WIP Operation Resources - APPLIED_RESOURCE_UNITS, APPLIED_RESOURCE_VALUE ?

What do we mean by these columns functionally. I have gone through ETRMS, but couldnt understand them. Can anyone explain what these columns mean functionally.
These columns are present in WIP_OPERATION_RESOURCES table.
Also what is the difference between Assigned Units and Applied Resource Units?
Thanks and Regards,
Upendra
Edited by: upendra.pisupati on Dec 20, 2011 10:46 PM

Hi,
These columns talk about how many resources are actually consumed during production run of the job.
For example, if you have defined routing to consume 2 Hrs of resource on item basis at operation 10,
(in resource master, say cost of resource is $15 / Hr)
and if you have created a job for 7 Ea of finished good and you are moving 10 Ea of from step 10 (Queue) to 10 (To Move)
Applied Resource Units will be 7 Ea * 2 Hrs / item = 14 Hrs
Applied Resource Value will be 7 Ea * 2 Hrs / item * $15 / Hr = $ 210
Additionally, you can also have manual resource transactions, due to which applied_resource_units and applied_resource_value can get updated.
Thanks,
Hrishikesh

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Hi Experts,
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Hi,
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Sreekanth Munagala wrote:
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Purpose
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union
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union
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union
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bom_resources r
where r.resource_code = '&resource_name'
and r.organization_id = &org_id
and r.resource_id = X.resource_id
group by tablename;
Reference
Note 279142.1

