This is a somewhat theoretical question that risk being downvoted.
It would, however, be of great help, if answered, e.g. in cases of UI translation/localization.
What (if any) is the substantial difference in meaning between codec and protocol?
(Communications) Protocol
A Communications protocol defines in what manner two systems can communicate. It is not an actual program.
One could loosely compare a protocol to an interface as used in e.g. Java and C#. It defines how you are able to "speak" with an object that is an instance of a class that implements said interface. An example of a protocol is the Transmission Control Protocol (TCP) which, among other things, defines the format of each message sent over a network link.
Codec (encoder/decoder)
A codec, in the most general sense, is a program that is able to convert data to and from a set of specific formats, e.g. from the format that a video file is encoded on disk to a format that is understood by the graphics stack provided by the OS/kernel. FFmpeg, for example, contains a number of codecs to be able to play a wide variety of audio and video files.
Related
I need to do some NON-STANDARD signal processing operations with an RFID-reader, so I'd like to know if it is possible to extract antenna's individual analog (actually digital samples right after ADC) input signal samples with Motorola FX7500 (if you know how this works on FX7400 or FX9500, please do tell, could be helpful). Samples would be processed in a JAVA-based host computer program.
What I've already tried:
Investigating Motorola's own RFID3 API's possibilities, it doesn't go deep enough to actually get in touch with input analog signal samples.
Using LLRP to its full extent, it doesn't allow analog signal sample access either. RFsurvey-functionality would have been helpful to some extent, but FX7500 doesn't support it either.
Accessing RFID-reader's linux terminal, trying to find the driver function(s), that could listen the input sample stream. If current input sample(s) could be extracted from the input stream, I could (in theory) make a script, that would save a few of those sample values in a txt-file in the host computer during a tag inventory round. My linux skills are kinda bad, hence I ask this question.
The only realistic way to solution seems to be via linux terminal, so if you folks have any ideas about that (where to look and what to do), please advise!
Contents of reader:
rfidadm#FX7500abcdef:/$ ls -1
apps
bin
dev
etc
home
include
lib
linuxrc
media
mnt
platform
proc
readerconfig
run
sbin
sys
tmp
usr
var
I cannot completely rule that out, but it's highly unlikely you can get the raw signal digitized; the devices you're looking at aren't really software defined radio devices, typically.
"speaking" RFID physically is a bit different from "usual" wireless communication: The reader doesn't only observe the energy transmitted from the tag, but more importantly the fluctuations of energy extracted from the near field of the reader's antenna coil. Hence, you don't actually have a baseband of RF bandpass signal, but hardware-specific modulations of transmitted (and inversely, antenna-reflected) energy. Demodulation is hence usually done in specialized hardware.
However, do not fret: It's totally possible to build a software defined RFID reader. There have been several approaches to that, but personally, I trust these based on Ettus USRPs and/or GNU Radio best. Look through the results IEEExplore gives you, eg. this search.
Most probably this is not possible with the Motorola readers. What you can do, is use one of the RFID chipsets available on the market: either the AMS RFID IC's, or the Impinj RFID IC's. As far as I know, both IC's support retrieving the digital samples that are received. They also have a development kit to test-drive the IC's.
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I'm developing a music site which will stream audio files stored in a server to users, audio files will be played through flash player placed in a webpage..
As I heard I need to use a streaming media server for streaming audio files ( like 2mb to 3mb in size).. Do I need to use one?
I found some streaming media server softwares like http://www.icecast.org - but as in their documentation, It is used for streaming radio stations and live streaming purposes, but I just need to stream audio files faster and in low size (low bandwidth) with good quality..
I heard I need to encode the audio files first and then send them to listeners and in their end audio files need to be decoded again. Is that true? How can I do that? if I need to use a special web server, where should I host my files? Any good hosting providers?
if I host audio files in a normal web server, they will use HTTP or TCP to deliver my audio files to users/ listners but I found that HTTP and TCP are not good ways to use for multi media purposes like streaming audio and video files, and they are used for delivering HTML and stuff. I found I should use RSTP or UDP for streaming audio files.. What should I use?
I know that .MP3 files has much better quality than the other formats but it also gives huge size to the audio files.. which format should I use for audio files?
