClient::Init()
}
else /* CT_OPUS64 */
{
- iOPUSFrameSizeSamples = SYSTEM_FRAME_SIZE_SAMPLES_SMALL;
+ iOPUSFrameSizeSamples = SYSTEM_FRAME_SIZE_SAMPLES;
if ( eAudioChannelConf == CC_MONO )
{
diff --git a/src/clientsettingsdlg.cpp b/src/clientsettingsdlg.cpp
index 273eb045..c2c4a972 100755
--- a/src/clientsettingsdlg.cpp
+++ b/src/clientsettingsdlg.cpp
@@ -347,14 +347,14 @@ CClientSettingsDlg::CClientSettingsDlg ( CClient* pNCliP, QWidget* parent,
// set text for sound card buffer delay radio buttons
rbtBufferDelayPreferred->setText ( GenSndCrdBufferDelayString (
- FRAME_SIZE_FACTOR_PREFERRED * SYSTEM_FRAME_SIZE_SAMPLES_SMALL,
+ FRAME_SIZE_FACTOR_PREFERRED * SYSTEM_FRAME_SIZE_SAMPLES,
", preferred" ) );
rbtBufferDelayDefault->setText ( GenSndCrdBufferDelayString (
- FRAME_SIZE_FACTOR_DEFAULT * SYSTEM_FRAME_SIZE_SAMPLES_SMALL ) );
+ FRAME_SIZE_FACTOR_DEFAULT * SYSTEM_FRAME_SIZE_SAMPLES ) );
rbtBufferDelaySafe->setText ( GenSndCrdBufferDelayString (
- FRAME_SIZE_FACTOR_SAFE * SYSTEM_FRAME_SIZE_SAMPLES_SMALL ) );
+ FRAME_SIZE_FACTOR_SAFE * SYSTEM_FRAME_SIZE_SAMPLES ) );
// sound card buffer delay inits
SndCrdBufferDelayButtonGroup.addButton ( rbtBufferDelayPreferred );
@@ -472,9 +472,9 @@ void CClientSettingsDlg::UpdateSoundCardFrame()
const int iCurActualBufSize = pClient->GetSndCrdActualMonoBlSize();
// check which predefined size is used (it is possible that none is used)
- const bool bPreferredChecked = ( iCurActualBufSize == SYSTEM_FRAME_SIZE_SAMPLES_SMALL * FRAME_SIZE_FACTOR_PREFERRED );
- const bool bDefaultChecked = ( iCurActualBufSize == SYSTEM_FRAME_SIZE_SAMPLES_SMALL * FRAME_SIZE_FACTOR_DEFAULT );
- const bool bSafeChecked = ( iCurActualBufSize == SYSTEM_FRAME_SIZE_SAMPLES_SMALL * FRAME_SIZE_FACTOR_SAFE );
+ const bool bPreferredChecked = ( iCurActualBufSize == SYSTEM_FRAME_SIZE_SAMPLES * FRAME_SIZE_FACTOR_PREFERRED );
+ const bool bDefaultChecked = ( iCurActualBufSize == SYSTEM_FRAME_SIZE_SAMPLES * FRAME_SIZE_FACTOR_DEFAULT );
+ const bool bSafeChecked = ( iCurActualBufSize == SYSTEM_FRAME_SIZE_SAMPLES * FRAME_SIZE_FACTOR_SAFE );
// Set radio buttons according to current value (To make it possible
// to have all radio buttons unchecked, we have to disable the
diff --git a/src/global.h b/src/global.h
index 45bc4781..cb49542b 100755
--- a/src/global.h
+++ b/src/global.h
@@ -79,169 +79,169 @@ LED bar: lbr
// version and application name (use version from qt prject file)
#undef VERSION
-#define VERSION APP_VERSION
-#define APP_NAME "Jamulus"
+#define VERSION APP_VERSION
+#define APP_NAME "Jamulus"
// Windows registry key name of auto run entry for the server
-#define AUTORUN_SERVER_REG_NAME "Jamulus server"
+#define AUTORUN_SERVER_REG_NAME "Jamulus server"
// default names of the ini-file for client and server
-#define DEFAULT_INI_FILE_NAME "Jamulus.ini"
-#define DEFAULT_INI_FILE_NAME_SERVER "Jamulusserver.ini"
+#define DEFAULT_INI_FILE_NAME "Jamulus.ini"
+#define DEFAULT_INI_FILE_NAME_SERVER "Jamulusserver.ini"
// file name for logging file
-#define DEFAULT_LOG_FILE_NAME "Jamulussrvlog.txt"
+#define DEFAULT_LOG_FILE_NAME "Jamulussrvlog.txt"
// default oldest item to draw in history graph (days ago)
-#define DEFAULT_DAYS_HISTORY 60
+#define DEFAULT_DAYS_HISTORY 60
// System block size, this is the block size on which the audio coder works.
