search for: 48k

hi! i'm putting together a streaming relay network for the linux audio developer's conference #2 (http://www.zkm.de/lad), and i was wondering how much cpu power will be required on the master server. we need to record 2 48k stereo signals simultaneously and encode them into three streams each: 2x 54kbit/s 22k05 mono 2x 112kbit/s 48k stereo 2x 192kbit/s 48k stereo it would be very nice if the same box could handle the streaming to between 5 and 8 relay servers, so that we have only one single point of failure. ho...

...at 3:40 AM, Sean Darcy <seandarcy2 at gmail.com> wrote: > But when I run opusdec, > > opusdec 2-24-Overture_in_C_\(In_Memoriam\).opus > Decoding to 48000 Hz (2 channels) Opus files don't have a sampling rate anymore; the internal representation is most efficiently decoded to 48kHz. The metadata records the original sampling rate which make it possible to decode to a file with exactly as many samples as the original; but there would be no reason to do that when decoding for playback (for example.)

Jean-Marc Valin wrote: > OK, first thing I see is the configure options: > > ./configure --enable-valgrind --enable-sse --enable-fixed-point > --enable-epic-48k --enable-ti-c55x > > One question: "what do you think you're trying to do here?". Four of these five > options are more or less mutually exclusive! SSE is floating poing and x86 > specific, nothing to do with fixed-point and if you're on an x86, you don't > hav...

opusdec -V opusdec opus-tools f2a2e88 (using libopus unknown) I've got an opus file encoded from a .wav off a cd, 44100Hz: opusinfo 2-24-Overture_in_C_\(In_Memoriam\).opus Processing file "2-24-Overture_in_C_(In_Memoriam).opus"... New logical stream (#1, serial: 38134f1f): type opus Encoded with libopus unknown User comments section follows... ENCODER=opusenc from opus-tools

It's my understanding that the CELT layer of Opus has a maximum input sample rate of 48k, and frequencies above 20k are effectively not encoded. I've been trying to get up to speed on the specification, and studying its operation, but as far as I can infer, there is a fixed set of 21 bands distributed logarithmically to encode DC to 20k. If I were inclined to encode at say, 96k, an...

...inutes, WinAmp starts do blink that "red square" (packet loss or instability), and after about 20 minutes sounds starts to "pop", with small cuts, and then starts to cut a lot. I think the problem is progressive: more time, more cuts. We use SAM Broadcaster to encode AAC+ HE 48k. We're using the latest IceCast KH version, Win32. Other datail: all users have always "Lag: 0" on Admin interface If you want to listen what happens, if you can and have time (very appreciated) the stream is http://mp4.livemix.com.br/livemix - if possible using WinAmp. Wait abou...

...Files 968K build/lib/Transforms/Hello/CMakeFiles 964K build/utils/yaml-bench/CMakeFiles/yaml-bench.dir 964K build/lib/Transforms/Hello/CMakeFiles/LLVMHello.dir 964K build/lib/ExecutionEngine/RuntimeDyld/CMakeFiles/LLVMRuntimeDyld.dir/Targets 956K build/tools/llvm-mt 952K build/tools/clang/test/VFS 948K build/tools/clang/test/VFS/Output 940K build/tools/clang/test/SemaTemplate 940K build/CMakeFiles 936K build/tools/llvm-mt/CMakeFiles 936K build/tools/clang/test/SemaTemplate/Output 932K build/tools/llvm-mt/CMakeFiles/llvm-mt.dir 880K build/lib/Target/NVPTX/InstPrinter 872K build/lib/Target/NVPTX/In...

..., which in my > experience tends to come out at around 22-24kbps Based on the OddCastDSP status screen, I'm getting averages of around 30k in mono, about 42k in stereo. 30k is likely too high for 33.6 modems but don't most people use 56k modems these days? Most connections are around 48k I believe. Ross. --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To unsubscribe from this list, send a message to 'icecast-request@xiph.org' containing only the word 'unsubscribe' in the body. No subject is needed. Uns...

Joern Nettingsmeier wrote: > i'm putting together a streaming relay network for the linux audio > developer's conference #2 (http://www.zkm.de/lad), and i was wondering > how much cpu power will be required on the master server. > > we need to record 2 48k stereo signals simultaneously and encode them > into three streams each: > > 2x 54kbit/s 22k05 mono > 2x 112kbit/s 48k stereo > 2x 192kbit/s 48k stereo > > it would be very nice if the same box could handle the streaming to > between 5 and 8 relay servers, so that we hav...

Dear all, In "HowTos/Disk_Optimization", the calculated value of stride size and stripe width appears to have the "K" suffix incorrectly appended to them. Eg: * (64K/4K) = 16K * (3*16K) = 48K * (16K+16K) = 32K The values provided on the mkfs.ext3 command line however, do drop the "K" suffix. I'm no expert on RAID, but the "K" suffixes do look a bit suspect. Regards, Timothy Lee -------------- next part -------------- An HTML attachment was scrubbed... UR...

