WAV, A - law, 8 kHz, 8 bit, mono

Similar Questions

• Clip + no reading FLAC 96 kHz/24 bit?

  I couldn't find any where any technical...

  I have some files of professionals/audiofile music format FLAC 24 bit 96 kHz.

  He is professional MFSL vinyl LPs of recordings and original recording of my synth, Korg Oasys PCI.

  None has no DRM.

  When I try to play them on the Sansa Clip Plus they get jumped. No error message. It jumps right through them.

  I know that the files are OK. They do not play through my media streamer when I connect the Clip + via USB to my NAS.

  Is that what I can do to play on the Clip + as installing Rockbox, using .wav files, or is a material limitation

  On this aspect, why is there no specification anywhere (bitrate, resolution, etc.

  Seriously, there should be some tech specs in the manual...

  I guess I'll have to recode these for lower quality - but WHICH one is better than the Clip + is supported in terms of flow rate and sampling frequency?

  The DAC on the Clip + is only 16 bits, so there is no advantage to the use of 24-bit. Rockbox will decode any file FLAC, but they will be converted to 44.1 k/16 for playback. The sandisk firmware does decode flac 24 bit at all. BTW, you probably already know, LPs are less 16 bit, so there is no loss of quality to convert to 16-bit.

• Audio playback within an application

  So I try to get streaming audio working within my request - and I use the Simulator to make my tests.

  I am following the guidelines provided by a number of Web sites - including a Blackberry document and try to play a wav file, hosted on a site.

  In principle, this sounds simple.  In practice it won't work for me.  Specifically, I'm tryin to play the wav file http://java.sun.com/products/java-media/mma/media/test-wav.wav and I do this by using the following code snippet:

  try {}
  String wav = "http://java.sun.com/products/java"+ "
  'media/mma/media/test-wav.wav ';
  Player p = Manager.createPlayer (wav);
  p.Realize ();
  p.prefetch ();
  p.Start ();
  } catch (IOException x) {}
  x.printStackTrace ();
  } catch (MediaException x) {}
  x.printStackTrace ();
  }

  Well, the good news is that he plays.  The bad news is that it is just a bunch of stuttering cracked that is read.  And here's a little audio snippet end easily playable in the browser (for example)

  Any ideas as to what is happening here?

  Thanks in advance!

  Jim

  This problem has been resolved.  The codec used for this audio file was not supported by the Simulator (odd really - seen it's just PCM 11,025 kHZ 16 bit mono.)

  It was only a sample for the test... It would simply play in any old J2ME environment, I guess.

  My real data are in MP3 format - and I managed to get this working.

  Although you might wish to consider why audio PCM fails

  BTW the Simulator was in JDE 3.5 8 BB 8330

  Jim

• You will need to set audio track in CC animate...

  Quick question... is it hard to bring audio track in animate CC. do this in an hour or two.

  It is not difficult, but there are a lot of problems that you may encounter depending on what you intend to do with the sound.

  No matter what, make the sound 44.1 KHz, 16-bit (mono or stereo). Other bit depths are likely to fail to import, and other sampling rates can lead to synchronization problems.

  If it's an AS3 FLA, especially if you plan to synchronize animation to audio, store the file as a WAV file.

  If it's HTML5 Canvas, make it be MP3 before importing the file.

• import of MP3s with 32 000 rate of the samples?

  I am trying to import my mp3 convert files in Flash Pro CC and get the message that it cannot be done. The files have a rate of 32 samples, 000 kHz. 32 Kbps (mono) 64 Kbit/s, stereo. The voice if she ' doesn't only not to be of this quality.

  If the settings are wrong, I would like to know how to convert all my files without loss of quality and without becoming larger. Any know?

  The file size depends on the data rate, no sampling frequency. If you have the original high quality sounds, you must code them to 44.1 kHz, 16 bit, mono 64 Kbps. If you have not, then convert your current files in this format. The size of the file remains the same.

  I use a Converter audio simple app to make lots of sounds, but you can also use Adobe Media Encoder.

• Presets - 44000 sample/s audio?

  Hello! I know in Pro, you can customize your own presets, but Pro is too 'hard' to work with, elements is a lot easier for me. I understand that you can't create your own custom project settings, but as I go through them, they have all audio 48000. I need 44000.

  Is it possible to do?

  I try 48000, but it slows down just reading.