Most of the best quality audio files are more than 7mb so I'm planning to convert them my self using a software so I could get low size files with some level of good quality. If I'm converting my audio files what is the good BITRATE I should use for my files?
Any known best softwares for converting audio files while keeping
quality in a good level?
Note** - I know that I will not need complex requirements at the beginning of the site but I want to know the best ways like they are using for soundcloud.com
Here´s a reply from someone who actually runs a shoutcast radio station, is an audio-technician and web-designer. Below is knowledge gathered
from over 5000 hours of up-to-date research !
6)
Audio Software ?
You need to have software that can:
Convert to other bitrates and formats
Normalize the audiovolume to a same "normalized" level for all mp3´s. (-1 dB)
Cut-off silence at beginning and/or end.
Equalize the audio so it sounds good.
Add effects, Mix...etc.
Best,most-used, very solid and FREE is "Audacity"
5)
Good bitrate ?
If the bitrate is to high your listeners on slower connections wil suffer from "bufferunderuns"
ie: hickups / short breaks in the audio cause their connection cant keep up with the (to high) speed.
If its to low then the quality is no good.
Best choice is 128 kb/s it sounds good and wont cause underruns for most.
Best format is Mp3 since its the format that can be handled by most players and shoutcast-providers.
Using above your average filesize for a 4 Min track will be around 4 Mb.
Since Mp3 # 128kb/s is the most popular you will get the best price/quality-deal
from a shoutcast server provider .
5b)
Audio tagging ?
You did forget that one.
You need to make sure to have your audio-files "Tagged" ie: what is displayed in the
players as "Artist - Title" information is not taken from the filename..but instead from the (iD1/iD3) "Tag"
Best, most used, very solid and FREE software is: "mp3tag"
it can do "Bulk" also (a 1000 mp3´s at once)
http://www.mp3tag.de/en/
4)
Codec ?
You upload your files to a server in the format described above "Mp3 # 128 kb/s"
since its the most used format all players can play it.
Make sure you upload in the same format (above) as the output of the server
this will keep a (important) low processor-load on your server (it wont need to convert).
A Shoutcast-server (or other streamserver) will take take your separate mp3´s and convert them
into one single realtime stream, it will create multiple streams to multiple listeners (100´s).
It will also provide you with statistics (nr of listeners,from where,now playing,played before)
A listener can play it 2 ways:
a-From a embedded player embedded on your website.
b-Or by clicking a link on your websit which will open your stream in any (standalone) player
your visitor has installed ( Winamp, WindowsMediaPlayer, Realplayer, Quicktime, iTunes...etc)
A standalone will give best quality because it will have more/better audiocontrols (equalizer...etc)
Best practice is to offer BOTH a embedded player and a simple clickable link.
check out at least 20 radio-station-websites (both professional and amateurs)
to see how they do it.
Best , and free embedded-player right now is "jPlayer"
because its dual-mode (HTML5 / Flash) so ALL BROWSERS and ALL MOBILES will play it.
and its very well supported with a forum,tutorials...etc
http://www.jplayer.org
2)
Hosting providers ?
Google for "Shoutcast streaming" or "Shoutcast server"
compare 20 of them for best price / quality...research them again using Google.
They will have special shoutcast software (webbased) such as "Centova"
you control it from any browser, you can stream live to it...or create playlists that play unattended from the server while you sleep ("autodj")
You can create multiple playlists such that they will play at certain times/days/random...etc.
You could create your whole station based on autodj playlists only
like that you will not have to worry about your own upload-connection interrupting
and you can shutoff your own pc.
For autodj you want a shoutcast service with at least 5 Gb storage (mp3´s)
that will give you around 3 to 4 days music without repeats...using the playlists in a clever way
and taking into account that listeners will on average listen between 30 mins and 2 hours at certain times,..you can make sure that they will not hear the same tracks all the time.
If you insist to do "live" (realtime) broadcast (streaming) from your OWN computer (directly or via a stream-server-provider then most used software is "Sam broadcaster"
That is it...start with a good Shoutcast server provider, then built your website and create a clickable link to the stream, after that you do the embedded player.
To begin, let me clarify my understanding of your needs. Please add a comment and clarify in your question if these are wrong:
You intend to build a site that will play audio
Audio will not be one continuous stream, but will be made up of individual files
Your audio will generally be music
Now, on to your questions:
(1) As I heard I need to use a streaming media server for streaming audio files ( like 2mb to 3mb in size).. Do I need to use one?
