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rename SYSTEM_FRAME_SIZE_SAMPLES_SMALL to SYSTEM_FRAME_SIZE_SAMPLES

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This commit is contained in:
Volker Fischer 2020-04-15 15:29:43 +02:00
parent f51dd662c4
commit b335321950
6 changed files with 93 additions and 93 deletions
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4
src/channel.cpp
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@ -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();

22
src/client.cpp
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@ -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 )
{

12
src/clientsettingsdlg.cpp
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@ -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

124
src/global.h
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@ -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 ( \
"<br><div align=right><font size=-1><i>" \
"For more information use the ""What's " \
"This"" help (? menu, right mouse button or Shift+F1)" \
"</i></font></div>" )
#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 <stdint.h>
@ -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 ********************************************************************/

2
src/main.cpp
View File

@ -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;
}

22
src/server.cpp
View File

@ -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 );