search for: 44100

...reading, kloschi cfg-part: [..] # Output codec selection (Valid selections : MP3, OggVorbis, Ogg FLAC, AAC, AAC Plus) Encode=OggVorbis # General settings (non-codec related). Note : NumberChannels = 1 for MONO, 2 for STEREO BitrateNominal=48 BitrateMin=40 BitrateMax=56 NumberChannels=2 Samplerate=44100 # Ogg Vorbis specific settings. Note: Valid settings for BitrateQuality flag are (Quality, Bitrate Management) OggQuality=0 OggBitrateQualityFlag=Quality [..] log-part: [..] 04/06/06 01:20:59 Debug(liboddcast.cpp:2475): determining left/right max... 04/06/06 01:20:59 Debug(liboddcast.cpp:3456):...

Hi I am trying to evaluate the quality of the CELT codec by using the 0.8.0 testcelt tool to encode and decode the input. I want to test different bitrates and selected the below parameters for 64, 96, 128, 196, 256kB: ./celt-0.8.0/libcelt/testcelt.exe 44100 2 256 46 $1.sw $1-64kb.sw ./celt-0.8.0/libcelt/testcelt.exe 44100 2 192 46 $1.sw $1-96kb.sw ./celt-0.8.0/libcelt/testcelt.exe 44100 2 128 46 $1.sw $1-128kb.sw ./celt-0.8.0/libcelt/testcelt.exe 44100 2 96 46 $1.sw $1-196kb.sw ./celt-0.8.0/libcelt/testcelt.exe 44100 2 64 46 $1.sw $1-256kb.sw can s...

...require to pull the MP3 stuff properly - I may be missing some vital library in my gentoo build that I can go and build and recompile the tool to get MP3 extraction to work, I'm hoping. :) localhost 13 # file *.mp3 110.mp3: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz 111.mp3: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz 56.mp3: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz 84.mp3: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz - Steve -- Sent from Steve Webb&...

Full_Name: Timo Becker Version: 2.1.0 OS: Windows XP Submission from: (NULL) (193.170.87.36) Hi! The following command causes Windows XP to freeze so that I can only pull the plug: > plot(seq(1, 44100), rnorm(44100), type="l") This does not happen with > plot(seq(1, 44100), rnorm(44100)) and > plot(seq(1,100), rnorm(100), type="l") platform i386-pc-mingw32 arch i386 os mingw32 system i386, mingw32 status major 2...

...; >> Is there any way to bring the bitrate in ices2 down below 32 kbps? > > Generally the trick for this is to downsample the audio before encoding. > You can ask ices to do this with a resample stanza in the config file: > > <resample> > <in-rate>44100</in-rate> > <out-rate>22050</out-rate> > </resample> Thanks, that does it. > >> Specifically, I have a machine with 2 versions of oggenc on it, one >> from debian woody, and one compiled from source. The woody version will >> encode...

I'm using ices2's metadata facility to update the name of a track in a vorbis stream: <input> <module>alsa</module> <param name="rate">44100</param> <param name="channels">2</param> <param name="device">hw:1,0</param> <param name="metadata">1</param> <param name="metadatafilename">/var/tmp/metadata.live</param> </input&...

...tag 3 is ALBUM=Bbbbbbbbb [2005-08-15 04:06:30] INFO metadata/metadata_thread_signal tag 4 is ORGANIZATION=Test Radio Stream [2005-08-15 04:06:30] INFO metadata/metadata_thread_signal Updating metadata [2005-08-15 04:06:30] INFO audio/resample_initialise Initialised resampler for 1 channels, from 44100 Hz to 22050 Hz [2005-08-15 04:06:30] DBUG encode/encode_clear Clearing encoder engine [2005-08-15 04:06:30] INFO encode/encode_initialise Encoder initialising in VBR mode: 1 channel(s), 22050 Hz, quality -1.010000 [2005-08-15 04:06:30] INFO audio/resample_initialise Initialised resampler for 2 c...

Hi Marc, thanks for the quick reply. Marc Gimpel wrote: > It would appear the the 'pcm2speex.read(frame, 0, frame.length)' is > blocking which means that it is waiting for data from the underlying > inputstream (i.e.AudioInputStream(t.input)). If it could read > sufficient data it would transcode it. If it recieved an EOF, it > should do some zero padding and then

...a: 8 bits left, value: 0 [wmv3 @ 0xb7e21b50]Extra data: 8 bits left, value: 0 [wmv3 @ 0xb7e21b50]Extra data: 8 bits left, value: 0 Input #0, asf, from 'VODChapter_20100323_09030000_12350000_Ch04.wmv': Duration: 03:31:55.2, start: 5.000000, bitrate: 1114 kb/s Stream #0.0: Audio: wmav2, 44100 Hz, mono, 32 kb/s Stream #0.1: Audio: wmav2, 44100 Hz, mono, 32 kb/s Stream #0.2: Audio: wmav2, 44100 Hz, mono, 32 kb/s Stream #0.3: Audio: wmav2, 44100 Hz, mono, 32 kb/s Stream #0.4: Audio: wmav2, 44100 Hz, mono, 32 kb/s Stream #0.5: Audio: wmav2, 44100 Hz, mono, 32 kb/s St...