(3A) if I host audio files in a normal web server, they will use HTTP or TCP to deliver my audio files to users/ listners but I found that HTTP and TCP are not good ways to use for multi media purposes like streaming audio and video files, and they are used for delivering HTML and stuff.
Nonsense. Streaming media servers, such as SHOUTcast/Icecast, are actually just HTTP servers that send content as it comes in from an encoder. The client doesn't know the difference between it and HTTP. Metadata is interleaved into the content stream at the client's request (made with a special request header), but it is still compatible with HTTP.
HTTP is a protocol that is good for transferring any type of content. Ever download something from a website? That would have been with HTTP.
If it's good enough for YouTube, Sound Cloud, Pandora, and just about everyone else, it's probably good enough for you as well, 'eh?
(3B) I found I should use RSTP or UDP for streaming audio files.. What should I use?
TCP is an underlying network protocol that ensures reliable transmission. Packets are received in the proper order, and are acknowledged so that any lost packets can be re-transmitted. There is some overhead with this. The reason UDP is sometimes used is that it provides lower latency at the cost of being unreliable. This is fine for telephony communications, but is pointless for media that is not time sensitive, such as a bunch of audio files coming from a server. In fact, if you get a few too many corrupt packets, your audio player will often simply stop decoding the file, and would need to be restarted.
RTSP is way overkill for your needs. It supports a bunch of stuff for media control, variying bitrate on the fly, etc. This is not appropriate for your situation. Perhaps if you were streaming live video, or lengthy content, this would be more appropriate.
(2) I heard I need to encode the audio files first and then send them to listeners and in their end audio files need to be decoded again. Is that true? How can I do that? if I need to use a special web server, where should I host my files? Any good hosting providers?
You need to pick a codec for encoding audio that the client supports. I assume you will be using HTML5 with a Flash fallback. Unfortunately, there is no codec available that is universally supported. See the chart here: http://html5doctor.com/html5-audio-the-state-of-play/#support
(4) I know that .MP3 files has much better quality than the other formats but it also gives huge size to the audio files.. which format should I use for audio files?
Check your assumptions at the door, you are very wrong here. Keep in mind that the raw PCM data is often 8 times larger than MP3 (depending on chosen bitrate of course). In any case, you will want to encode to AAC, MP3, and Vorbis for widest client compatibility. aacPlus is an extension of AAC and is generally considered the standard for decent quality audio at relatively low bitrates. A 128kbit stream in AAC will sound better than a 128kbit stream in MP3.
(5) Most of the best quality audio files are more than 7mb so I'm planning to convert them my self using a software so I could get low size files with some level of good quality. If I'm converting my audio files what is the good BITRATE I should use for my files?
This question is very subjective. Personally, as a musician and audiophile, I prefer to hear stuff in its original quality. I use FLAC for compressing my music library, as the quality is lossless. For your needs, this will take up way too much bandwidth. Most folks don't know the difference between a 128kbit MP3 and the original. Many "premium" internet radio stations offer 128kbit aacPlus and 256kbit MP3. Pandora offers 96kbit MP3 for regular users, and 192kbit MP3 for premium users. Experiment, and pick a set of bitrates that work well for you and users.
Always keep the original around. It doesn't have to be on your servers, but you need it. If you re-compress a file that was already lossy compressed, then you are losing additional quality. If you make 3 compressed versions of one source, make sure you're doing so from the original source.
(6) Any known best softwares for converting audio files while keeping quality in a good level?
If it is legal for you to use, take a look at FFMPEG. It can handle just about any codec you can think of. As a word of caution though, do look into it to make sure you are paying all of the license fees necessary. Some of the codecs contained within are patented. I'm not a lawyer, and have yet to be able to figure out the legalities of using them on a commercial site. All I know is that it is heavily debated.
