I've to say this's the first non-mechanical keyboard which actually made it! For the first few months i.e.
Otherwise I've never encountered a keyboard which has lasted more than 2 months without a replacement.
The only complaint is that it's keys have become a little bit hard now; but that's not much of a problem (I'm a man) and despite the fact that this's not a mechanical keyboard, it doesn't have any multimedia keys.
Although HP claims this keyboard only works with Windows, I can confirm it's a standard plug and play keyboard.
Tuesday, November 11, 2014
Logitech MK220/M150 review
This Microsoft keyboard from Logitech is only intended for people with either 2 fingers or people who type with 2 fingers.
This 100% Microsoft, and Microsoft only compatible keyboard and mouse ensures tired arms cause --
1) It does not contain a palm rest
2) The keyboard does not have a lower edge boundary. The lowermost rows of keys make up the lower boundary of the keyboard.
Thus, if you attempt to rest your arms on the table any one of atrl, fn, alt, space, arrow keys and Logitech's favourite Windows keys will be pressed accidentally.
This's not the first time I've encountered a bad ergonomic keyboard from Logitech.
Another stupidity the company has done is that the mouse takes in 2 AA batteries making it heavy and keyboard takes in 2 AAA batteries in an attempt to make it light.
Anything more than 1M and the keyboard mouse starts giving communication issues. Maybe cause I made it work with Linux.
The DPI of the mouse is too low. So it has NO application in gaming or designing space. Buy a Microsoft keyboard/mouse for that . Logitech sells the same (it's only purpose being manufacturing HID devices to operate Windows) thing but at a higher price.
This 100% Microsoft, and Microsoft only compatible keyboard and mouse ensures tired arms cause --
1) It does not contain a palm rest
2) The keyboard does not have a lower edge boundary. The lowermost rows of keys make up the lower boundary of the keyboard.
Thus, if you attempt to rest your arms on the table any one of atrl, fn, alt, space, arrow keys and Logitech's favourite Windows keys will be pressed accidentally.
This's not the first time I've encountered a bad ergonomic keyboard from Logitech.
Another stupidity the company has done is that the mouse takes in 2 AA batteries making it heavy and keyboard takes in 2 AAA batteries in an attempt to make it light.
Anything more than 1M and the keyboard mouse starts giving communication issues. Maybe cause I made it work with Linux.
The DPI of the mouse is too low. So it has NO application in gaming or designing space. Buy a Microsoft keyboard/mouse for that . Logitech sells the same (it's only purpose being manufacturing HID devices to operate Windows) thing but at a higher price.
Friday, November 7, 2014
Booting GPT disks with Gigabyte's hybrid EFI.
I stated in my previous post, how horrible Gigabyte motherboards are. I continue to do the same now.
Basically Microsoft Gigabyte's hybrid EFI is nothing but a piece of Windows software which enables Windows to access 3+TB hard drive.
There's nothing EFI in the BIOS. It cannot boot GPT disks, it cannot run EFI applications and it does not provide any EFI API to the OS.
Basically Microsoft Gigabyte's hybrid EFI is nothing but a piece of Windows software which enables Windows to access 3+TB hard drive.
There's nothing EFI in the BIOS. It cannot boot GPT disks, it cannot run EFI applications and it does not provide any EFI API to the OS.
Sunday, November 2, 2014
Digital audio and sound cards interface (as provided to the OS).
PCM Or pulse code modulation is a method to digitize audio.
Just like a digital camera, the pixels are replaced with 'samples per second' and the bit depth is also in bits, which specifies the detail which each sample holds.
A sample contains the amplitude and frequency of the analog sound at a given point in time. Since analog is analog, it can have an infinite range of amplitude. But a digital sample can have only limited. Thus each sample also has a depth which's the amount of information stored in each sample (specified in bits, commonly 16, 24, 32) which then defines the range of amplitude the same can have. The depth is also called the 'sampling format' cause it defines the encoding of each sample as stored in a digital medium. There are various encoding schemes available like signed 8/16/24/32 bit, unsigned 8/16/24/32 bit, LE 8/16/24/32 bit, BE 8/16/24/32 bit etc...
To record a whole time range of an analog sound (or a complete waveform), samples are taken at regular intervals. This interval is called the sampling rate. This's analogous to the FPS in a video.
It happens that sampling rate limits the maximum frequency PCM can store (for some reason, probably cause low sampling rate has a high probability of missing out peaks). 44.1 KHZ sampling rate can store frequencies from 20 to 20,000 Hz; however higher the sampling rate, the better.
A sound card is a device which either converts analog signals to digital or opposite or both.
The sound cards do I/o with the OS in PCM format. For if you're playing an audio file which's not stored as PCM, the audio player will first have to decode it to PCM and then send it to the sound card. Same thing happens for recording. The audio sent by the sound card to the OS is almost always in PCM.
A sound card will support various sampling formats of I/O at various sampling rates.
