%2d: %7s, %s, %s, width: %5u, height: %5u, start addr: 0x%08x, addr offset: 0x%08x
%2d: %7s, %s, start addr: 0x%08x, end addr: 0x%08x, pitch: %6u, compressed tag start: 0x%04x, compressed tag end: 0x%04x, Compression mode: %s
Error param 0: %I32u (0x%08I32x)
Error param 1: %I32u (0x%08I32x)
Graphics error #: %I32u (0x%08I32x)
Graphics error description: %s
Graphics error state data.
Interrupt error status: %s
Not yet reported by this app: FIFO error state data.
Not yet reported by this app: IOIF error state data.
Last ten Screen IDs [start from oldest]:
#<
Invalid error code>
%B: internal error in ihex_read_section
%B:%u: bad extended linear start address length in Intel Hex file
%B:%u: bad extended start address length in Intel Hex file
<
WARNING>
Actionscript error message: %s
<
b>
Job Monitor Error Info<
/b>
<
b>
Memory Framework Error Info<
/b>
(nStart + nNumToAdd) <
= pArray->
mnLength
- unable to initialize heap
- unable to open console device
----- InitializeDataGate Failed: Unable to load .db file -----
----- InitializeDataGate Failed: Unable to load .xml file -----
-----OTPGameSetupMessage FromString start ----
-----OTPGameSetupMessage Pack start ----
-----OTPGameSetupMessage ToString start ----
-----OTPGameSetupMessage Unpack start ----
-----SubOTPGameSetupMessage FromString start ----
-----SubOTPGameSetupMessage Pack start ----
-----SubOTPGameSetupMessage ToString start ----
-----SubOTPGameSetupMessage Unpack start ----
.....highlightmanager start add frame block time %f
@ERROR : Referenced positions are not available.
A PS3 game that wishes to support PL2 is faced with one of two choices on how to know when to enable it. The game code could have a front end menu option to allow the user to select between stereo and PL2. This is however very ugly to the end user as they now have to configure audio output in more than one place. The option is also confusing as it doesn't force stereo or PL2 mode if the OSD is set to a 5.1 or 7.1 output mode. For these reasons a menu option is not recommended. The better alternative is to decide up front if your game will always output either plain stereo or PL2. This would probably be the audio director's choice based on what they prefer.
A SSL error occured
A XML error occured
A connection error occured
A security error of unknown cause has been detected which has
A tiled area was accessed as a color buffer. Specifically, it means that a Tiled area and linear area were accessed during drawing or, in other words, the color buffer area that was specified in cellGcmSetSurface() may not have been set properly for the Tiled area. Other cases for which this error can occur are when the pitch sizes of both area are not set properly or if the pitch size specified in cellGcmSetSurface() is larger than the pitch size specified by cellGcmSetTileInfo().
ATTRIB ACCESS ERROR - Getter instancename invalid %s
A[CoreReplayStorage] Decode(), Error unable to decode frame, t = %f, a.time = %f, b.time = %f
ActiveRenderableStore Received FifaWorldLoadStart Msg
Adding gaps of silence between samples of a given length in time is now made simple via the new EVENT_DELAY event. On the older players, users were required to first calculate the end time of the previous request (based on its start time, current position, length, and the current mixer time). This was then used to choose an appropriate start time for the following request. The delay event makes this process simpler by allowing users to queue a delay, which is effectively a silent request.
Among the errors that can occur when accessing a Tiled area, this error is generated when the area is accessed as a depth buffer. The depth buffer area may have straddled a Tiled area and linear area or, in other words, the depth buffer area specified in cellGcmSetSurface() may not have been set properly for the Tiled area. Also, the pitch sizes of both area may not have been set properly or the pitch size specified in cellGcmSetSurface() may be larger than the pitch size specified by cellGcmSetTileInfo().
An IOIF error occurred. This error occurs when the RSX attempts to access main memory, but the access cannot be performed normally since the address (more precisely, the offset) has not been mapped. Be sure to confirm that a main memory area was specified in cellGcmInit(), that the area that was allocated by cellGcmMapMainMemory() has been passed to libgcm, and that the argument is an offset value that was converted from an address by cellGcmAddressToOffset().
An attempt was made to use a function that is not supported by the hardware. The RSX has various restrictions that appear in the documentation, and this error corresponds to those restrictions.