  What can I do? Any suggestions? Thank you.

  Kind of dumb question, my apologies!

  Under Edit > project settings > general, I can not change the audio

  For the DVD, the output format must be either PCM/WAV, AC3 DD @ 48 kHz 16-bit.

  Now, if we start with the Audio with a 44 KHz sampling frequency, PrE will convert to 48 KHz sampling frequency, when in line. Nothing to do at that time for you.

  For a little background, the 44.1 KHz is the sampling frequency of CD, but 48 kHz was chosen for DVD and SACD (Super Audio CD).

  Another aspect of the process in accordance with the standard and that the user sees same almost never, is that the Audio is converted to 32-bit, floating point, for precise Assembly.

  If you go on a DVD, then you want 48 KHz 16-bit Audio. If you need other formats, then you could squeeze a little more space for the size of the file, by dropping the sampling frequency Audio up to 44.1 KHz CD standard, but not by much. Personally, I wouldn't worry on this subject. Now, if you absolutely need the 44.1 KHz, most of the formats will allow you to choose which.

  Good luck

  Hunt

  PS - For DVD, you can't choose 44.1 KHz, because is not in the standard.

• Codecs to generate AVI with Premiere Pro CS4 files?

  I would like some feedback on which codecs you think are the best to generate AVI files. I tend to use Microsoft Video 9 or Indeo 5.10 for most exporters of my AVI, but I was wondering if there are traps with codecs that I don't know?

  My priority is the video quality, but I also care about how many compression, it is (IE, reduce to a minimum the amount of space is consumed on the hard drive), and I want to minimize the time required to export a media file to avi.  Are there other factors should I consider in deciding which codec to use?

  For what it's worth, I am running CS4 on a Windows XP computer.

  Here's what I observed in my tests...

  In the order of the smallest file avi (most compression) to the larger avi file (compression less):

  Microsoft Video 9

  Indeo 5.10

  Indeo 4.5

  Cinepak Codec (AVI is more than triple the size of the Indeo codec)

  Microsoft RLE (AVI is almost double the size of the one generated by the Cinepak)

  Microsoft video 1 (AVI is almost three times the size of that generated by the Cinepak)

  In order of time the shorter 'export' longer-term export times to a 13 second clip of test, with the codec on a maximum quality:

  Microsoft video 1

  Microsoft Video 9

  Microsoft RLE

  Indeo 5.10

  Cinepak

  Indeo 4.5

  As far as I can tell from the eye or video files, the quality that results is similar on all of them, but I'm missing something?  I must admit, I have not looked at the video and sound quality as carefully as I would like.

  Welcome to the forum,

  If the conversion program offers it (most do, but some do not), the DV - AVI (AVI specific flavor) w / PCM/WAV Audio to 48 kHz 16-bit, would be the best.

  Also, there may be something useful in this ARTICLE, even if it was written for PrElements and not PrPro.Still, perhaps a useful tip.

  I also recommend HD of Jeff Belluno to SD TUTORIAL.

  Good luck

  Hunt

• How to decrease the output volume of the .wav file to a speaker attached to the output NI 9263 module

  Hello

  I found an example of http://zone.ni.com/devzone/cda/epd/p/id/4182 and edited for my purposes. On this basis, I have 16-bit mono speech files with sampling rate of 16 kHz for a speaker attached to the output NI 9263 module. However, this forum is without volume control and the result is extremely strong. I have attached photos of my diagrams below.

  NDS read Wave File.vi returns a 1 d mono 16‑bit whole array. I multiplied this table by a command button, so I could adjust the constant which I multiply with the table 1 d as I thought that the size of integers are related to the amplitude of the signal. However, reducing the constant only reduces the quality of the word and is still very strong.

  Is it the right approach to reduce the volume? And if not, what? Thanks in advance

  Yawei7 wrote:

  I thought that the size of integers are related to the amplitude of the signal.

  It's quite right...!
  And your approach is perfect...!

• WAV FILES

  I'm working on for my Codian 4500's and MPS800s custom wav files.

  Problem I have is I create .wav and download it, but it does not work.

  For the codian, I guess that it is looking for voice_prompt_first_participant.wav of file name?

  The format that he needs is not compressed, mono, 16 kHz and 16-bit resolution.  My option of voice of windows 7 is 16 kHz, 16-bit, 1-channel, but don't say no compressed?

