Recently, I discover that my tutorial videos could be seen at 1.5x playback speed without losses in quality (they are actually better to see, as I normally speak slowly). My problem is that if I change the speed of the video when using a video editor, like Kdenlive, the audio becomes distorted and turns into a mess (higher pitch, I believe).
How could I obtain the same quality as VLC "playback fast" and Youtube "playback speed 1.5" for the audio track? I'm a layman in audio/video editing, so I'm also satisfied with partial answers, like the identification of which terms I should search for in this case.
It might be better to take your audio track and use something like Sound Forge to automatically remove silence. Just be sure to add a pad to that (built into sound forge) otherwise the speech will sound way to chopped and fast.
Aside from that, you could also use Vegas to (then) chop the video to keep pace with your new speech rate. Vegas is a video editing program that is best for this kind of down and dirty editing.
is there a scenario where we can use the Google Resonance Audio SDK not with headphones, but with real speakers (e.g. mounted in a 360° cyrcle setting)?
Or are all algorithms not working for real speaker outputs?
Thank you!
Currently, Resonance Audio is optimized for headphone playback. For example, HRTF processing is done in the Ambisonics domain, without generating (virtual) speaker signals - this is because it is a much more efficient way of generating binaural output.
However, in the Resonance Audio open source release, the Ambisonic Codec class can readily be used to decode Ambisonics to any arbitrary loudspeaker array. To use that with the rest of the Resonance Audio system, however, it would be necessary to modify/extend the audio processing graph by adding a new decoder node.
Please, feel free to add a feature request and, depending on popularity, we might consider adding that in the future!
I have built a source client using Portaudio and LAME which streams the microphone input to an Icecast server to be listened to online via the HTML5 tag. I have managed to (supposedly) get the quality of the stream to MP3 320kbps at 44.1kHz and am looking for a way to confirm this using tests and or benchmarks.
I have an indication that these stats are somewhat correct from looking at stream inspectors in software such as iTunes and VLC, but I am looking to get a more in-depth data set.
What I basically want is to be able to test how much of the original file is being lost over the stream and if or how much the quality changes depending on environmental conditions of the broadcaster or streamer.
Does anyone know of any tools, frameworks to get some hard numbers or representations of this data?
If VLC tells you the stream is 320kbit CBR, then it is.
It sounds like what you're looking for is a comparison of the actual audio content. This is highly subjective. MP3 is built to use features of how our hearing works to save bandwidth. For example, quiet sounds are masked by loud sounds. High frequencies are harder to hear and are simply rolled off.
You can compare the spectral analysis between the original PCM-sampled waveform and the MP3 decoded waveform, but this doesn't tell you how humans interpret that sound. For that, you would have to survey humans.
I'm trying to make a video tutorial, so i decided to record the speeches using a TTS online service.
I use Audacity to capture the sound, and the sound was clear !
After dinning, i wanted to finish the last speeches, but the sound wasn't the same anymore, there is a background noise(parasite) which is disturbing, i removed it with Audacity, but despite this, the voice isn't the same ...
You can see here the difference between the soundtrack of the same speech before and after the occurrence of the problem.
The codec used by the stereo mix peripheral is "IDT High Definition Codec".
Thank you.
Perhaps some cable or plug got loose? Do check for this!
If you are using really cheap gear (built-in soundcard and the likes) it might very well also be a problem of electrical interference, anything from ...
Switching on some device emitting a electro magnetic field (e.g. another monitor close by)
Repositioning electrical devices on your desk
Changes in CPU load on your computer (yes i'm serious!)
... could very well cause some kinds of noises with low-fi sound hardware.
Generally, if you need help on audio sounding wrong make sure that you provide a way to LISTEN to the files, not just a visual representation.
Also in your posted waveform graphics i can see that the latter signal is more compressed, which may point to some kind of automated levelling going on somewhere in the audio chain.
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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