An error occurred attempting to load an asset
An internal error occured
An unknown error type occurred.
An unspecified error has occured.
An unsupported command was executed. Like graphics error 1, this error is generated when an unsupported command is passed to the RSX. While graphics error 1 is generated when an Invalid command is encountered in the command fetch engine, this error occurs when an Invalid command is encountered in the rendering pipeline. Since an undefined command is fetched and only the unit in which the error occurs differs, the command buffer may also have been corrupted or overwritten in this case.
Buffer error in compressed datastream in %s chunk
CFileAsBuffer::LoadFile - CreateFileMapping failed! - %s (Error %d)
CFileAsBuffer::LoadFile - MapViewOfFile failed! - %s (Error %d)
ClipPlaneEnable changed by Effect! (not allowed)
Collation error - not relevant
Connect failed with error %d
DLC: Start unloading dll [%s].
Data error in compressed datastream in %s chunk
Define Restart Interval %u
Defines the channel in the destination submix to which the routing happen. Note that channel indexes start at 0.
Defines the initial gain value that the fader is going to start at.
Defines the maximum delay length to be allocated for a Pan3D instance. The delay line is used for simulating the delay effect of the direct sound path from the sound source, the variously delayed indirect paths (reflections, diffractions)that occur due to geometry, and the Doppler effect. A larger value normally allows the Pan3D instance to accommodate more indirect paths. If the size of the room that the Pan3D instance is simulating is large, CONSTRUCTORPARAM_MAXDELAY will also need to be large in order to enable hearing of at least all the first order reflections. A general usage guideline is to set CONSTRUCTORPARAM_MAXDELAY to double the size of room that Pan3D is simulating (converted to seconds based on the speed of sound), within the constraints of available memory.
DitherEnable changed by Effect! (not allowed)
ERROR THRESHOLD:
ERROR non of the above
ERROR: SCRAPE Compile error line %d: %s
ERROR: SCRAPE Error converting MSAA type from constant %s
ERROR: SCRAPE Error converting format from constant %s
ERROR: SCRAPE Error duplicate label name
ERROR: SCRAPE Error duplicate variable name %s
ERROR: SCRAPE Error resolving constant %s
ERROR: SCRAPE Error tokenizing line
ERROR: XML parse error at line %d
Enable Board Debug Logs?
Enable Events
Enable Passives
Enable SaveContextQueryDetails in IdleController
Enable attitude override
Enable local players
Enable or disable interrupts. Privileged instruction.
Enable target override
Enable team comm
Error Code: %d
Error Rate On Iteration %d is: %lf
Error copying snapshot to '%s'
Error decoding compressed text
Error during receiving HTTP response
Error during sending HTTP request
Error in error handling
Error in user chunk
Error loading blob.
Error loading module '%s' from file '%s':
Error loading recorded replay data
Error text not found (please report)
Error, command [%s] 0 will disable the sort overload, [%s] 1 ->
... will enable it
Error, command [%s] 1 will enable GameplayProp debugging, [%s] 0 will disable it
Error, command [%s] 1 will enable bindpose on all players, [%s] 0 will disable it
Error, command [%s] 1 will enable face bindpose on all players, [%s] 0 will disable it
Error, command [%s] 1 will enable lightmap rendering, [%s] 0 will disable it
Error, command [%s] 1 will enable render stats, [%s] 0 will disable it
Error, command [%s] 1 will enable wireframe, [%s] 0 will disable it
Error: AddAllocator - error evaluating block '%s'
Error: AddAllocatorContainer - error evaluating block '%s'
Error: AddCategory - error evaluating block '%s'
Error: AddCategoryAllocator - error evaluating block '%s'
Error: SetAutoValidate - error evaluating block '%s'
Error: SetLogging - error evaluating block '%s'
Events.General.MatchStart #OSDK3
Events.Kickoff.Start #OSDK3
Events.Tutorial.DrillRestart #OSDK3
Failed to bind socket, error %d.
Failed to create socket, error %d.
