logic pro x joint stereo vs normal Archives – 10 (Ten) Crack Software Collection

 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
Jjoint, at low bitrates, audio codecs had the choice of reducing the frequency range http://replace.me/8406.txt downsampling see the MP3 FAQ 7 or introducing unpleasant audio artifacts. To conduct business and deliver products and services, Pearson collects and uses ссылка на продолжение information in several ways in connection with this site, including:. AC3 ] Secondly, when the audio is encoded, the joint stereo information must be preserved.
 
 

[UPDATED] Audio FAQ – Doom9’s Forum – Best Free Plugins for Logic Pro X

 

Was the singer hitting some of the notes sharp? Flex Pitch see Figure 5 analyzes the exact pitch curve of each note and lets you adjust not only its tuning, but also the way the pitch drifts at the beginning and end of the note, or the amount of vibrato.

Figure 5 Flex Pitch editing. In many modern productions, drums are the backbone of the song. They are normally recorded first, so that they can be used as a time reference when recording the other instruments. However recording real drums requires skill, patience, a large room, and multiple microphones. And depending on the style, you can choose between the virtual Drum Kit Designer plug-in for acoustic performances, or Drum Machine Designer for electronic music.

Drummer works much like a human drummer, except better. You give the virtual drummer a set of broad instructions, regarding your song and the different sections, and then you let him play. Unlike with MIDI programming where you have complete control and responsibility over each individual note, the drummer brings his own musicianship to the plate.

You may tell him how many fills you want, or how complex you want his fills to be, but the drummer decides which exact notes to play.

Figure 6 The Drummer editor. Because you can use only one set of instruction for one Drummer region, you will need to create new Drummer regions every time you need the drummer to change his playing for example play louder and more complex, or switch from the hi-hat to the toms. The Arrangement global track can help you lay out the song structure elegantly, and the Drummer track can then be automatically populated with one region for each section.

In January , Logic Pro X version The multi-touch screen of your iPad can become a control surface that helps you mix or even play instruments using a variety of controls. Figure 7 Logic Remote App. Logic Remote can be used for functional applications. Say you are recording your voice from a record booth. You make a mistake and want to quickly restart the recording.

You can use the iPad to relocate the playhead and start a new recording without having to get out of the booth. The iPad app can also be used for plain fun. Logic Pro X offers a dizzying array of tools that are meant to help you do just that with speed and ease, your ideas are still fresh when you capture them, and the results are on par with the professional music productions that inspired you in the first place. I would like to receive exclusive offers and hear about products from Peachpit and its family of brands.

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Why Logic Pro X Is the Right Choice for Professional Music Production | | Peachpit

 

How can I achieve good quality results with MP3? For movie soundtracks kbps and above will usually have acceptable quality. To increase quality, use VBR and joint stereo. The recommended commandline for transparent quality with the Lame encoder is -V2 which can be used in BeSweet -V settings require lame 3. The recommended version is lame 3.

This preset will result in transparency with the vast majority of samples, and usually a bitrate of kbps for movie soundtracks. What version of Lame is included with BeSweet? The latest versions of BeSweet now include Lame 3. Recent full packages of belight include lame 3. As mentioned before, any lame encoder version 3. Most lame users are using the latest stable version or 3. How can I fix this? Ignore it or use a good non-dshow player such as winamp. If the length still appears to be wrong, use BeSplit, with the following command : Code:.

Last edited by tebasuna51; 26th September at What is OGG? What about Vorbis? What’s the difference? It have better performance and compatibility than Vorbis. See also both encoders in hydrogenaudio wiki: Opus and Oggenc ] Vorbis is an open-source general purpose lossy encoded audio compression format developed to be free and completely unpatented.

You may have also heard of OGM, which is a modification of OGG to support video, other types of audio, chapters, subtitles, and more. How can I encode to Vorbis? Also the Opus tools , and here Opus 1. And ffmpeg support also Vorbis and Opus codecs. The rest of links are oldies. As usual Doom9 has an excellent guide to get you started. A good start point for quality q settings is around q2 and this will provide you with a file that averages 96kbps.