// All other block sizes must be a multiple of this size.
// Note that the UpdateAutoSetting() function assumes a value of 128.
-#define SYSTEM_FRAME_SIZE_SAMPLES_SMALL 64 // TODO this is temporary and shall be replaced by SYSTEM_FRAME_SIZE_SAMPLES later on
-#define DOUBLE_SYSTEM_FRAME_SIZE_SAMPLES ( 2 * SYSTEM_FRAME_SIZE_SAMPLES_SMALL )
+#define SYSTEM_FRAME_SIZE_SAMPLES 64
+#define DOUBLE_SYSTEM_FRAME_SIZE_SAMPLES ( 2 * SYSTEM_FRAME_SIZE_SAMPLES )
// default server address
-#define DEFAULT_SERVER_ADDRESS "jamulus.fischvolk.de"
-#define DEFAULT_SERVER_NAME "Central Server"
+#define DEFAULT_SERVER_ADDRESS "jamulus.fischvolk.de"
+#define DEFAULT_SERVER_NAME "Central Server"
// download URL
-#define SOFTWARE_DOWNLOAD_URL "http://sourceforge.net/projects/llcon/files"
+#define SOFTWARE_DOWNLOAD_URL "http://sourceforge.net/projects/llcon/files"
// determining server internal address uses well-known host and port
// (Google DNS, or something else reliable)
-#define WELL_KNOWN_HOST "8.8.8.8" // Google
-#define WELL_KNOWN_PORT 53 // DNS
-#define IP_LOOKUP_TIMEOUT 500 // ms
+#define WELL_KNOWN_HOST "8.8.8.8" // Google
+#define WELL_KNOWN_PORT 53 // DNS
+#define IP_LOOKUP_TIMEOUT 500 // ms
// defined port numbers for client and server
-#define LLCON_DEFAULT_PORT_NUMBER 22124
-#define LLCON_PORT_NUMBER_NORTHAMERICA 22224
+#define LLCON_DEFAULT_PORT_NUMBER 22124
+#define LLCON_PORT_NUMBER_NORTHAMERICA 22224
// system sample rate (the sound card and audio coder works on this sample rate)
-#define SYSTEM_SAMPLE_RATE_HZ 48000 // Hz
+#define SYSTEM_SAMPLE_RATE_HZ 48000 // Hz
// define the allowed audio frame size factors (since the
// "SYSTEM_FRAME_SIZE_SAMPLES" is quite small, it may be that on some
// computers a larger value is required)
-#define FRAME_SIZE_FACTOR_PREFERRED 1 // 64 samples accumulated frame size
-#define FRAME_SIZE_FACTOR_DEFAULT 2 // 128 samples accumulated frame size
-#define FRAME_SIZE_FACTOR_SAFE 4 // 256 samples accumulated frame size
+#define FRAME_SIZE_FACTOR_PREFERRED 1 // 64 samples accumulated frame size
+#define FRAME_SIZE_FACTOR_DEFAULT 2 // 128 samples accumulated frame size
+#define FRAME_SIZE_FACTOR_SAFE 4 // 256 samples accumulated frame size
// define the minimum allowed number of coded bytes for CELT (the encoder
// gets in trouble if the value is too low)
-#define CELT_MINIMUM_NUM_BYTES 10
+#define CELT_MINIMUM_NUM_BYTES 10
// Maximum block size for network input buffer. It is defined by the longest
// protocol message which is PROTMESSID_CLM_SERVER_LIST: Worst case:
// (2+2+1+2+2)+200*(4+2+2+1+1+2+20+2+32+2+20)=17609
// We add some headroom to that value.