When PCM is stored raw on a digital medium, the file usually has an extension '.wav'.
Wav can store audio in stereo. To accomplish this, the samples are placed in pairs in format such as LRLRLR; this's send to the sound card which knows this formatting. It'll send the left sample to the left speaker and right sample to the right speaker.
Now for multichannel audio. The PCM format can be encoded to store multichannel audio in similar way as stereo, where each channel will have it's individual sampling rate and L/R channel; this includes the subwoofer (LFE) channel.
For multichannel PCM to be decoded and changed to analog, the sound card needs to be told by the OS to operate in multichannel mode; in this mode the analog ports which otherwise were used for input may change their purpose for output.
Just like a digital camera, the pixels are replaced with 'samples per second' and the bit depth is also in bits, which specifies the detail which each sample holds.
A sample contains the amplitude and frequency of the analog sound at a given point in time. Since analog is analog, it can have an infinite range of amplitude. But a digital sample can have only limited. Thus each sample also has a depth which's the amount of information stored in each sample (specified in bits, commonly 16, 24, 32) which then defines the range of amplitude the same can have. The depth is also called the 'sampling format' cause it defines the encoding of each sample as stored in a digital medium. There are various encoding schemes available like signed 8/16/24/32 bit, unsigned 8/16/24/32 bit, LE 8/16/24/32 bit, BE 8/16/24/32 bit etc...
To record a whole time range of an analog sound (or a complete waveform), samples are taken at regular intervals. This interval is called the sampling rate. This's analogous to the FPS in a video.
It happens that sampling rate limits the maximum frequency PCM can store (for some reason, probably cause low sampling rate has a high probability of missing out peaks). 44.1 KHZ sampling rate can store frequencies from 20 to 20,000 Hz; however higher the sampling rate, the better.
A sound card is a device which either converts analog signals to digital or opposite or both.
The sound cards do I/o with the OS in PCM format. For if you're playing an audio file which's not stored as PCM, the audio player will first have to decode it to PCM and then send it to the sound card. Same thing happens for recording. The audio sent by the sound card to the OS is almost always in PCM.
A sound card will support various sampling formats of I/O at various sampling rates.
When PCM is stored raw on a digital medium, the file usually has an extension '.wav'.
Wav can store audio in stereo. To accomplish this, the samples are placed in pairs in format such as LRLRLR; this's send to the sound card which knows this formatting. It'll send the left sample to the left speaker and right sample to the right speaker.
Now for multichannel audio. The PCM format can be encoded to store multichannel audio in similar way as stereo, where each channel will have it's individual sampling rate and L/R channel; this includes the subwoofer (LFE) channel.
For multichannel PCM to be decoded and changed to analog, the sound card needs to be told by the OS to operate in multichannel mode; in this mode the analog ports which otherwise were used for input may change their purpose for output.
Understanding ALSA device, subdevice and cards.
ALSA calls each audio controller a 'card'; each card has 'devices'; a device is something which's capable of either processing an audio stream (for playback) or sending an audio stream (capture).
ALSA gives these cards and devices numbers starting from 0.
There are also subdevices which are a part of an input or output device. There must be at least 1 subdevice. In context of output devices, a device having multiple subdevices means the hardware can do mixing, i.e. it can take multiple streams of PCM and mix them to produce a single output. This's called hardware mixing and no, multichannel is a different thing. The no. of streams a device can take depends on the no. of sub devices a device has. Using ALSA API, you can send audio to each of these subdevices simultaneously; the result will be seen in the output audio.
Subdevice in capturing means the card can take and digitize input of multiple audio streams at once.
Usually subdevice have 'modes' in which they operate. For output subdevices, it means the multichannel mode they operate in. Like 2 or 4 or 6 or 7 etc... which's activate other multichannel ports on the sound card.
For input subdevice it means from which port will the input be taken from (like line in, mic, front mic etc...)
There's a ncurses based mixer which alsa provides name alsamixer... change your volumes from there. You may use amixer which you can use to change volume levels via command.
Here you can set your base and treble settings also (it might be present depending on your sound card). Press m for mute/unmute.
In the playback section, can set the mode the soundcard is (same as subdevice mode). Options will be 2ch, 4ch, 5ch etc... Also you'll be also to select the output levels of multiple speakers in the multichannel setup. In stereo mode (2ch) the front speaker levels specify the volume of the 2 channels.
In the playback section you might also see mixers which specify input device... here it will stream the input from that device to the output device. In the same playback section and in some devices you will be able to set the mode of the recording for the input jack on your sound card (line in or mic in).
In case there's a lot of disturbance in the output audio, turn off a few reluctant channels.
Again depending on the audio device, you might be able to configure the output of the digital SPDIF interface, this interface has it's own separate PCM. S/PDIF will also contain an optional called S/PDIF Playback Source... here you gotta set from where will the audio be streamed from.
In the capture section, you'll find 'mux' which acts as a sort of amplification to the input.