Unlike MP3, when discussing Vorbis, we dont mention bitrates but discuss in terms of -q modes quality. As vorbis is an open-source audio compression format, there are many encoders available, and although they are often referred to as ‘forks’, they are fully compliant Vorbis encoders. A list of the most common encoders follows: 1 Xiph 1. Download link. Aoyumi’s earlier tunings are now incorporated into the official libvorbis 1.

Homepage , libvorbis dlls. There are many more tunings available, the best source for your Vorbis software needs is usually to check out the Vorbis page at Rarewares. Note that for some compiles you’ll also need libmmd.

Get them both from RareWares. Which encoder version should I use? What are the best settings? Vorbis 1. AoTuV versions perform consistently superior in vorbis listening tests. Combined versions are available as well. In general, the -q settings are excellent VBR presets and usually don’t need additional parameters. For more information see Hydrogenaudio’s recommended Vorbis encoders and settings.

Is it possible to transcode to 5. Yes, Vorbis supports multichannel audio. Correct channel coupling has yet to be implemented as of Vorbis 1. Will Vorbis in AVI ever work? Vorbis in AVI will probably never work due to the complexity of Vorbis streams and the difficulty of correctly synching such a stream in AVI.

However, AVI works with many other things that it’s not supposed to so who knows. Is it possible to pass 5. The vorbis tools for encoding and decoding ogg are part of most modern linux distributions, check your distributions’ repositories for binary packages. As always debian users can get plenty of vorbis apps from www. You can also grab and build the source from vorbis.

Vorbis encoding is also supported in VLC and several other encoding tools. Last edited by tebasuna51; 29th September at What is AC3? Dolby Digital? AC3 is most commonly used in 5. AC3 has great popularity, and is used in LaserDisc and DVD formats at kbps and also quite commonly in cinema, usually at kbps.

Are there any freeware encoders? Of the several commercial encoders available, all known commercial encoders are based on the Dolby software encoder. The only free encoder is part of the ffmpeg project and is generally considered to be not as high quality as a commercial encoder using the Dolby libraries. However, incompatibilities with AC3 specs should be solved now, and using BeSweet 1.

FFmpeg, a native linux application, supports AC3 encoding via libavcodec, as does mplayer, vlc, and other libavcodec-based apps. An AC3 transcoding example commandline is: Code:. Attached Files AC3Util.

Last edited by tebasuna51; 4th October at AC3 Playback Questions 7. Which software do I need to install for AC3 playback? Download Valex’s AC3filter audio filters and install.

This filter is highly recommended over any other AC3 capable filter, it’s very feature rich and best of all it’s open source. Please read over his excellent documentation for a full understanding of all the capabilities of the AC3filter. Valex frequently updates the package so you might check Doom9’s news or the AC3filter page often. Secondly, you will need to use your favorite media player to playback the media container ie. With LAV Filters or a correct player the rest of questions in this post are oldies.

How do I determine if the filters are registered and installed correctly? A dialog should appear and you can double click the Direct Show category. This will display all DS filters that the system has registered and are available for use.

Double click on a filter to see the path where the filter file resides on your drive ie. If the filter doesn’t show up in GraphEdit then most likely a problem with the install occurred and you should attempt an uninstall of the AC3filter and then reinstall using the supplied installer. If the filter still isn’t being loaded in your media player or GraphEdit then try manual registering. First verify where the filter physically resides by doing a search for ac3filter.

At this point we can now register the filter by typing regsvr32 ac3filter. As an alternative you could try the Moonlight Odio Dekoda from Elecard here has much less features than Valex’s AC3filter but better than Intervideo 9. Typically this problem is due to the way your system or media player is configured. Some users have also reported they have to use the Default Direct Sound device selected for proper playback ie.

Default Direct Sound Audigy. Why does the sound sometimes stop in the middle of my movie and then come back on? Drop outs of audio over SPDIF can be caused by a poor quality power supply or you may need to purchase a surge protector to regulate current. Also, audio drop outs can be caused by a poor quality optical or coax cable. Some users have reported poor PC power supplies to be the cause as well. As a last thought, I have experienced drop outs if broken muxing tools such as NanDub was used with too tight of settings ie.

