From b335321950d441724f5aafb0bcc275df1547e574 Mon Sep 17 00:00:00 2001 From: Volker Fischer Date: Wed, 15 Apr 2020 15:29:43 +0200 Subject: [PATCH] rename SYSTEM_FRAME_SIZE_SAMPLES_SMALL to SYSTEM_FRAME_SIZE_SAMPLES --- src/channel.cpp | 4 +- src/client.cpp | 22 +++---- src/clientsettingsdlg.cpp | 12 ++-- src/global.h | 124 +++++++++++++++++++------------------- src/main.cpp | 2 +- src/server.cpp | 22 +++---- 6 files changed, 93 insertions(+), 93 deletions(-) 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 ( \ "
" \ "For more information use the ""What's " \ "This"" help (? menu, right mouse button or Shift+F1)" \ "
" ) -#define _MAXSHORT 32767 -#define _MAXBYTE 255 // binary: 11111111 -#define _MINSHORT ( -32768 ) +#define _MAXSHORT 32767 +#define _MAXBYTE 255 // binary: 11111111 +#define _MINSHORT ( -32768 ) #if HAVE_STDINT_H # include @@ -268,7 +268,7 @@ typedef unsigned char uint8_t; /* Pseudo enum definitions -------------------------------------------------- */ // definition for custom event -#define MS_PACKET_RECEIVED 0 +#define MS_PACKET_RECEIVED 0 /* Classes ********************************************************************/ diff --git a/src/main.cpp b/src/main.cpp index 90e137c8..d90355a5 100755 --- a/src/main.cpp +++ b/src/main.cpp @@ -118,7 +118,7 @@ int main ( int argc, char** argv ) "--fastupdate" ) ) { bUseDoubleSystemFrameSize = false; // 64 samples frame size - tsConsole << "- using " << SYSTEM_FRAME_SIZE_SAMPLES_SMALL << " samples frame size mode" << endl; + tsConsole << "- using " << SYSTEM_FRAME_SIZE_SAMPLES << " samples frame size mode" << endl; continue; } diff --git a/src/server.cpp b/src/server.cpp index 16acb80a..91f97809 100755 --- a/src/server.cpp +++ b/src/server.cpp @@ -32,7 +32,7 @@ CHighPrecisionTimer::CHighPrecisionTimer ( const bool bNewUseDoubleSystemFrameSi { // add some error checking, the high precision timer implementation only // supports 64 and 128 samples frame size at 48 kHz sampling rate -#if ( SYSTEM_FRAME_SIZE_SAMPLES_SMALL != 64 ) && ( DOUBLE_SYSTEM_FRAME_SIZE_SAMPLES != 128 ) +#if ( SYSTEM_FRAME_SIZE_SAMPLES != 64 ) && ( DOUBLE_SYSTEM_FRAME_SIZE_SAMPLES != 128 ) # error "Only system frame size of 64 and 128 samples is supported by this module" #endif #if ( SYSTEM_SAMPLE_RATE_HZ != 48000 ) @@ -127,7 +127,7 @@ CHighPrecisionTimer::CHighPrecisionTimer ( const bool bUseDoubleSystemFrameSize } else { - iNsDelay = ( (uint64_t) SYSTEM_FRAME_SIZE_SAMPLES_SMALL * 1000000000 ) / + iNsDelay = ( (uint64_t) SYSTEM_FRAME_SIZE_SAMPLES * 1000000000 ) / (uint64_t) SYSTEM_SAMPLE_RATE_HZ; // in ns } @@ -267,7 +267,7 @@ CServer::CServer ( const int iNewMaxNumChan, &iOpusError ); Opus64Mode[i] = opus_custom_mode_create ( SYSTEM_SAMPLE_RATE_HZ, - SYSTEM_FRAME_SIZE_SAMPLES_SMALL, + SYSTEM_FRAME_SIZE_SAMPLES, &iOpusError ); // init audio encoders and decoders @@ -320,7 +320,7 @@ CServer::CServer ( const int iNewMaxNumChan, } else { - iServerFrameSizeSamples = SYSTEM_FRAME_SIZE_SAMPLES_SMALL; + iServerFrameSizeSamples = SYSTEM_FRAME_SIZE_SAMPLES; } @@ -953,7 +953,7 @@ JitterMeas.Measure(); } else if ( vecAudioComprType[i] == CT_OPUS64 ) { - iClientFrameSizeSamples = SYSTEM_FRAME_SIZE_SAMPLES_SMALL; + iClientFrameSizeSamples = SYSTEM_FRAME_SIZE_SAMPLES; if ( vecNumAudioChannels[i] == 1 ) { @@ -988,7 +988,7 @@ JitterMeas.Measure(); // is false and the Get() function is not called at all. Therefore if the buffer is not needed // we do not spend any time in the function but go directly inside the if condition. if ( ( vecUseDoubleSysFraSizeConvBuf[i] == 0 ) || - !DoubleFrameSizeConvBufIn[iCurChanID].Get ( vecvecsData[i], SYSTEM_FRAME_SIZE_SAMPLES_SMALL * vecNumAudioChannels[i] ) ) + !DoubleFrameSizeConvBufIn[iCurChanID].Get ( vecvecsData[i], SYSTEM_FRAME_SIZE_SAMPLES * vecNumAudioChannels[i] ) ) { // get current number of OPUS coded bytes const int iCeltNumCodedBytes = vecChannels[iCurChanID].GetNetwFrameSize(); @@ -1028,7 +1028,7 @@ JitterMeas.Measure(); opus_custom_decode ( CurOpusDecoder, pCurCodedData, iCeltNumCodedBytes, - &vecvecsData[i][iB * SYSTEM_FRAME_SIZE_SAMPLES_SMALL * vecNumAudioChannels[i]], + &vecvecsData[i][iB * SYSTEM_FRAME_SIZE_SAMPLES * vecNumAudioChannels[i]], iClientFrameSizeSamples ); } } @@ -1038,7 +1038,7 @@ JitterMeas.Measure(); if ( vecUseDoubleSysFraSizeConvBuf[i] != 0 ) { DoubleFrameSizeConvBufIn[iCurChanID].PutAll ( vecvecsData[i] ); - DoubleFrameSizeConvBufIn[iCurChanID].Get ( vecvecsData[i], SYSTEM_FRAME_SIZE_SAMPLES_SMALL * vecNumAudioChannels[i] ); + DoubleFrameSizeConvBufIn[iCurChanID].Get ( vecvecsData[i], SYSTEM_FRAME_SIZE_SAMPLES * vecNumAudioChannels[i] ); } } } @@ -1131,7 +1131,7 @@ JitterMeas.Measure(); } else if ( vecAudioComprType[i] == CT_OPUS64 ) { - iClientFrameSizeSamples = SYSTEM_FRAME_SIZE_SAMPLES_SMALL; + iClientFrameSizeSamples = SYSTEM_FRAME_SIZE_SAMPLES; if ( vecNumAudioChannels[i] == 1 ) { @@ -1153,7 +1153,7 @@ JitterMeas.Measure(); // is false and the Get() function is not called at all. Therefore if the buffer is not needed // we do not spend any time in the function but go directly inside the if condition. if ( ( vecUseDoubleSysFraSizeConvBuf[i] == 0 ) || - DoubleFrameSizeConvBufOut[iCurChanID].Put ( vecsSendData, SYSTEM_FRAME_SIZE_SAMPLES_SMALL * vecNumAudioChannels[i] ) ) + DoubleFrameSizeConvBufOut[iCurChanID].Put ( vecsSendData, SYSTEM_FRAME_SIZE_SAMPLES * vecNumAudioChannels[i] ) ) { if ( vecUseDoubleSysFraSizeConvBuf[i] != 0 ) { @@ -1173,7 +1173,7 @@ opus_custom_encoder_ctl ( CurOpusEncoder, OPUS_SET_BITRATE ( CalcBitRateBitsPerSecFromCodedBytes ( iCeltNumCodedBytes, iClientFrameSizeSamples ) ) ); opus_custom_encode ( CurOpusEncoder, - &vecsSendData[iB * SYSTEM_FRAME_SIZE_SAMPLES_SMALL * vecNumAudioChannels[i]], + &vecsSendData[iB * SYSTEM_FRAME_SIZE_SAMPLES * vecNumAudioChannels[i]], iClientFrameSizeSamples, &vecbyCodedData[0], iCeltNumCodedBytes );