You'll see a no. of 'input source'; each mark an input subdevice. You may change their value to mark an input port like mic, front mic, line in etc...
The various capture channels set the volume for the various input source.
ALSA maintains a table of possible sample rate and sampling formats the device takes. Not all possible combinations can be taken in by the card.
ALSA provides plugins via which you can access the device; a device has to be named this way –
:,,
Unlike with OSS where you can send raw PCM directly to a device which was spawned in /dev, with alsa, these devices are still there but sending streams to them is not that simple and controlling them is a different challenge. These ALSA devices are present in /dev/snd. Control* mark control of the device/subdevice and pcm* mark the device to which PCM has to be sent or PCM has to be received from. Alsa-lib is a set of libraries which make the task easy. These plugins are provided by alsa-lib.
Some sample plugins –
hw – for raw device access for both playback and recording. The stream sent to the device should be something accepted by the sound card (suitable format/sample rate).
plughw – This's transparent conversions between various wave sample format and sample rate to something which can be accepted by the sound card.
You may omit one or both of,. ALSA will be on it's instincts to select one.
The plugin with the hardware is also called the device. Thus “:,,” will also be called a device.
Some plugins take in these devices (i.e. plugin in a plugin) e.g. –
tee:\'hw:0,0\',\'hw:0,3\',raw pro_logic_ii_the_other_side_44khz.wav
The tee plugin which'll work like tee – stream to 2 device off which one device may or may not be a file. raw pro_logic_ii_the_other_side_44khz.wav is the input file.
Similarly there's a file plugin.
ALSA gives these cards and devices numbers starting from 0.
There are also subdevices which are a part of an input or output device. There must be at least 1 subdevice. In context of output devices, a device having multiple subdevices means the hardware can do mixing, i.e. it can take multiple streams of PCM and mix them to produce a single output. This's called hardware mixing and no, multichannel is a different thing. The no. of streams a device can take depends on the no. of sub devices a device has. Using ALSA API, you can send audio to each of these subdevices simultaneously; the result will be seen in the output audio.
Subdevice in capturing means the card can take and digitize input of multiple audio streams at once.
Usually subdevice have 'modes' in which they operate. For output subdevices, it means the multichannel mode they operate in. Like 2 or 4 or 6 or 7 etc... which's activate other multichannel ports on the sound card.
For input subdevice it means from which port will the input be taken from (like line in, mic, front mic etc...)
There's a ncurses based mixer which alsa provides name alsamixer... change your volumes from there. You may use amixer which you can use to change volume levels via command.
Here you can set your base and treble settings also (it might be present depending on your sound card). Press m for mute/unmute.
In the playback section, can set the mode the soundcard is (same as subdevice mode). Options will be 2ch, 4ch, 5ch etc... Also you'll be also to select the output levels of multiple speakers in the multichannel setup. In stereo mode (2ch) the front speaker levels specify the volume of the 2 channels.
In the playback section you might also see mixers which specify input device... here it will stream the input from that device to the output device. In the same playback section and in some devices you will be able to set the mode of the recording for the input jack on your sound card (line in or mic in).
In case there's a lot of disturbance in the output audio, turn off a few reluctant channels.
Again depending on the audio device, you might be able to configure the output of the digital SPDIF interface, this interface has it's own separate PCM. S/PDIF will also contain an optional called S/PDIF Playback Source... here you gotta set from where will the audio be streamed from.
In the capture section, you'll find 'mux' which acts as a sort of amplification to the input.
You'll see a no. of 'input source'; each mark an input subdevice. You may change their value to mark an input port like mic, front mic, line in etc...
The various capture channels set the volume for the various input source.
ALSA maintains a table of possible sample rate and sampling formats the device takes. Not all possible combinations can be taken in by the card.
ALSA provides plugins via which you can access the device; a device has to be named this way –
Unlike with OSS where you can send raw PCM directly to a device which was spawned in /dev, with alsa, these devices are still there but sending streams to them is not that simple and controlling them is a different challenge. These ALSA devices are present in /dev/snd. Control* mark control of the device/subdevice and pcm* mark the device to which PCM has to be sent or PCM has to be received from. Alsa-lib is a set of libraries which make the task easy. These plugins are provided by alsa-lib.
Some sample plugins –
hw – for raw device access for both playback and recording. The stream sent to the device should be something accepted by the sound card (suitable format/sample rate).
plughw – This's transparent conversions between various wave sample format and sample rate to something which can be accepted by the sound card.
You may omit one or both of
The plugin with the hardware is also called the device. Thus “
Some plugins take in these devices (i.e. plugin in a plugin) e.g. –
tee:\'hw:0,0\',\'hw:0,3\',raw pro_logic_ii_the_other_side_44khz.wav
The tee plugin which'll work like tee – stream to 2 device off which one device may or may not be a file. raw pro_logic_ii_the_other_side_44khz.wav is the input file.
Similarly there's a file plugin.
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