I've been using http://www.yagosta.com for years for a music company client. Free service and SSssooooo easy. Requires NO tech knowledge. I haven't updated this site in several years but you can see what it looks like at the following link. They probably have plenty of new designs which you can customize too. Perfectly adequate for most requirements.
http://www.bluedotmusic.net/selector01.html
i come accross a half duplex ASCII protocol that uses following message format again and again:
[STX][dev.addr.][sequence number][message/commands...][ETX][checksum]
not only in industrial rs485 devices, but also in this consumer device:
www.kaleidescape.com/go/control-protocol
so if this protocol is so widely adopted, why can't i find informations about it.. what is it called? how does one implement such a protocol...
i wanna use this kind of protocol with my own uC projects...
i definitely like it's conceptual simplicity, compared to other protocols..
i like this Sequence Number/Repeat Flag thing...
The sequence number is a single byte that conveys both a sequence number (legal values: 0 to 7) and a bit-flag indicating that the command block is being repeated due to a communications breakdown. The sequence number is used as an identity stamp for each command block
this question is the only valuable information i could find...
How do you design a serial command protocol for an embedded system?
is there a book about the design and implemention of ASCII protocols for uC use?
I'm planing to build a simple audio interface. For that I just want to know in which format the ASIO drivers deliver data to a program usually? I couldn't figure that out of the specifications or find that somewhere else. I don't want to write an own driver, I just want to deliver my data in the same format.
I've been doing some ASIO development, and from testing on 7 systems with 10 different soundcards (internal and external), most of them use ASIOSTInt32LSB and some use ASIOSTInt16LSB for output. With these two formats implemented, I've yet to see a soundcard that uses anything else.
Of course this is just plain old trial and error and not an exact approach by any means.
We have some raw voice audio that we need to distribute over the internet. We need decent quality, but it doesn't need to be of musical quality. Our main concern is usability by the consumer (i.e. what and where they can play it) and size of the download. My experience has shown that mp3s do not produce the best compression numbers for voice audio, but I am at a loss for what the best alternatives are. Ultimately we would like to automate the conversion process to allow the consumer to choose the quality vs. size level that they would like.
You should give Opus a try. Example compression command line:
ffmpeg -i x.wav -b:a 32k x.opus
Start here.
As you rightly point out, voice compression is different from general audio compression. You'll find many codecs dedicated to telephony applications, ranging from PCM and ADPCM through later packet based encodings such as CELP used on GSM cellular networks.
Still, VOIP voice encoding is slightly different from that due to the medium used. you can find a good, free (unencumbered and open source (BSD)) library for speech encoding/decoding in the Speex software library.
Again, which you choose depends on the speech you're encoding and the medium it's being transmitted over. Also note that many libraries have several algorithms they can use depending on the circumstances, and some will even switch on the fly based on conditions of the sound and network.
To get more help, narrow your question down.
-Adam
The most frequently used compression formats used in live voice audio (like VoIP telephony) are μ-Law (mu-Law/u-Law is used in the US) and a-Law (used in Europe, etc.) which, unlike Uncompressed PCM, don't support as wide of a frequency range (a smaller range of possible values ignores sounds outside of the necessary spectrum and requires less space to store).
For usability sake it is easiest to use mpeg compressions (mp2/3/4) for streaming to standard media players as the algorithms are readily available and typically quite fast and almost all media players should support it, but for voice you might try to specify a lower bitrate or do your conversion from a lower quality file in the first place (WAV can be at several sampling rates and voice requires a much lower sampling rate than music or effects, it's basically like frame-per-second on video). Alternatively you can use Real Media, WMA or other proprietary formats, but this would limit usability since the users would require specific third party software for playback, though WMA has an excellent compression ratio as well as compression options specific to voice audio.
Assuming your users will be running Windows, there is a WMA speech compression codec that you can use with the Windows Media Encoder SDK. Failing that, you can use ACM to use something like G723/G728, ADPCM, mu-law or a-law, some of which are installed as standard on Windows XP & above. These can be packaged inside WAV files. You'll need to experiment a little to find the right bitrate/quality (probably don't bother with mu-law or a-law). With voice data you can get away with quite low sample rates - e.g. 16000 or 8000, as there isn't much above 4Khz in the human spoken voice.
I think AMR is one of the best speech codecs. I was using it about a year ago and I remember that quality was very good and size levels were rather small.
One drawback, especially in your case is that, as far as I know, it isn't supported by wide range of media players. QuickTime and RealPlayer are two which I know to play .amr files.
Try speex ... unencumbered by patents, good performance both sizewise and CPU-wise. I've been having good luck using it on iPhone.