search for: 22050

Hi, A question: does vorbis rc3 support 22050 Hz sample rate? rc2 used to support it, but I can't seem to make it work with rc3 :(( <p>Akos <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 'vorbis-request@xiph.o...

...Live Radio Stations on the net. INFO metadata/metadata_thread_signal tag 2 is ARTIST=xxx INFO metadata/metadata_thread_signal Updating metadata INFO audio/downmix_initialise Enabling stereo->mono downmixing INFO audio/resample_initialise Initialised resampler for 1 channels, from 44100 Hz to 22050 Hz INFO encode/encode_initialise Encoder initialising in VBR mode: 1 channel(s), 22050 Hz, quality 1.000000 INFO stream/ices_instance_stream Connected to server: viabazar.net:8000/radiokey.ogg INFO audio/resample_initialise Initialised resampler for 1 channels, from 44100 Hz to 22050 Hz DBUG en...

...n 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 mono wavs down to 16kbps, but the (later) hand comp...

...configuration with ices and darkice, but I always get an configure encoder error: [2003-03-17 16:07:17] INFO input-oss/oss_open_module Opened audio device /dev/dsp at 2 channel(s), 44100 Hz [2003-03-17 16:07:17] INFO audio/resample_initialise Initialised resampler for 2 channels, from 44100 Hz to 22050 Hz [2003-03-17 16:07:17] INFO encode/encode_initialise Encoder initialising with bitrate management: 2 channels, 22050 Hz, minimum bitrate 20000, nominal 22000, maximum 30000 [2003-03-17 16:07:17] EROR encode/encode_initialise Failed to configure managed encoding for 2 channel(s), at 22050 Hz, wi...

I need to encode a WAV to OGG but must resample to 22050 samples per second, but can't do that with oggdropXPd.exe. The application always crashes. What I do is: Start oggdropXPd.exe. Right-click on fish, choosing 'Encoding Options'. Under Advanced Options, I set Re-sample to 22050 samples per second. Click 'Accept'. Drag WAV on fish...

...Is the following code correct for compressing audio? The output I get is so extremely small, but what is more important: if I pass it through zlib, it gets at least 50% smaller! Init: int bitrate = max_bandwidth * 8; speex_bits_init(&libspeex_bits); if (sample_rate > 22050) libspeex_enc_state = speex_encoder_init(&speex_wb_mode); else libspeex_enc_state = speex_encoder_init(&speex_nb_mode); speex_encoder_ctl(libspeex_enc_state, SPEEX_SET_SAMPLING_RATE, &sample_rate); speex_encoder_ctl(libspeex_enc_st...

...elay>2</reconnectdelay> > <reconnectattempts>5</reconnectattempts> > <maxqueuelength>80</maxqueuelength> > > <downmix>1</downmix> > > <resample> > <in-rate>22050</in-rate> > <out-rate>11000</out-rate> > </resample> > > <encode> > <nominal-bitrate>16000</nominal-bitrate> > <samplerate>11000</samplerate> >...

Hi, I am testing the encoder example and I have a question about it... How can I encode a mono file in 22050 and 44100? I've changed the init values to: ret=vorbis_encode_init_vbr(&vi,1,22050,.5) ; and in the loop: // uninterleave samples for(i=0;i<bytes/2;i++) { buffer[0][i]=((readbuffer[i*2+1]<<8)|(0x00ff&(int)readbuffer[i*2]))/32768.f; } .. what happe...

I wish to transcode an AVI from 29fps to 15fps theora. I have tried many combinations. The nearest I have so far is to: 1) ffmpeg -i input.avi -f avi -vcodec rawvideo -acodec pcm_s16le -r 15 -ar 22050 -ac 1 intermediate.avi then 2) ./ffmpeg2theora-0.15.linux.bin -f avi -c1 -H 22050 -o out.ogg intermediate.avi This gives me an output I want, however, the intermediate uncompressed stage generates 36Mb/sec of data, which is impractical. So it would be better if I could send the uncompressed ou...

