구차니의 잡동사니 모음

virtualbox 4.3.12 버전에서 사용가능한 네트워크는

Not attached

NAT - Network Address Translation

NAT Network

Bridged networking

Internal networking

Host-only networking

Generic networking
7가지인데

인터넷이 되는 설정은 NAT / Bridged networking 뿐이다?
NAT Network 로 설정하면
NAT Network 로 설정한 VM 들 끼리는 통신이 가능하지만
외부로 인터넷 접속이 안된다 -_-

(제대로 된 테스트 필요)

	internet	other VMs
Not attached	x	x
NAT	o	x / DHCP
NAT Network	x	o / DHCP
Bridged networking	o	o (단, 다른 VM도 Bridged)
Internal networking
Host-only networking	x	x

[링크 : https://www.virtualbox.org/manual/ch06.html]
[링크 : https://www.virtualbox.org/manual/ch06.html#network_nat_service] 

 Network Address Translation (NAT) is the simplest way of accessing an external network from a virtual machine. Usually, it does not require any configuration on the host network and guest system. For this reason, it is the default networking mode in VirtualBox.

 The Network Address Translation (NAT) service works in a similar way to a home router, grouping the systems using it into a network and preventing systems outside of this network from directly accessing systems inside it, but letting systems inside communicate with each other and with systems outside using TCP and UDP over IPv4 and IPv6.

 Internal Networking is similar to bridged networking in that the VM can directly communicate with the outside world. However, the "outside world" is limited to other VMs on the same host which connect to the same internal network.

ffmpeg 컴파일 옵션중에
  --enable-thumb           compile for Thumb instruction set

thumb 활성화가 있고 cpu도 지원을 해서

Processor       : ARM926EJ-S rev 5 (v5l) BogoMIPS        : 226.09 Features        : swp half thumb fastmult edsp java

 --enable-armv5te 랑 같이 썼더니.... 안되잖아!!!! ㅠㅠ

단점들을 나열해 볼게요.    1. 분기 명령어를 제외하고는 조건부 실행이 안됩니다.  2. 레지스터 사용이 R0~R7으로 제안 됩니다. 3. Immediate 상수 값의 사용 범위가 제한적입니다. 4. Inline barrel shifter의 사용이 제안적입니다. 5. Exception 처리를 할 수 없습니다. [링크 : http://recipes.egloos.com/viewer/5651064]

mplayer에서는 -ao alsa:device=hw:0.0 식으로 음악을 바로 재생을 할 수 있는데..
ffmpeg은 입력을 받아서 저장은 되도.. 플레이는 안되나.. 보다 싶긴하다.

$ man ffmpeg    Video and Audio grabbing        If you specify the input format and device then ffmpeg can grab video and audio directly.                ffmpeg -f oss -i /dev/dsp -f video4linux2 -i /dev/video0 /tmp/out.mpg        Or with an ALSA audio source (mono input, card id 1) instead of OSS:                ffmpeg -f alsa -ac 1 -i hw:1 -f video4linux2 -i /dev/video0 /tmp/out.mpg        Note that you must activate the right video source and channel before launching ffmpeg with any TV viewer        such as <http://linux.bytesex.org/xawtv/> by Gerd Knorr. You also have to set the audio recording levels        correctly with a standard mixer.

$ mam mplayer AUDIO OUTPUT DRIVERS (MPLAYER ONLY)        Audio output drivers are interfaces to different audio output facilities.  The syntax is:        -ao <driver1[:suboption1[=value]:...],driver2,...[,]>               Specify a priority list of audio output drivers to be used.        If  the  list has a trailing ',' MPlayer will fall back on drivers not contained in the list.  Suboptions        are optional and can mostly be omitted.        NOTE: See -ao help for a list of compiled-in audio output drivers.        EXAMPLE:                  -ao alsa,oss,                       Try the ALSA driver, then the OSS driver, then others.                  -ao alsa:noblock:device=hw=0.3                       Sets noblock-mode and the device-name as first card, fourth device.

mp3만 가속됨
flac wav 는 거의 cpu를 소비하지 않는 수준
wmv ogg aac aacp는 우걱우걱이라 armv5te 에서는 무리 

전화해서 물어봐도
데이터에는 손상이 없다고 괜찮다고 라는데
무료라고는 하지만 영...

유료 고객이었다면 다르게 대응했을까 하는 생각이 조금 들긴 하지만
유료였다면 아마.. 2일이나 접속도 안되는게 말이 되냐고 아마 길길이 날뛰었을지도 ㅋ 

i.mx283 / armv5te 아키텍쳐의
dsp enchancement 명령을 사용하기 위한 컴파일 옵션

--cpu=armv5te --enable-armv5te 

코덱마다 다르고 시간마다 달라 정형화하기 힘들지만
classic channel 인터넷 방송은 정상재생이냐 끊어지냐 수준
80% vs 13% 급이고

mbc fm 은
거의 차이가 없는 78% 이나
load average가 5.0 vs 4.8 정도로 유의미하게 나진 않는다

[링크 : http://ffmpeg.org/pipermail/ffmpeg-user/2011-May/000833.html]

[링크: http://qntm.org/files/perl/perl_kr.html]
[링크: http://www.perl.or.kr/perl_iyagi]

[링크 : http://www.perl.or.kr/perl_iyagi/dir] 디렉토리 목록 얻기
[링크 : http://gypark.pe.kr/wiki/Perl/배열]
[링크 : http://perldoc.perl.org/functions/open.html]
[링크 : http://perldoc.perl.org/functions/sort.html]
[링크 : http://perltricks.com/article/17/2013/4/12/Perl-for-loops] 
[링크 : http://www.tutorialspoint.com/perl/perl_qx.htm ] 