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Hello,
i have problem :)
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Dear Mr.Riccardo
I would like to bring to your notice a document from Metalink. It says that we cannot do move transaction of Serial controlled Items in Desktop forms.....
Subject: Is Serial Number Association Functionality Only Available Using Supply Chain Mobile Applications?
Doc ID: Note:265512.1 Type: HOWTO
Last Revision Date: 01-APR-2004 Status: PUBLISHED
The information in this article applies to:
Oracle Work in Process - Version: 11.5.9
This problem can occur on any platform.
Goal
Is the new functionality of Serial Number Association only available through Mobile Applications?
Fix
This functionality provides support for standard discrete jobs with or without a routing.
The User Interface for this functionality is via Mobile Applications
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disabling the last.fm playlist plugin because account is not configured
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snippet from mpd.log:
Jun 28 00:54 : client: [0] process command "playlistfind filename "Lossy - Tagged and Sorted/Wild Oats/WO-0001 - Kyle Hall - Worx Of Art EP 1 - 2008/Kyle Hall - 01 - Fuse N Me .mp3""
Jun 28 00:54 : client: [0] command returned 0
Jun 28 00:54 : client: [0] process command "addid "Lossy - Tagged and Sorted/Wild Oats/WO-0001 - Kyle Hall - Worx Of Art EP 1 - 2008/Kyle Hall - 01 - Fuse N Me .mp3""
Jun 28 00:54 : add to playlist: Lossy - Tagged and Sorted/Wild Oats/WO-0001 - Kyle Hall - Worx Of Art EP 1 - 2008/Kyle Hall - 01 - Fuse N Me .mp3
Jun 28 00:54 : database: get song: Lossy - Tagged and Sorted/Wild Oats/WO-0001 - Kyle Hall - Worx Of Art EP 1 - 2008/Kyle Hall - 01 - Fuse N Me .mp3
Jun 28 00:54 : client: [0] command returned 0
Jun 28 00:54 : client: [0] process command "playid "7""
Jun 28 00:54 : playlist: play 7:"Lossy - Tagged and Sorted/Wild Oats/WO-0001 - Kyle Hall - Worx Of Art EP 1 - 2008/Kyle Hall - 01 - Fuse N Me .mp3"
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Jun 28 00:54 : decoder_control: mixramp_start = NULL
Jun 28 00:54 : decoder_control: mixramp_prev_end = NULL
Jun 28 00:54 : client: [0] command returned 0
Jun 28 00:54 : client: [0] process command "status"
Jun 28 00:54 : client: [0] command returned 0
Jun 28 00:54 : client: [0] process command "idle"
Jun 28 00:54 : client: [0] command returned 1
Jun 28 00:54 : client: [0] process command "playlistid "7""
Jun 28 00:54 : client: [0] command returned 0
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Jun 28 00:54 : mad: LAME peak found: 0.000000
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# blocks. Setting this block is optional, though the server will only attempt
# autodetection for one sound card.
# See <http://mpd.wikia.com/wiki/Configuration#Audio_Outputs> for examples of
# other audio outputs.
# An example of an ALSA output:
audio_output {
type "alsa"
name "My ALSA Device"
device "hw:0,3" # optional
## format "44100:16:2" # optional
## mixer_type "hardware" # optional
## mixer_device "default" # optional
## mixer_control "Master" # optional
## mixer_index "0" # optional
# An example of an OSS output:
#audio_output {
# type "oss"
# name "My OSS Device"
## device "/dev/dsp" # optional
## format "44100:16:2" # optional
## mixer_type "hardware" # optional
## mixer_device "/dev/mixer" # optional
## mixer_control "PCM" # optional
# An example of a shout output (for streaming to Icecast):
#audio_output {
# type "shout"
# encoding "ogg" # optional
# name "My Shout Stream"
# host "localhost"
# port "8000"
# mount "/mpd.ogg"
# password "hackme"
# quality "5.0"
# bitrate "128"
# format "44100:16:1"
## protocol "icecast2" # optional
## user "source" # optional
## description "My Stream Description" # optional
## genre "jazz" # optional
## public "no" # optional
## timeout "2" # optional
## mixer_type "software" # optional
# An example of a recorder output:
#audio_output {
# type "recorder"
# name "My recorder"
# encoder "vorbis" # optional, vorbis or lame
# path "/var/lib/mpd/recorder/mpd.ogg"
## quality "5.0" # do not define if bitrate is defined
# bitrate "128" # do not define if quality is defined
# format "44100:16:1"
# An example of a httpd output (built-in HTTP streaming server):
#audio_output {
# type "httpd"
# name "My HTTP Stream"
# encoder "vorbis" # optional, vorbis or lame
# port "8000"
## quality "5.0" # do not define if bitrate is defined
# bitrate "128" # do not define if quality is defined
# format "44100:16:1"
# max_clients "0" # optional 0=no limit
# An example of a pulseaudio output (streaming to a remote pulseaudio server)
#audio_output {
# type "pulse"
# name "My Pulse Output"
## server "remote_server" # optional
## sink "remote_server_sink" # optional
## Example "pipe" output:
#audio_output {
# type "pipe"
# name "my pipe"
# command "aplay -f cd 2>/dev/null"
## Or if you're want to use AudioCompress
# command "AudioCompress -m | aplay -f cd 2>/dev/null"
## Or to send raw PCM stream through PCM:
# command "nc example.org 8765"
# format "44100:16:2"
## An example of a null output (for no audio output):
#audio_output {
# type "null"
# name "My Null Output"
# mixer_type "none" # optional
# This setting will change all decoded audio to be converted to the specified
# format before being passed to the audio outputs. By default, this setting is
# disabled.
#audio_output_format "44100:16:2"
# If MPD has been compiled with libsamplerate support, this setting specifies
# the sample rate converter to use. Possible values can be found in the
# mpd.conf man page or the libsamplerate documentation. By default, this is
# setting is disabled.
#samplerate_converter "Fastest Sinc Interpolator"
# Normalization automatic volume adjustments ##################################
# This setting specifies the type of ReplayGain to use. This setting can have
# the argument "off", "album" or "track". See <http://www.replaygain.org>
# for more details. This setting is off by default.
#replaygain "album"
# This setting sets the pre-amp used for files that have ReplayGain tags. By
# default this setting is disabled.
#replaygain_preamp "0"
# This setting enables on-the-fly normalization volume adjustment. This will
# result in the volume of all playing audio to be adjusted so the output has
# equal "loudness". This setting is disabled by default.
#volume_normalization "no"
# MPD Internal Buffering ######################################################
# This setting adjusts the size of internal decoded audio buffering. Changing
# this may have undesired effects. Don't change this if you don't know what you
# are doing.
#audio_buffer_size "2048"
# This setting controls the percentage of the buffer which is filled before
# beginning to play. Increasing this reduces the chance of audio file skipping,
# at the cost of increased time prior to audio playback.
#buffer_before_play "10%"
# Resource Limitations ########################################################
# These settings are various limitations to prevent MPD from using too many
# resources. Generally, these settings should be minimized to prevent security
# risks, depending on the operating resources.
#connection_timeout "60"
#max_connections "10"
#max_playlist_length "16384"
#max_command_list_size "2048"
#max_output_buffer_size "8192"
# Character Encoding ##########################################################
# If file or directory names do not display correctly for your locale then you
# may need to modify this setting.
#filesystem_charset "UTF-8"
# This setting controls the encoding that ID3v1 tags should be converted from.
#id3v1_encoding "ISO-8859-1"
# SIDPlay decoder #############################################################
# songlength_database:
# Location of your songlengths file, as distributed with the HVSC.
# The sidplay plugin checks this for matching MD5 fingerprints.
# See http://www.c64.org/HVSC/DOCUMENTS/Songlengths.faq
# default_songlength:
# This is the default playing time in seconds for songs not in the
# songlength database, or in case you're not using a database.
# A value of 0 means play indefinitely.
# filter:
# Turns the SID filter emulation on or off.
#decoder {
# plugin "sidplay"
# songlength_database "/media/C64Music/DOCUMENTS/Songlengths.txt"
# default_songlength "120"
# filter "true"
Whenever I run xbmc using the hdmi digital output it just plays mp3 and flac files without problems.
Anyone any clues on how to solve this issue?
TIA, dafart
Since
Last edited by dafart (2010-07-01 00:38:09)

aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: NVidia [HDA NVidia], device 3: NVIDIA HDMI [NVIDIA HDMI]
Subdevices: 1/1
Subdevice #0: subdevice #0
Which correspondents fine with the
device "hw:0,3"
in my mpd.conf alsa section.
Also sound worked fine withh this config before I started installing ffmpeg-svn and alikes
hokasch:
I tried changing the mixer to "Master" but that did not change much.

WIP Operation Resources - APPLIED_RESOURCE_UNITS, APPLIED_RESOURCE_VALUE ?

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