...gt; the net. INFO metadata/metadata_thread_signal tag 2 is ARTIST=xxx > INFO metadata/metadata_thread_signal Updating metadata > INFO audio/downmix_initialise Enabling stereo->mono downmixing > INFO audio/resample_initialise Initialised resampler for 1 channels, from > 44100 Hz to 22050 Hz INFO encode/encode_initialise Encoder initialising in > VBR mode: 1 channel(s), 22050 Hz, quality 1.000000 INFO > stream/ices_instance_stream Connected to server: > viabazar.net:8000/radiokey.ogg INFO audio/resample_initialise Initialised > resampler for 1 channels, from 44100 Hz to...

...ormal such applications, in that instead of reading a file at once, I am attempting to read a frame, and compress, then uncompress, and then playback that frame before I move on to the next frame. While initializing the device for reading the sound information, I give it the following parameters: 22050 (which is the frequency with which to take sound samples) AL_FORMAT_MONO16 (which is the OpenAL format in use) BUFFERSIZE (which is 1024, the size of the frames which I read) As for the arguments for "Compress()", pData is the ALchar pointer which contains the data. iFrameSize is the int...

...lerate currently MUST match the channels and samplerate given in the parameters to the oss input module above or the remsaple/downmix section below. --> <encode> <quality>0</quality> <samplerate>22050</samplerate> <channels>1</channels> </encode> <!-- stereo->mono downmixing, enabled by setting this to 1 --> <downmix>1</downmix> <!-- resampling. Set to the frequency...

...UM=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 channels, fro...

...o the Compress function. Could you suggest a frequency which is more suitable? Please note, that the program gives us proper playback sounds if I don't run the compression/decompression functions. IRTEZA On 9/8/06, Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > > > 22050 (which is the frequency with which to take sound samples) > > 22050 isn't recommended > > > AL_FORMAT_MONO16 (which is the OpenAL format in use) > > BUFFERSIZE (which is 1024, the size of the frames which I read) > > Where did you read that you get to choose the frame...

...from 44100. But works with VBR (quality setting) works no matter what sample rate I choose . [root@sanctuar ices]# ./ices ices-live.xml [2002-03-22 12:51:23] INFO ices-core/main ices started... [2002-03-22 12:51:23] INFO input-oss/oss_open_module Opened audio device /dev/dsp at 1 channel(s), 22050 Hz [2002-03-22 12:51:23] INFO encode/encode_initialise Encoder initialising with bitrate management: 1 channels, 22050 Hz, minimum bitrate -1, nominal 128000, maximum -1 Segmentation fault <p><p>Thanks. <p>--- >8 ---- List archives: http://www.xiph.org/archives/ icecast p...

On 2006-06-21T15:21+0100 Karl Heyes wrote: > You capture at 48000 but resample from 44100, that will cause problems. > You resample to 22050 so that will affect quality, you haven't said what > bitrate you want to target for, so resampling/downmixing may be required > anyway. I am unsure about these settings. Any particular suggestions? I don't quality with voice, just clarity. > >Can I cut the delay? > A few...

Ok. Now it will run. But when i try to play back my stream(either ogg/vorbis or mp3) it sounds like the chipmunks. Could it be that I'm broadcasting at a sample rate of 22050 in mono? This shouldn't make a difference as the source is also 22050 samplerate and mono(its the line-in plugin for winamp). I'm using Winamp 2.81 with the is the Oddcast DSP for Winamp 2.xx plugin to 'feed' the server. --- >8 ---- List archives: http://www.xiph.org/archi...

I'm using liveice and lame on another stream... ample rate of 22050 bitrate of 24000 It has worked fine for over 9 months now...except it crashes all the time because of liveice I think. >>> darkeye@tyrell.hu 8/28/02 9:15:58 AM >>> Matthew Mencel wrote: > Can the bitsPerSample be set any higher than 16? I try to set it to > 20 or 24 and...

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