On Sat, 7 Jun 2003, Karl Heyes wrote: > your sound card/driver does not deal with 44100hz samplerates, try 48000 > instead. Remember to change the resample tags according. Are there many cards in this category? I'dve thought that all cards would support 44100, it being the standard that it is. Your comments however suggest experience of this problem. Geoff. --- >8 ----...

Hi, It is better to replace this line in function filterbank_new: max_mel = toBARK(EXTRACT16(MULT16_16_Q15(QCONST16(.5f,15),sampling))); to max_mel = toBARK(EXTRACT16(sampling/2)); It gives the same but it seems to be faster and avoids overflow on 44100 kHz that prevents denoiser to process 44100 streams. (Yes I know that Speex should not pack 44100 streams but it does now and I use it). Best Regards, Dmitry Yakimov

...unless you're really > lucky with the rate), the cost shouldn't be too high. BTW, do you know > how often the rate gets updated? The rate is, by default, updated every 10 seconds. And sometimes (especially with the not-so-recent unreviewed patches) we do get lucky and resample from 44100 to 44100 Hz, and the next 10 seconds from 44100 to 44101 Hz. -- Alexander E. Patrakov

...e library I was trying to see if I could grab some real time audio, encode it and write to a file using arecord in conjunction with celtenc. I tried this using the below command but unfortunately the encoder just exists straight away with no errors, anybody else tried this? arecord -t wav -c 2 -r 44100 -f S16_LE | celtenc - test.oga Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Encoding 44100 Hz audio using stereo (93 bytes per packet) paul at x61laptux:~/src/celt-0.5.2/tools$ thank you, Paul.

Is There any intension to deal with mixing two or more streams encoded with "speex". Most voice wise applications (such as conference) need this feature. Oded Rephael <p><p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'speex-dev-request@xiph.org'

On Sun, 2003-11-23 at 03:51, Kerry Cox wrote: > Hmmm, I guess more is incorrect here than I thought. I changed the > resample in-rate to be 44100 like it should have have been. > But still no joy. Here is the error message as shown in the error.log > file. I'm looking things over and am not seeing my error. > It looks like the audio resample for this particular stream is good. > > [root@icecast conf]# ices /usr/local/icec...

...0 with the default packages. After install of ices and icecast & relative libraries (with no errors), I've started the icecast server. When starts ices, this is the debug: --- INFO ices-core/main ices started... INFO input-oss/oss_open_module Opened audio device /dev/dsp at 2 channel(s), 44100 Hz INFO input-oss/oss_open_module Started metadata update thread INFO signals/signal_usr1_handler Metadata update requested INFO metadata/metadata_thread_signal tag 1 is TITLE=Live Radio Stations on the net. INFO metadata/metadata_thread_signal tag 2 is ARTIST=xxx INFO metadata/metadata_thread...

Hi all, I am using paperclip plugin to upload mp3''s. Before I save the mp3 I would like to convert it to a smaller size. I am using ffmpeg library and in my Track model I am calling: before_save :convert_mp3 def convert_mp3 system("ffmpeg -i #{mp3.to_file.path} -vn -ar 44100 -ac 2 -ab 64 -f mp3 #{mp3.to_file.path}") end But this fails. Am I missing something? Thanks Pete -- Posted via http://www.ruby-forum.com/.

Is there any way to bring the bitrate in ices2 down below 32 kbps? I'm setting up a demo for someone of how to use linux to do net radio broadcasting. The setup I'm thinking of is to use ardour plus jack to mix two (maybe more) input sources (live audio and recorded tracks/programmes), then send the mixed audio to ices2 for streaming to icecast2, using the jackified version of ices2. This

Is there any way to bring the bitrate in ices2 down below 32 kbps? I'm setting up a demo for someone of how to use linux to do net radio broadcasting. The setup I'm thinking of is to use ardour plus jack to mix two (maybe more) input sources (live audio and recorded tracks/programmes), then send the mixed audio to ices2 for streaming to icecast2, using the jackified version of ices2. This

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