테스트용으로
channel  / samping rate / bitrate  등을 변경하기 위한 방법

#!/bin/sh # CBR - channel / bitrate / resample # VBR - channel / quality / resample # bitrate - CBR # MPEG1 - 320 256 224 192 160 128 118 96 80 64 56 48 40 32 # MPEG2 - 160 144 128 112 96 80 64 56 48 40 32 24 16 8 # bitrate - VBR # 9 8 7 6 5 4 3 2 1 0 # sampling # mpeg1-3 48 44.1 32 # mpeg2-3 24 22.05 16 # mpeg2.5-3 12 11.025 8 for rate in 48 44.1 32 do         for bps in 320 256 224 192 160 128 118 96 80 64 56 48 40 32         do                 lame -b ${bps} -m s --resample ${rate} $1 C${bps}bps_${rate}KH_2ch.mp3                 lame -b ${bps} -m m --resample ${rate} $1 C${bps}bps_${rate}KH_1ch.mp3         done done for rate in 48 44.1 32 do         for bps in 9 8 7 6 5 4 3 2 1 0         do                 lame -V ${bps} -m s --resample ${rate} $1 V${bps}bps_${rate}KH_2ch.mp3                 lame -V ${bps} -m m --resample ${rate} $1 V${bps}bps_${rate}KH_1ch.mp3         done done

---
mp3가 mpeg1 layer3인데.
mpeg2 layer3 와
mpeg2.5 layer3는 도무지 뭔지 ㅠㅠ

-m mode               mode = s, j, f, d, m               Joint-stereo is the default mode for stereo files with VBR  when  -V  is  more  than  4  or  fixed               bitrates  of  160kbs  or  less.   At  higher fixed bitrates or higher VBR settings, the default is               stereo.               (s)imple stereo               In this mode, the encoder makes no use of potentially existing correlations between the two  input               channels.   It  can, however, negotiate the bit demand between both channel, i.e. give one channel               more bits if the other contains silence or needs less bits because of a lower complexity.               (j)oint stereo               In this mode, the encoder will make use of a correlation between both channels.  The  signal  will               be  matrixed into a sum ("mid"), computed by L+R, and difference ("side") signal, computed by L-R,               and more bits are allocated to the mid channel.  This will effectively increase the  bandwidth  if               the  signal  does  not have too much stereo separation, thus giving a significant gain in encoding               quality.               Using mid/side stereo inappropriately can  result  in  audible  compression  artifacts.   To  much               switching  between mid/side and regular stereo can also sound bad.  To determine when to switch to               mid/side stereo, LAME uses a much more sophisticated algorithm than that described in the ISO doc‐               umentation, and thus is safe to use in joint stereo mode.               (f)orced MS stereo               This  mode  will  force  MS stereo on all frames.  It is slightly faster than joint stereo, but it               should be used only if you are sure that every frame of the input file has very little stereo sep‐               aration.               (d)ual mono               In  this  mode,  the  2  channels  will  be totally independently encoded.  Each channel will have               exactly half of the bitrate.  This mode is designed for applications like dual languages  encoding               (for example: English in one channel and French in the other).  Using this encoding mode for regu‐               lar stereo files will result in a lower quality encoding.               (m)ono               The input will be encoded as a mono signal.  If it was a stereo signal, it will be downsampled  to               mono.  The downmix is calculated as the sum of the left and right channel, attenuated by 6 dB.

--resample sfreq               sfreq = 8, 11.025, 12, 16, 22.05, 24, 32, 44.1, 48               Select output sampling frequency (only supported for encoding).               If not specified, LAME will automatically resample the input when using high compression ratios.

CBR (constant bitrate, the default) options:        -b n   For MPEG1 (sampling frequencies of 32, 44.1 and 48 kHz)               n = 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320               For MPEG2 (sampling frequencies of 16, 22.05 and 24 kHz)               n = 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160               Default is 128 for MPEG1 and 64 for MPEG2.        --cbr  enforce use of constant bitrate        ABR (average bitrate) options:        --abr n               Turns  on encoding with a targeted average bitrate of n kbits, allowing to use frames of different               sizes.  The allowed range of n is 8 - 310, you can use any integer value within that range.               It can be combined with the -b and -B switches like: lame --abr 123 -b 64 -B 192 a.wav a.mp3 which               would limit the allowed frame sizes between 64 and 192 kbits.               The  use  of  -B  is NOT RECOMMENDED.  A 128 kbps CBR bitstream, because of the bit reservoir, can               actually have frames which use as many bits as a 320 kbps frame.  VBR modes minimize  the  use  of               the  bit  reservoir,  and  thus  need  to allow 320 kbps frames to get the same flexibility as CBR               streams.        VBR (variable bitrate) options:        -v     use variable bitrate (--vbr-old)        --vbr-old               Invokes the oldest, most tested VBR algorithm.  It produces very good quality files, though is not               very fast.  This has, up through v3.89, been considered the "workhorse" VBR algorithm.        --vbr-new               Invokes the newest VBR algorithm.  During the development of version 3.90, considerable tuning was               done on this algorithm, and it is now considered to be on par with the original --vbr-old.  It has               the added advantage of being very fast (over twice as fast as --vbr-old).        -V n   0 <= n <= 9               Enable  VBR  (Variable BitRate) and specifies the value of VBR quality (default = 4).  0 = highest               quality.

[링크 : http://tuxtweaks.com/2008/08/how-to-resample-mp3-audio-files-on-linux-using-lame/]