-#define MAX_SIZE_BYTES_NETW_BUF 20000
+#define MAX_SIZE_BYTES_NETW_BUF 20000
// minimum/maximum network buffer size (which can be chosen by slider)
-#define MIN_NET_BUF_SIZE_NUM_BL 1 // number of blocks
-#define MAX_NET_BUF_SIZE_NUM_BL 20 // number of blocks
-#define AUTO_NET_BUF_SIZE_FOR_PROTOCOL ( MAX_NET_BUF_SIZE_NUM_BL + 1 ) // auto set parameter (only used for protocol)
+#define MIN_NET_BUF_SIZE_NUM_BL 1 // number of blocks
+#define MAX_NET_BUF_SIZE_NUM_BL 20 // number of blocks
+#define AUTO_NET_BUF_SIZE_FOR_PROTOCOL ( MAX_NET_BUF_SIZE_NUM_BL + 1 ) // auto set parameter (only used for protocol)
// default network buffer size
-#define DEF_NET_BUF_SIZE_NUM_BL 10 // number of blocks
+#define DEF_NET_BUF_SIZE_NUM_BL 10 // number of blocks
// audio mixer fader maximum value
-#define AUD_MIX_FADER_MAX 100
+#define AUD_MIX_FADER_MAX 100
// maximum number of recognized sound cards installed in the system,
// definition for "no device"
-#define MAX_NUMBER_SOUND_CARDS 129 // e.g. 16 inputs, 8 outputs + default entry (MacOS)
-#define INVALID_SNC_CARD_DEVICE -1
+#define MAX_NUMBER_SOUND_CARDS 129 // e.g. 16 inputs, 8 outputs + default entry (MacOS)
+#define INVALID_SNC_CARD_DEVICE -1
// define the maximum number of audio channel for input/output we can store
// channel infos for (and therefore this is the maximum number of entries in
// the channel selection combo box regardless of the actual available number
// of channels by the audio device)
-#define MAX_NUM_IN_OUT_CHANNELS 64
+#define MAX_NUM_IN_OUT_CHANNELS 64
// maximum number of elemts in the server address combo box
-#define MAX_NUM_SERVER_ADDR_ITEMS 6
+#define MAX_NUM_SERVER_ADDR_ITEMS 6
// maximum number of fader settings to be stored (together with the fader tags)
-#define MAX_NUM_STORED_FADER_SETTINGS 200
+#define MAX_NUM_STORED_FADER_SETTINGS 200
// defines for LED level meter CMultiColorLEDBar
-#define NUM_STEPS_LED_BAR 8
-#define RED_BOUND_LED_BAR 7
-#define YELLOW_BOUND_LED_BAR 5
+#define NUM_STEPS_LED_BAR 8
+#define RED_BOUND_LED_BAR 7
+#define YELLOW_BOUND_LED_BAR 5
// range for signal level meter
-#define LOW_BOUND_SIG_METER ( -50.0 ) // dB
-#define UPPER_BOUND_SIG_METER ( 0.0 ) // dB
+#define LOW_BOUND_SIG_METER ( -50.0 ) // dB
+#define UPPER_BOUND_SIG_METER ( 0.0 ) // dB
// Maximum number of connected clients at the server. If you want to change this
// paramter you have to modify the code on some places, too! The code tag
// "MAX_NUM_CHANNELS_TAG" shows these places (just search for the tag in the entire code)
-#define MAX_NUM_CHANNELS 50 // max number channels for server
+#define MAX_NUM_CHANNELS 50 // max number channels for server
// actual number of used channels in the server
// this parameter can safely be changed from 1 to MAX_NUM_CHANNELS
// without any other changes in the code
-#define DEFAULT_USED_NUM_CHANNELS 10 // default used number channels for server
+#define DEFAULT_USED_NUM_CHANNELS 10 // default used number channels for server
// Maximum number of servers registered in the server list. If you want to
// change this parameter, you most probably have to adjust MAX_SIZE_BYTES_NETW_BUF.