Failed to listen on socket, error %d
Failed to set socket to nonblocking, error %d
Fatal (internal) error in %s, line %d: %s
Fatal error while creating .fixup
FogEnable changed by Effect! (not allowed)
FogStart changed by Effect! (not allowed)
GetFramesFromReplayStoragesOnly, Error unnable to Decode frame
Give me an error string
Grad start colour X
Grad start colour Y
Grad start colour Z
HW Replay FrameReset start
HW Replay FrameReset, invalidate frame start
I@Omterma
Error Driver NOT loaded yet. Contact Henrik
If [scopeBlock] [scopeParm] [optional: comparison value]- start of conditional execution block. Needs matching endif command
If a fade is already in progress when a new fade request is made, the new fade request will be applied starting at the most recent gain value of the fade that was in progress. Even if this new fade request has a start time in the future, any fade that is currently in progress will be stopped at the most recent gain value. If the specified start time of a fade is already in the past, but the projected end time of the fade has not yet passed, the fade is started immediately at the gain value that is supposedly in progress. If the projected end time is also in the past, the end gain is set immediately.
If instead, the user wishes to specify an minimum amount of time that a player should remain silent between requests, then the EVENT_DELAY should be used to effectively produce a start time which is relative to the completion of the previous request.
Ifnot [scopeBlock] [scopeParm] [optional int compare value]- start of conditional execution block. Needs matching endif command
InboxStep: Sending START event!
Invalid memory block: end of track %d %s (addr 0x%08X)
start of track %d %s (addr 0x%08X)
Invalid position error: end of track %d %s (%.2f, %.2f)
start of track %d %s (%.2f, %.2f)
It's an error to set both read_data_fn and write_data_fn in the
Libpng error no. %s: %s
Loop or previous error loading module '%s'
METRIC: Season Start - %s - %f seconds
METRIC: Season Start Time: %d seconds
MRecordTimeStart = %lld
Memory Replay FrameReset start
Modify Start Time
MovieEncoder_Avi: Allocator is NULL - unable to allocate compression buffer.
MovieEncoder_Avi: Allocator is NULL - unable to allocate index block.
MovieEncoder_Avi: Allocator is NULL - unable to allocate index buffer.
MovieEncoder_Avi: Allocator is NULL - unable to free compression buffer.
MovieEncoder_Avi: Allocator is NULL - unable to free index buffers.
NStart >
= 0
NStart <
pArray->
mnLength
NStart <
= pArray->
mnLength
New Start Time
Note also that it is possible to explicitly set a start gain prior to a given fade. This can be achieved by specifying a fade of duration 0 with a start time of 0 and an end gain equal to the gain value desired. This will only take if both these parameters (fadeTime and startTime) are set to 0. One example of when this feature might be useful is if the user wants to ramp up a signal from 0 but the current gain of the fader is not 0, such as at PlugIn startup (when the gain is set to 1). In this case, the user would specify two fades in succession, one to set the current gain to 0, and a second one to initiate the ramping up of the signal. Since the user may require that the initial gain setting (to 0 in this case) not delay the subsequent signal ramp up, both these gain fade events may be applied within the same lock (EA::Audio::Core::Lock()/Unlock()). As such, it is a special case, since otherwise the Gain Fader does not support queuing of gain fade events. It should also be noted that this feature should be used with extreme caution, since changing the gain value over a fade duration of 0 will introduce a click if the PlugIn is passing a signal at the time the gain is changed. It should only be used if there is no signal being passed or it is known that the current signal amplitude is 0.
Note that although the start time is specified with very high precision that allows sample accurate results, rounding errors may occur in some cases that could cause the sample number to be off by 1 sample from where the desired results are. This would most often be a problem if attempting to start a new sample relative to another sample that is playing and being pitched, as the other sample position will be located at a fractional position within a single sampling point, due to the nature of resampling. If possible, the stitching mechanism provided by queueing samples one after another in a single SndPlayer1 instance should be used if seamless stitching is required.
Note that although the startTime is specified with very high precision that allows sample accurate results, rounding errors may occur in some cases that could cause the sample number to be off by 1 sample from where the desired results are. This would most often be a problem if attempting to start a new sample relative to another sample that is playing and being pitched, as the other sample position will be located at a fractional position within a single sampling point, due to the nature of resampling. If possible, the stitching mechanism provided by queueing samples one after another in a single GenericPlayer instance should be used if seamless stitching is required.