  I also a few tens of window, but it does not define the recording under 2 voice channel 16 bit 44100 hz (CD quality)...

  Question is what are the critical parameters and length parameters to get the wav file to work...

  I also if I register on the 10 windows machine I convert m4a to wav and if I save on windows 7 I need to convert wma to wav...

  Any idea of a more effective way to make wav files?

  VR

  Chet

  For a free audio tool, it's more than perfect:

  http://sourceforge.NET/projects/Audacity/

  This has worked very well for all my audio use case and is available for various operating systems.

• Distortion after you convert the wav to mp3 file

  Hello

  I work with a sampling frequency .wav file 8 k, 16-bit, mono a Skype conference call that was recorded by the computer. Wav is ACM Microsoft file: Microsoft ADPCM. The audio quality of the original is not optimal, as you can imagine, but when I do my changes and convert to mp3 that sounds horrible. I don't know the technical term for it, but it sounds muffled and distorted - the effect you get when cut you your hand in front of the mouth and the move while speaking. I am at a loss for what to do. Is there a setting I need to change before conversion? I know it's a sampling rate low (I usually work with 44.1 K), which would contribute to the problem?

  Any help is greatly appreciated!

  Thank you.

  In a quick test, I was able to get significantly better results with an 8K, 16-bit source conversion to MP3 when you change the MP3 setting to 16 kHz.  In the file > save as dialogue, after selecting the MP3, click on the button 'Edit' next to 'Type of sample' and change the sampling frequency.  The file size was virtually unchanged, but the audio quality has been increased and by looking at the two files in spectral view showed a marked improvement.

  Edit: Oops, I answered by e-mail and didn't know that this has been posted on the forum Audition 3.  I suspect the results will be even better with the 3 MP3 encoder - both are by Fraunhofer.  Here were the differences I've seen when writing 8 k 16 K vs.  The resulting file is 29KO in both cases.

  

  MP3 recorded to 8K sampling frequency setting

  -------------------------------------------------------

  

  MP3 recorded at 16K sample rate setting.
  

• failure to export a wav (damaged codec?)

  I use energyXT 2.5 (on win7), which is a software of recording for a good while now without any problem, I use it to record old cassettes and records CD and finalize audio wav file. Everything is fine until around November 24, 2014, this is happening, check and back up fine, but when export to wav finishes the resulting file is also great it should be at 4.3 GB, but don't play that 4 min 01 sec (should be around 48 min) tried another record (11 altogether different recording sessions) same thing (produced a file correctly size but only about first min 11 parts), tried all methods of saving ACE, destinations files, update driver, dianosis system, hp (without help) support, energyXT support (no response), energyXT uninstalled/reinstalled and no solution, I found similar problems people have described and sounds like a corrupted audio codec can be blamed which would explain why the uninstall/reinstall didn't work. If so, what could do this? do not have change all parameters (or re - open the program) between the last successful export (nov 21) and first failure (24 November), the only reason I can think is an automatic update of a certain type screwed up something pretty bad, but I don't know (my last solution is to install on and implementing a dedicated) (, computer disconnected, restored to a version of win I KNOW works). How can I fix this before it comes to that?

  recovery system was my first idea, but recovery registry filled and emptied, and I can go back too far. I had hopes for audacity, it came with energy xt on a disc with a few other audio stuff freeware but older version would not manage the sampling frequency my audio interface to not use but improved version appears comparable if not better than energyxt, and even still it fails AND the exact same way (produces a playable track of 4 min 01 sec to 4.3 GB after the recording of the disc even at the same rate of 192 kHz and exporting) lossless WAV). something is wrong with the way windows renders 32 bit wav files float. (oh that 16-bit works but I have wasted enough of my time) so I'm going or backup I need to do anyway and restore factory image from manufacturer, which may be a bit drastic, but will for sure work or idea mentioned before where I just use a different tour according to the question if I can acquire one and how long my patience. I very much appreciate your suggestions in my digital tribulations! Thank you

• Error WAV header to import at PPCC, 2014

  So, I have 48 KHz 32 bit PS, MP3 CBR 256 Kbps files I export out of hearing to the fp 48 K / 32 bits as Uncomp. TA5

  When I try to import in PPCC, I get the error: "cannot open the file due to an error header.