More about muxing the AC3 follows below. Check your receiver manual to see if your amp supports DRC Dynamic Range Control which can increase the volume dynamically for AC3, thus making lighter sounds play louder and compressing the overall range.

If your passing the sound over SPDIF then the majority of the filters settings will have no effect in order to pass the AC3 stream to your amp properly. Remember, the AC3 packets can not be altered or your amp would not detect the stream properly. Does the muxing software and settings affect AC3 playback? Until now we were pretty much limited to using NanDub for AVI muxing but recently it was discovered that the muxing code in NanDub is serious flawed. Please read the forum threads and the AVImux documentation for more details.

Additionally, using a modern container format such as OGM or Matroska will cut down on muxing problems, these formats generally have no problems muxing AC3. What effects might I see if the AC3 files have been muxed improperly? Another common problem might be excessive CD drive activity as mentioned above.

Watch a movie closely when the camera pans from left to right or vice versa, it should be very smooth. An easy way to compare is to watch your ripped AVI without any audio muxed and then watch the same clip after you have muxed the AC3, you should not see any difference in regards to smooth playback ie.

One last common symptom of a poorly muxed file is to hear the audio stuttering or dropping out during play, especially during high action scenes. Why doesn’t WinAmp playback AC3 files properly?

WinAmp doesn’t support AC3 files natively. Valex has once again come to our rescue and provided an excellent plugin though. The AC3 plugin can be found here. VideoLan’s VLC player version 0. Alternatively, you could first extract the 5. An example command line is: BeSplit -core -input live. If you’re still having problems troubleshooting, you can render the file in GraphEdit to get an idea on which filters might be needed. That’s how you’ll know if DirectShow isn’t able to properly render the file.

Disable or uninstall the Morgan Stream Switcher filter. This filter can cause playback problems e. The filter name is mmswitch. DirectX 8. WinXP should have 8. Any tips for Soundblaster sound cards? Also, uncheck “decoded AC3″ and uncheck ” bass redirection “. Windows At least install service pack 2 or later. Install SoundBlaster drivers v In the SB mixer, check ” digital out only “.

The Creative LiveWare software 5. If you currently own a SoundBlaster 5. Also, see the Creative FAQ for the 5. Your quality might be much better using one or the other. It has been reported that the Audigy card might suffer with AV sync problems using Wave Out as the audio device.

Issues range from bad drivers, jitter, SNR to out of voltage specifications which could damage your external amp. The intent was to show cost effective sound cards which are easy to configure for multi-channel and SPDIF playback. That info is outdated] All the popular players for linux xine, mplayer, vlc include AC3 decoding support ‘out of the box’.

Steps to using alsa are as follows: 1. Make sure your sound card is supported by checking alsa-project. Make sure your kernel has alsa support compiled in. ALSA is the default sound architecture in the 2. You will be prompted to select your soundcard. If you have no error messages, then run alsasmixer to set your default levels. On my card, this is referred to as “IEC output”. Install a player with alsa support.

The simplest way to see if mplayer has alsa support is to type ‘mplayer -ao help’ and see if alsa is one of the choices. Then add a ‘-ao alsa’ to your playback command or your mplayer. For a guide and discussion, see this post and the comments that follow.

Last edited by tebasuna51; 27th September at What is DTS? Forget any other software mentioned here. Gabest has written a filter that allows you to access the WinDVD decoder from many directshow players. Although libdts is still in the early stages of development, either of these projects should be an excellent choice for free DTS playback. Several tools can be used to mux DTS into popular containers. The only downside to using DTS files in our rips would be the file size. DTS files are typically around kpbs and up to kbps, compare this to a Dolby Digital file which are commonly kbps or kbps.

So as you can see by the bitrate difference, DTS files can be very large. However, a few very good uses come to mind for using the DTS tracks besides muxing them into a container in their raw format. In theory, transcoding a DTS file to an Ogg Vorbis or MP3 file could provide a better quality destination because of the source being so clean.

How can I transcode a DTS file? One could also use DTSdec assuming a large amount of HD space to decode to wav and then encode to any desired format. Recent versions of foobar support DTS playback and transcoding to any of fb2k’s many output formats. Thanks to Kurtnoise there is an easy guide on transcoding DTS with fb2k to a variety of formats.

