similar to: Standard encoding rates?

On Tue, Apr 05, 2005 at 08:26:45AM -0700, Ralph Giles wrote: > AM radio is lower quality (mono) but I don't know > what the digital equivalent would be. Just a minor nit-pick: AM radio can be stereo. However its use is almost nonexistent. See <http://users.hfx.eastlink.ca/~amstereo/amstereo.htm> for more information. > Telephone is nominally 8 kHz mono > (i.e. really bad)

Hi list, I have an 44100 Hz file with human voice, stereo with 16Bit. When convertig this to 8 kHz, mono I loose a lot of quality and have some ground noise. I tried several sox options but without success. Can somebody help.... best regards Thomas

On Tue, Apr 05, 2005 at 02:26:03AM -0500, Hal Vaughan wrote: > Is there a list somewhere of "standard" encoding rates? I know, for example, > CDs are encoded at 44100, as is a lot of digital sound, but I've seen > programs that specify different levels of quality (like radio, phone, tape, > CD) and I'd like to know if there are some encoding rates that are

Hi, Can someone please comment about the below issue [root at host0040 kaushal]# file obd-demo.mp3 obd-demo.mp3: MPEG ADTS, layer III, v1, 256 kBits, 44.1 kHz, Monaural [root at host0040 kaushal]# sox obd-demo.mp3 -e stat sox: Failed reading obd-demo.mp3: Do not understand format type: mp3 [root at host0040 kaushal]# sox -V obd-demo.mp3 -r 8000 -c 1 -t ul -w vm-intro.ulaw sox: Failed reading

One thing that has always bothered me about the ogg format is the distortion of high frequency sounds - even at data rates as high 128 and 160 kbps. I find the best way around this is to resample the wav file to 48 kHz (using SoundForge 6.0) before encoding (using CDex) to ogg. It takes a while, and adds a lot of extra wear and tear on my drive, but what a difference! The result is an 80k ogg file

Hi everyone, I recently began using libspeex 1.2 Beta 1 on Windows using MS Visual C++. I have gotten a decoder and an encoder to work fine from the excellent sample code posted at the website. But I face a problem. I am working on using Speex in a program to play and create audio books encoded using Speex (currently testing it only; for these tests, I do not use Ogg to save the encoded

At present my stream is at 11.025 kHz and I want it to be at 44.1 kHz. Input is coming from line-in on my sound blaster card under linux (RH 9) using the sb driver. I presume that it is icecast that sets the sample rate on the dsp in the card, though when I change the settings in icecast.xml and darkice.cfg as show below the stream becomes choppy; or rather the sampling doesn't seem to

In this topic on Hydrogen Audio(http://www.hydrogenaud.io/forums/index.php?showtopic=105586) someone asked a question about the sample rates that FLAC supports for ReplayGain. The outcome was that the current documentation of MetaFLAC is outdated since Commit http://git.xiph.org/?p=flac.git;a=commit;h=0554a4aee6966bc5b251364753ef85de72dfab19 because as of 1.3.0 FLAC supports Replaygain with many

On Wed, 10 Nov 2004 16:08:21 -0800 (PST) Josh Coalson <xflac@yahoo.com> wrote: > > Is there someway of figuring out if a sample rate is valid? > > that's the right way. But it doesn't tell me that the sample rate is invalid it tells me FLAC__SEEKABLE_STREAM_ENCODER_STREAM_ENCODER_ERROR or FLAC__STREAM_ENCODER_NOT_STREAMABLE. > the reason it's being rejected is

Hi all, I've got a 400MHz XScale-PXA255 board, and I want to stream voice from it over a network connection at 28.8baud. This calls for a capable voice encoder which can encode at about 24kbps. I was damn happy when I found Speex and said goodbye to MP3 :) However, i'm still a long way from realtime encoding using speexenc, is this possible? Using the fixed point math option in

Hi all, I'm trying to use the FLAC C libraries to encode audio. I'm doing something like: FLAC__seekable_stream_encoder_set_channels(pflac->fse, 1); FLAC__seekable_stream_encoder_set_sample_rate(pflac->fse, 11025); FLAC__seekable_stream_encoder_set_bits_per_sample(pflac->fse, 8); if ((bps = FLAC__seekable_stream_encoder_init(pflac->fse)) !=

________________________________ Hi David, Thank you very much for your reply. Since I need to resample the audio in the program itself, I decided to try out the resampling API in speex. But now, I have another problem. The resampled sound is very much distorted and clicks appear quite often. (I have attached the source code I used for testing it below). The test data I had was a file sampled

Hi, I'd like to know the reasons why CELT supports only signals with sampling frequency in the range of 32-96 kHz. In effect, it can clearly outperform speex at high bitrates, and has potential to be used in high quality voice communications even for 11, 16 and 22 kHz speech signals. It could also compete with SILK codec (to be soon released by Skype). See this page for more specifications

Hi, recently I made some test with the opus tools (enc and dec) and I'm very (and positively) surprised about the resultant quality. But the only think that I miss is the ability to change the low pass filter frequency via "--lowpass" option or similar. For example at a quality or 96 kbps the cut off of the filter starts at 16Khz and is completely cut at 20 Khz. But in case of

----- Original Message ----- From: "Alexander Chemeris" <Alexander.Chemeris at sipez.com> To: "Vincent Burel" <vincent.burel at vb-audio.com> Cc: "Conrad Parker" <conrad at metadecks.org>; <speex-dev at xiph.org>; "Jean-Marc Valin" <jean-marc.valin at usherbrooke.ca> Sent: Thursday, November 13, 2008 11:31 PM Subject: Re:

Jean-Mark sarkasm. Jean-Markasm. (Bonus points for providing an actual noisy WAV! ^_^) On 30/10/2017 20:28, Jean-Marc Valin wrote: Hi, Before I comment on the graphics you posted to visualize the difference between two audio signals, I'd like to ask for your help in evaluating my JPEG encoder. I've encoded an image with JPEG and then computed the difference with the original. I then

Hi, Is there any difference between decoding of CBR or VBR streams in realtime in terms of CPU usage/cost? Are sample rates other than 44100Hz supported by Vorbis officially ? If it is supported, is it efficient to use them or is Vorbis specifically (or at the moment) optimized for 44100Hz ? Thanks in advance. --- Mete BALCI Senior Game Programmer Momentum DMT Istanbul, TURKEY

the Vorbis FAQ says: "mid to high quality (8kHz-48.0kHz, 16+ bit, polyphonic) audio and music at fixed and variable bitrates from 16 to 128 kbps/channel." What is the difference between Hz and bitrate? Doesn't MP3 support higher bitrates? Pointers for more reading are welcome.

Will oggenc have a lowpass switch? I would prefer to lowpass at 15-16khz at -q3 for use with FM broadcasting. The additional frequencies would be chopped off anyway by the transmiters hardware lowpass filter so the encoder could use the addition bits for other purposes. It could be enforced that the lowpass can only be reduced and not increased from the default. This would stop people

Hello, I was wondering about the lowpass filter applied at the different quality levels. It seems that the 16kHz cut-off is still there at quality 3. I didn't really abx it, so maybe my mind plays tricks on me. :) Please can someone enlighten me what is used for -q0, -q1 and so on. (lame tells me whats used when encoding tracks) I know I shouldn't judge by bandwidth but I would like to