  8.1 to win

  Hearing of CC 2014

  First Pro CC 2014

  16 GB of ram

  Wave files have a size limit of 4 GB by the original specification.  Hearing supports several methods of writing wave files larger than 4 GB (for example RF64).  However, these formats does not support the first at this time. You try to save as a 24-bit instead of 32 bit (float) wave files, in order to reduce the file size without affecting significantly the quality.

• Can anything convert ALAC 24-bit or higher?

  Personally, I would like to convert it to a minimum of 32-bit floating-point ALAC, but I'd settle for 24-bit integer. 16-bit is too. I heard that the ALAC has the ability to support 384 kHz, 32-bit floating point, maybe even 64-bit Floating Point (IEEE) the way WAV files can.

  Can not export or convert to 32-bit ALAC with a minimum of 44.1 kHz and 256 Kbps?

  SteveG (AudioMasters) thanks for the recommendation of Foobar2000, but it seems sketchy. Whatever it is, I did a short study in order to determine which program should provide the best quality of conversion ALAC as well as inadvertently determine best quality (in my opinion) output formats:

  Surprisingly (not really), iTunes is the best converter ALAC (free), may be replaced by Logic Pro X, nothing else. It is possible to convert to 32-bit Float ALAC, 192 kHz and maybe even of 384 kHz, but not unless the source file is already 384 kHz. For 32-bit conversion, the source must be exported from Adobe Audition (CS6 or above I think) as 32-bit integer. In iTunes, it converts the floating-point format. 44.1 kHz being more than substantial, a float of 32 bit 44.1 kHz ALAC equals perfect compressed lossless Apple available.

  Furthermore, I have determined that supplied iTunes perfect conversion of any waveform is registration, although ALAC iTunes conversions, as well as others, perhaps, reduce the peaks that exceed the threshold, but convert otherwise perfectly. without zooming on the threshold, you would never know there was a difference. This is recommended only with files that have reduced overall gain or who have already had dynamic range compression or fixed limitation.

  Better yet, Adobe Audition can export files of Point floating (IEEE) FLAC 32 bits with original sampling frequencies (so you can resume 44.1 kHz) for perfect conversion, and I mean perfect. I checked the sections that were exact matches as far down as 1 hudred-millionieme of a second.

  Perfect lossless Compression: Minimum 44.1 kHz, 32-bit floating point (IEEE), FLAC

  I edit in WAV format, export WAV to Apple to convert to the ALAC, FLAC WAV export and destroy all WAV, FLAC become my source. The routine is perfect for me.

  I can share my pages of notes of my studies over the past two weeks to determine all the data and the conclusions that I have collected with anyone.

• Add its mp who has different Khz for the video...

  Hi, I have a few video which is Format VP6 754 Kbps with audio mpeg 44. Khz 29.970 fps and a few audio which I want to add that is MP3 (mpeg) 256 Kbps 48.0 Khz

  The warnings before I start adding the sound of the video?

  Thank you

  Michael

  Michael,

  You may find that things are better in PrPro, if use you SoundBooth, Audition or even free audio editor, Audacity, to convert your mixed @ 48 KHz 16-bit PCM/WAV sources.

  PrPro can generally handle a lot of mixed Audio sources, but sometimes fall. With the MPEG Audio, one of these stumbles often is OOS.

  You can certainly try these sources and if you have any questions, go to Plan B (as described above).

  Good luck

  Hunt

• DV - 32 kHz or 48 kHz sequence?

  I tried looking for things before posting, but I've seen conflicting opinions on this.

  I've got a few SD DV tapes I capture and editing in CS4.  I think that most of them is 12 bit audio resulting in 48 kHz.  If the destination format is a SD DVD, no matter if I put the sequence as 48 kHz?  An announcement of States which, since it ended up being converted, it does not matter.

  Or is it better to determine the flow of the spring and then match the sequence? So if it's 12 bit audio choose 32 bit and if audio 16 bit 48 kHz?

  Because I am not sure that all tapes are 12-bit, if I don't lose quality by simply using a 48 kHz sequence I guess that's my preference.

  Thank you

  BJBBJB1

  Yes. In this case I choose a 48 KHz 16-bit project and let PrPro take care of things from there. In most cases, he may ingest a certain Audio files of different appearance standard and comply with them. If you are having problems, just rip the Audio and in SoundBooth, Audition or free Audacity, make a Save_As PCM/WAV 48 kHz 16-bit and than import.

  Good luck

  Hunt