The only player that currently supports this is VLC. This method will bypass your sound card all together, sound cards might alter the stream and render the audio useless so the bypass is preferable.

Keep in mind that your sound card should be capable of passing compliant Redbook audio. DTS 5. Unfortunately it seems most complaint WDM drivers are not capable of this for the moment but if your running Win9x then the VxD drivers should work. I have encoded a 5.

What about playing DTS over spdif under linux? Due to recent developments with libdts and alsa, this should now be fully working. First, follow the alsa installation steps described in Q19 of the AC3 FAQ, and make sure your spdif output is activated.

You’ll also need a player with dts support. Both VLC and mplayer can be compiled for this if you have libdts installed. Then simply add a ‘-ac hwdts’ to your mplayer commandline, or activate the spdif output option in the sound options of VLC.

What is the difference between DTS bitrates , and ? There are two kind of bitrates for DTS. When you encode a. If you encode to a. BTW, you can rename safely the.

Same differences can be aply to “Targeted BitRate” For what eac3to, when extract a DTS, show: “Patching bitdepth to 24 bits The DTS stream is not changed at all, only a metadata in the header is changed because some decoders can use that data wrongly to decode the stream to 16 bits instead 24 or better. The problem is a wrong understanding of the metadata, it is not the bitdepth of the DTS the lossy encodes don’t have bitdepth, only lossless like DTS-MA , it is the “Source PCM Resolution” used to encode that DTS, but no matter the source resolution was 16 or any we never can recover the same precission with a lossy encoder.

The lossy decoders always work internally with float samples and downsample to 16 bits int it is not recommended at all, if we decode a lossy format is to recode to other format for compatibility or to edit, and then must preserve the better bitdepth than support the next encoder or editor.

What is AAC? AAC is able to include 48 full-bandwidth up to 96 kHz audio channels in one stream plus 15 low frequency enhancement LFE, limited to Hz channels, up to 15 data streams and much more [At this moment is the more compatible audio format for stereo audio. That make to AAC the recommended lossy format to recode when we need save space or after a needed edition. LC means “low complexity” and HE means “high efficiency”.

Note that HE-AAC is meant to be used for low bitrate encoding and it is great for multichannel files smaller filesize. If you want to see comparison with other popular audio codecs at 64kbps, look here. The point of SBR is to encode at low bitrate the entire frequency range without introducing additional artifacts.

Traditionally, at low bitrates, audio codecs had the choice of reducing the frequency range and downsampling see the MP3 FAQ 7 or introducing unpleasant audio artifacts. Therefore this is why using a non-SBR decoder is a very bad idea, as your bandwidth frequency response will be severly limited. It can also be seen why this is called Spectral Band Replication, as the frequencies are replicated, and not reconstructed.

How can I encode AAC? Like you can see here the encoder CoreAudioToolbox. Last edited by tebasuna51; 25th January at Reason: Update. How can I split and join AAC files? See this thread for full information. Splitting: Code:. Credits: A big thanks goes to all the people mentioned below, their contributions to the many parts of the FAQ and their general forum help is invaluable.

Linked to Wilbert’s guide. Last edited by tebasuna51; 13th October at Thread Tools. The time now is User Name. Remember Me? Mark Forums Read. The connection can be either coax or optical fibre meaning in general it should provide less noise interference from my experience varies depending on hardware and source.

Is it possible to backup music from a concert dvd into an audio cd? Checkout BeSure from doom9’s download page. When transcoding 5. The 2. If you have elected to receive email newsletters or promotional mailings and special offers but want to unsubscribe, simply email ask peachpit. On rare occasions it is necessary to send out a strictly service related announcement.

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Page 1 of 1. Like this article? We recommend. Logic Pro X We recommend Like this article? Overview Pearson Education, Inc. Collection and Use of Information To conduct business and deliver products and services, Pearson collects and uses personal information in several ways in connection with this site, including: Questions and Inquiries For inquiries and questions, we collect the inquiry or question, together with name, contact details email address, phone number and mailing address and any other additional information voluntarily submitted to us through a Contact Us form or an email.

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