Note that although the startTime is specified with very high precision, which in most cases allows sample accurate results, some errors may occur. This can happen when the relative time is large. It may also occur because of rounding errors that could cause the sample number to be off by 1 sample from where the desired results are. This would most often be a problem if attempting to start a new sample relative to another sample that is playing and being pitched, as the other sample position will be located at a fractional position within a single sampling point, due to the nature of resampling. If possible, the stitching mechanism provided by queueing samples one after another in a single SamplePlayer instance should be used if seamless stitching is required.
Note that start time is not precise, so it is not meant for seamless stitching. The primary use for this is to allow users to pre-buffer a stream by queueing it up with a start time that is way in the future (See EA::Audio::Core::HwPlayer::EVENT_MODIFYRELATIVESTARTTIME).
On the PC, the Dac sends its output through a DirectSound secondary buffer with the cooperative level set to PRIORITY, and the focus set to GLOBAL. The priority used allows other code to create DirectSound buffers outside of EAAudioCore if required, and the code should get along (they will both be heard). The GLOBAL focus means that even if the application loses focus, EAAudioCore will still be heard (this greatly simplifies the complexities of handling focus at a low level). If the application does not require EAAudioCore to continue being heard when losing focus, it should set a master volume gain PlugIn on the mastering voice to 0.0 when focus is lost, and set it back to its previous value when focus is regained. Another option is to mute the master Voice via the Voice::Mute() call. If the application requires a complete pausing of audio when it loses focus, it can alternatively call the EA::Audio::Core::Dac::STOP and EA::Audio::Core::Dac::START events.
Output file write error --- out of disk space?
PANIC: unprotected error in call to Lua API (%s)
Player International Leave Start Date Offset
PlayerContractManager::InitPlayersContract - [current date %d] [season start date %d] [season end date %d]
PointScaleEnable changed by Effect! (not allowed)
PointSpriteEnable changed by Effect! (not allowed)
RNACore: Error unrecognized vertex-element %s
RNACore: Error unrecognized vertex-format %s
RangeFogEnable changed by Effect! (not allowed)
ReplayStreamAbstract Replay FrameReset start
Reporting mem/job error to bugsentry....
Reporting server error to bugsentry....
Restart AI
Reverb 0 Envelope Start Level
Reverb 1 Envelope Start Level
Runtime error
START FROZEN
Saving Job Error Info...
Saving Memory Error Info...
Saving Server Error Info...
Season Start S/M %s
Server error - activation check error
Server error - application not found
Server error - can not update statistics
Server error - error during creating activation
Server error - error during verification
Server error - global verify error
Server error - internal error
Server error - invalid geographical region
Server error - license end date reached
Server error - license expired
Server error - project (CPA) not active
Server error - purchase error
Server error - purchase not found
Server error - registration required
Server error - serial revoked too often
Server error - serial revoked too often within timeframe
Server error - start date not reached
Server error - too many activation on different PC's
Server error - too many activations on same PC
Server error - too many activations within timeframe
Server error - too many total activations
Server error - unknown server error
Server error - wrong/invalid serial
Sp -= %d. Creates %d bytes of space for this subroutine's local variables. Usually at the start of a subroutine.
SpecularEnable changed by Effect! (not allowed)
Start Audio Logic Init
Start Fade
Start Gain
Start Menu
Start Mode
Start Of Frame 0x%02x: width=%u, height=%u, components=%d
Start Of Scan: %d components
Start Time
Start Time Offset
Start loaddb %s
Start of Image
Start savedb %s
StencilEnable changed by Effect! (not allowed)
Storagestart = %f, storageend = %f, time = %f
Syntax error in subpattern name (missing terminator)
System error %d
The CameraView RigOp requires the Animation Channel Map Rig Feature to be enabled in order to work correctly. Please open your Camera Rig, enable the Animation Channel Map feature and set up mappings for Position, Orientation, Roll, Field of View and Clip Planes.
The CameraView RigOp requires the Projection Matrix Channels Rig Feature to be enabled in order to work correctly. Please open your Camera Rig and enable the Projection Matrix Channels feature.