Sorry for the crosspost but it's relevant to Vorbis and Icecast I believe. I'm seeing more and more streaming stations using AACplus, with many listeners being amazed at the sound quality. Most say that a 48kb/s sounds better than a 128kb/s MP3, which would put Ogg Vorbis at around 96kb/s IMO. That means only half the bitrate is required in AACplus compared to Ogg Vorbis for the same sound quality. Up until this codec was available, Ogg Vorbis compared favourably with all the others. Is there anyt...

Sorry for the crosspost but it's relevant to Vorbis and Icecast I believe. I'm seeing more and more streaming stations using AACplus, with many listeners being amazed at the sound quality. Most say that a 48kb/s sounds better than a 128kb/s MP3, which would put Ogg Vorbis at around 96kb/s IMO. That means only half the bitrate is required in AACplus compared to Ogg Vorbis for the same sound quality. Up until this codec was available, Ogg Vorbis compared favourably with all the others. Is there anyt...

Hi, I just built a sample application with speex resampler in linux and I tried to resample 8K sine wave tone mono to 48k using speex_resample_process_int. I am using a tool called EAQUAL for audio quality. I find the quality of Speex resampler to be decreasing when I increase the quality q of the resampler init function. Can some one give me pointers regarding this?? As per the API, if the quality factor is increas...

...eCast on a Raspberry Pi and can access the stream ONLY if I use an ffmpeg statement to start the stream. Unfortunately, the stream is choppy and the sound quality is poor. I?ve tried using ICES, but I don?t get any sound. Terminal command that works: ffmpeg -ac 2 -f alsa -i hw:0,0 -acodec mp3 -ab 48k -ac 2 -content_type audio/mpeg -f mp3 icecast://source:password at ipaddress:port/stream ICES terminal command that doesn?t work: ices2 /home/pi/ices-2.0.2/conf/ices-alsa.xml What am I missing? Thanks for any help/suggestions.

...% 0.25K 393 15 1572K filp 4816 3355 69% 0.03K 43 112 172K size-32 2904 2810 96% 0.09K 66 44 264K sysfs_dir_cache 2058 1937 94% 0.58K 343 6 1372K proc_inode_cache 1728 1215 70% 0.02K 12 144 48K anon_vma 1650 1590 96% 0.25K 110 15 440K skbuff_head_cache 1498 1493 99% 2.00K 749 2 2996K size-2048 1050 1032 98% 0.55K 150 7 600K inode_cache 792 767 96% 1.00K 198 4 792K size-1024 649 298 45%...

Hi there, we use IceCast KH on our webradio, no commercial. We have an issue that we dont know where to look. When listening using WinAmp (but in other players too like JW Player), after 70 or 90 minutes, we realize that the "green square" on top of WinAmp becomes a "red square". About 60 minutes after start, sometimes the "red square" blinks, and then more

...es sense, since this codec can work offline too). Opus_custom seems to be needed if you have a certain frame size constraint with low latency. I don't have that. I can use the stock frame sizes ( 120, 240, 480, 960, 1920, and 2880 sample/frame). My conclusion is that I could set up Opus for 48K (stereo), and in reality run it at 44.1K, as long as I use stock frame sizes, and it would be fine. The only issue I can think of is any of the perceptual stuff will be off by -8%, e.g. crossover/mask frequencies, etc. Or is that true? And with Opus_custom is all that stuff recalculated? If I...

...C as the wins server what causes that? They all have very similar ping times from a given client and are spread across our network as the clients are.) Thanks, Duncan > 26029: /usr/local/samba/sbin/smbd -D > 00010000 2832K r-x-- /usr/local/samba/sbin/smbd > 002E2000 48K rwx-- /usr/local/samba/sbin/smbd > 002EE000 1320K rwx-- [ heap ] > FD080000 1224K rw-s- dev:32,0 ino:301936 > FD200000 1736K rw-s- dev:32,0 ino:301937 > FD400000 20288K rw-s- dev:32,0 ino:301938 > FE900000 168K rw-s- dev:32,0 ino:301940 > FE930000 24K r...

...Le jeudi 12 mai 2005 ? 08:49 +0200, Pierre a ?crit : > Pierre wrote: > > Jean-Marc Valin wrote: > > > >> OK, first thing I see is the configure options: > >> > >> ./configure --enable-valgrind --enable-sse --enable-fixed-point > >> --enable-epic-48k --enable-ti-c55x > >> > >> One question: "what do you think you're trying to do here?". Four of > >> these five > >> options are more or less mutually exclusive! SSE is floating poing and > >> x86 > >> specific, nothing to do wit...

Hi Jean-Marc, I use the resampler to convert various sampling frequencies to 48 kHz on my Blackfin platform (fixed-point) 48K -> 16K speex -> 48K chain does not sound very good compared to plain 16K. But the main issue is when processing loud signals, I have truncation (and not clipping/saturation) I could hear it and see it with various music and speech messages. See example.png. I also ran a test with a chirp sig...