-#define MAX_NUM_SERVERS_IN_SERVER_LIST 200
+#define MAX_NUM_SERVERS_IN_SERVER_LIST 200
// defines the time interval at which the ping time is updated in the GUI
-#define PING_UPDATE_TIME_MS 500 // ms
+#define PING_UPDATE_TIME_MS 500 // ms
// defines the time interval at which the ping time is updated for the server
// list
-#define PING_UPDATE_TIME_SERVER_LIST_MS 2000 // ms
+#define PING_UPDATE_TIME_SERVER_LIST_MS 2000 // ms
// defines the interval between Channel Level updates from the server
-#define CHANNEL_LEVEL_UPDATE_INTERVAL 200 // number of frames at 64 samples frame size
+#define CHANNEL_LEVEL_UPDATE_INTERVAL 200 // number of frames at 64 samples frame size
// time-out until a registered server is deleted from the server list if no
// new registering was made in minutes
-#define SERVLIST_TIME_OUT_MINUTES 60 // minutes
+#define SERVLIST_TIME_OUT_MINUTES 60 // minutes
// poll time for server list (to check if entries are time-out)
-#define SERVLIST_POLL_TIME_MINUTES 1 // minute
+#define SERVLIST_POLL_TIME_MINUTES 1 // minute
// time interval for sending ping messages to servers in the server list
-#define SERVLIST_UPDATE_PING_SERVERS_MS 59000 // ms
+#define SERVLIST_UPDATE_PING_SERVERS_MS 59000 // ms
// time until a slave server registers in the server list
-#define SERVLIST_REGIST_INTERV_MINUTES 15 // minutes
+#define SERVLIST_REGIST_INTERV_MINUTES 15 // minutes
// defines the minimum time a server must run to be a permanent server
-#define SERVLIST_TIME_PERMSERV_MINUTES 1440 // minutes, 1440 = 60 min * 24 h
+#define SERVLIST_TIME_PERMSERV_MINUTES 1440 // minutes, 1440 = 60 min * 24 h
// length of the moving average buffer for response time measurement
-#define TIME_MOV_AV_RESPONSE_SECONDS 30 // seconds
+#define TIME_MOV_AV_RESPONSE_SECONDS 30 // seconds
// Maximum length of fader tag and text message strings (Since for chat messages
// some HTML code is added, we also have to define a second length which includes
// this additionl HTML code. Right now the length of the HTML code is approx. 66
// character. Here, we add some headroom to this number)
-#define MAX_LEN_FADER_TAG 16
-#define MAX_LEN_CHAT_TEXT 1600
-#define MAX_LEN_CHAT_TEXT_PLUS_HTML 1800
-#define MAX_LEN_SERVER_NAME 20
-#define MAX_LEN_IP_ADDRESS 15
-#define MAX_LEN_SERVER_CITY 20
-#define MAX_LEN_VERSION_TEXT 20
+#define MAX_LEN_FADER_TAG 16
+#define MAX_LEN_CHAT_TEXT 1600
+#define MAX_LEN_CHAT_TEXT_PLUS_HTML 1800
+#define MAX_LEN_SERVER_NAME 20
+#define MAX_LEN_IP_ADDRESS 15
+#define MAX_LEN_SERVER_CITY 20
+#define MAX_LEN_VERSION_TEXT 20
// common tool tip bottom line text
-#define TOOLTIP_COM_END_TEXT tr ( \
+#define TOOLTIP_COM_END_TEXT tr ( \
"