The Delay PlugIn allows for the application of delay to the audio signal. This delay can be applied with a variable amount of feedback, ranging from no feedback (0) to near full feedback (0.99). The delay amount is specified in seconds, and a delay of 0 turns the PlugIn off. The PlugIn allows for dynamically changing the delay time and feedback amount by smoothly cross fading from one set of parameters to another. This cross fade currently occurs over a 128 sample period. In the case where a larger buffer is required due to an increased delay time, the PlugIn will allocate a larger buffer and seamlessly transfer the existing buffered data to it. It will not start the cross fade until this new larger buffer has filled up the new sample locations, which is the time difference between the new delay time and that supported by the previous buffer size. The buffer will not be resized when a smaller delay time is requested, though this does not effect the delay itself, only how much buffer it is using. To avoid resizing when using a larger delay time, the use of the constructor parameter maxDelaySeconds to presize the delay buffer is required is strongly recommended.
The GainFader PlugIn provides a mechanism for changing the audio signal gain at a specified start time, over some time period, and according to some amplitude change characteristic. Currently three fade characteristics are supported
linear amplitude, linear power, and sine amplitude.
The GenericPlayer provides an open system for registering and playback any generic file format which has been registered with the GenericFormatRegistry. We currently provide support for the most common formats. But These formats can be authored by users to provide playback support for additional file formats. The GenericFormatRegistry should be initialized once at start up with all the necessary formats. Note that theregistration order of the formats determines the identification order. See the EA::Audio::Core::GenericFormatRegistry documentation for details
The Play event specifies format information about the source sample data that will be submitted and enters the PacketPlayer into an active state where submitted packets will start being processed.
The Play event will begin generation of the sine wave at the specified start time with the currently set playback frequency.
The ReverbIR1 PlugIn also supports the enveloping of either or both of the impulse responses. This allows for the possibility of shortening a given impulse response by fading it out to zero over a specified amount of time (i.e., ATTRIBUTE_REVERB0ENVELOPELENGTH for reverb 0).A start gain and envelope shape also must be specified via attributes (i.e., ATTRIBUTE_REVERB0ENVELOPESTARTLEVEL and ATTRIBUTE_REVERB0ENVELOPETYPE for reverb 0 respectively). This enveloping feature should be used with care, as the artificial shortening of an impulse response with a ramp does not exactly reflect a real-world scenario. If employed, users should experiment with the various enveloping parameters and impulse response(s) to confirm the resulting reverb suits their needs. The potential benefit with the use of enveloping is that it provides the user with some control over a given impulse response, allowing for greater variability and/or the use of less impulse response files. In the extreme, a user might use a single long impulse response, and simply adjust its length via enveloping to tailor it to the various reverberent spaces of the game. Combining enveloping with the use of two active individually gained reverb files can result in a high degree of flexibility and real-time creative control.
The absolute start time is applied to all requests, even if they are queued up behind each other. The start time will never force a request to prematurely interrupt a request ahead of it in the queue.
The primary use for this is to allow users to pre-buffer a stream by queueing it up with a startTime a long way in the future (e.g. currentTime + 1000000 seconds), then checking on its status with EVENT_GETREQUESTBUFFERED. When the user decides that they are ready to start the stream they can then use this event to modify the start time to zero, thus triggering the stream to start.
The primary use for this is to allow users to pre-buffer a stream by queueing it up with a startTime of SamplePlayer::REQUESTSTARTTIME_NEVER (i.e. will not start), then checking on its buffer status with SamplePlayer::EVENT_GETBUFFERLEVEL to determine when the data has arrived. When the user decides that the request should begin, they can then allow the playback of the streamed request by changing its start time. Using this event, they modify the start time to SamplePlayer::REQUESTSTARTTIME_IMMEDIATE (or 0.0f), thus triggering the stream to start.
The start event tells the AiffWriter to begin writing samples to the specified AIFF file.
The start event tells the SampleCapture PlugIn to begin recording samples and returning them to the client.
The start time can be used to determine when the sine wave will begin generating output. The resolution of the start time is sample accurate.
The start time is applied to all requests, even if they are queued up behind each other. When the current request finishes playing, the next request's start time is evaluated, and the sample will either stitch directly to the end of the previous request if the start time was less than the current system time at that point, or the sample will wait until the system time reaches its start time. Note that the start time will never force a request to prematurely interrupt a request ahead of it in the queue.
The startTime is applied to all requests, even if they are queued up behind each other. When the current request finishes playing, the next request's startTime is evaluated, and the sample will either stitch directly to the end of the previous request if the startTime was less than the current system time at that point, or the sample will wait until the system time reaches it's start time. Note that the startTime will never force a request to prematurely interrupt a request ahead of it in the queue.
The startTime is applied to all requests, even if they are queued up behind each other. When the current request finishes playing, the next request's startTime is evaluated. The startTime will be relative to when the play request was delivered to the player
not relative to the completion time of the previous request. The sample will then either stitch directly to the end of the previous request if the calculated startTime was less than the current system time at that point, or the sample will wait until the system time reaches its start time. Note that the startTime will never force a request to prematurely interrupt a request ahead of it in the queue.
This attribute specifies how long a run of zeros (seconds) is required in the audio input in order for the PlugIn to declare a zero run and reset its beat detection state machine and confidence metric. This is useful to give the algorithm a hint that the song has changed, and therefore all past input audio data and the associated calculations the Beat Detector made based on it is irrelevent and it should start fresh in its calulcations. It is recommended to use a very small value so that the zero detect algorithm responds quickly to a zero run and, for example, sets a correspondingly low confidence value quickly.
This attribute specifies the start level of the first envelope. A value of 1.0 means that the envelope will start at a normal level, while 2.0 signifies a start level of double the normal level. This and all other envelope related attributes is respected only when enveloping is enabled at construction time (ENVELOPINGMODEPARAM_ENABLE).
This attribute specifies the start level of the second envelope. A value of 1.0 means that the envelope will start at a normal level, while 2.0 signifies a start level of double the normal level. This and all other envelope related attributes is respected only when enveloping is enabled at construction time (ENVELOPINGMODEPARAM_ENABLE).
This attribute specifies whether or not to enable hard clipping by the Dac PlugIn. The default is set to true for backward compatibility. For platforms with lower end processors, the attribute provides a way to save CPU cycles by turning off clipping. This should only be done if users have ensured that the final Dac audio output is never in need of clipping by staying within the range of +/- 1.0, and/or they have confirmed that the platform(s) being used employs its own clipping. In this latter case where platforms automatically provide their own clipping, the Dac PlugIn will not do any additional clipping and so this attribute is ignored.
This constructor parameter specifies if an envelope can be used. An envelope may be used to dynamically change a reverb while using the same reverb file. Enabling the use of an envelope will result in a small amount of memory being allocated for enveloping purposes, the size of which depends on how long the reverb files are. An envelope scales the reverb file from the beginning of the impulse response with a starting gain set by the envelope start level attribute (i.e., ATTRIBUTE_REVERB0ENVELOPESTARTLEVEL for reverb0) down to a gain of zero. The gain decreases based on the envelope type (i.e., ATTRIBUTE_REVERB0ENVELOPETYPE for reverb0), which currently can be either linearly or exponentially shaped. Envelope duration is set by the envelope length attribute (i.e., ATTRIBUTE_REVERB0ENVELOPELENGTH for reverb0), which specifies in seconds the time over which the envelope decays from its starting gain to zero.
This error is generated when invalid data is specified for a command. Basically, the RSX interprets a command and data together as a set, and this error indicates that the command is supported but the data for that command is a value outside of the allowable range. Be sure to verify that the argument values that are set for each command generation function fall within the ranges that are specified in the libgcm Reference.
This error is generated when the RSX performs a vertex or index fetch from an invalid offset. This often occurs when the value of the offset argument of cellGcmSetVertexDataArray() is invalid or when the index that was being used by the previous vertex shader is not currently being used, but it has not been invalidated and a vertex fetch occurs unexpectedly. The error also occurs when the value of the indices argument for specifying the offset to the index for cellGcmSetDrawIndexArray() is invalid.
This event allows users to modify the start time of a request after queueing it up with EVENT_PLAY.
This event allows users to modify the start time of a request after queueing it up with EVENT_PLAY1.
This event results in the scheduling of a gain fade. A relative start time offset from the current system time for the change is specified in seconds. A start time of 0.0 will always cause the fade to start immediately. A fade time is also specified in seconds, and denotes the duration of the amplitude change. This change (or fade) will then occur over the fade time duration to smoothly arrive at the specified final gain value, given by the endGain parameter.
This is an input parameter used to specify the request handle (returned from the EVENT_PLAY event) of the request that the user would like to modify the start time of.
This is an input parameter used to specify the request handle (returned from the EVENT_PLAY event) of the request that the user would like to modify the start time of.
This is an input parameter used to specify the request handle (returned from the EVENT_PLAY1 event) of the request that the user would like to modify the start time of.
This parameter is used to change the relative start time of a previously queued request. The units are in seconds, and are relative the mixer time when the event is submitted to EAAudioCore.
This parameter specifies the start time offset of the fade in seconds, relative to the current time.
To trigger starting of a sample in the future, simply indicate how many seconds in the future you want the sound to be triggered from the current system time. A value of REQUESTSTARTTIME_IMMEDIATE or 0.0f will always play the sample as soon as possible. Using a value of REQUESTSTARTTIME_NEVER will queue a request on the player but not allow it to start playback. This technique can be used with EVENT_MODIFYSTARTTIME to first prebuffer a streamed request and then start playback instantly at an appropriate time.
TransferManager::Unable to locate (%d) teams after (%d) passes. Found (%d) teams for player ( %d ). Budgets ignored (%s)
UNKNOWN ERROR CODE!!!!
Unable to add comment node to parent <
%s>
!
Unable to add declaration node to parent <
%s>
!
Unable to add element node to parent <
%s>
!
Unable to add value node of type %d to parent <
%s>
!
Unable to allocate memory for attribute '%s' in element %s!
Unable to allocate memory for name!
Unable to allocate memory for value!
Unable to allocate string buffer!
Unable to dump given function
Unable to expand string buffer to %d bytes!
Unable to find equivalent output section for symbol '%s' from section '%s'
Unable to generate a unique filename
Unable to get JSP URL
Unable to get ModuleFileName
Unable to open %s
Unable to replace slotted sample, no matching loader table.
Unable to replace slotted sample. Selected asset is larger than the load buffer. (bufferSize=%u, slotSize=%u, assetSize=%u)
Unable to replace slotted sample. Selected asset is larger than the target slot.
Unable to trace. No PS3 dump file has been loaded.
Unable to write international text
Undocumented error #%d
Unexpected error retrieving current renderstate!
Unknown error code
Used to indicate where playback should begin within the given asset. Times are given relative to the start of the sample. For example, if the given SndPlayer asset is 4.0 seconds long, then passing a value of 2.0 will skip the first half of the asset and only play back the last half of the file. Note that no declicking of the seek point is done by the SndPlayer1 PlugIn.
Used to indicate where playback should begin within the given asset. Times are given relative to the start of the sample. For example, if the given SndPlayer asset is 4.0 seconds long, then passing a value of 2.0 will skip the first half of the asset and only play the last half. Note that no declicking of the seek point is done by the player plug-in. This means that users need to ensure that the given seek time is an appropriate location in the asset.
WSAStartup failed, error %d.
When the START event is called, sample recording commences. Callbacks will start occurring every time the SampleCapture PlugIn has processed a frame of samples. When STOP is called, sample recording terminates and a final callback will be made with any remaining data.
When the START event is called, the AIFF file header is written out. The AIFF file is then appended to each time the PlugIn processes the sample data. When STOP is called, the AIFF file header is updated with the correct number of samples. If the AiffWriter PlugIn is not stopped or released gracefully, then the header will not be updated, and the AIFF file will likely appear corrupt.
When the playback event occurs, the absolute start time will be evaluated from PLAYPARAM_RELATIVESTARTTIME plus the current system time. When the request reaches the start of the queue, the absolute start time will be compared with the system time. If the absolute start time elapses the current system time, the request will play without any delay. Otherwise, the request will be played when the system time has been reached. A value of 0.0 will always play the sample as soon as possible.
[DLC] Unable to generate dlc.toc file, skipping
[HIGHTLIGHT TEST]: SaveHighlight, invalid start [%d], clipId = [%d]
[Memory Stream] mHDDRecordTimeStart = %lld
[REPLAY COMPRESS]: decode time error in codec[0x%x] [0x%x, 0x%x] %s [%f]
[Replay] DeltaCompress, FullReset start
[Replay] Setting replay start time to %lld
[THREAD: %d] ScreenController::HidePlayerPopup -- start [%s]
[THREAD: %d] ScreenController::HidePopup -- start [%s]
[WARN] Time to copy %f (%lld) is before in memory replay start time %f (%lld)!