similar to: Question on Quality factor, Bitrate and decode table

Hi there, I'm using libvorbis in my program and need to encode to target bitrate. I know libvorbis prefer to use quality scale but I can't use it. I've found something at faq http://vorbis.com/faq/#quality *For now, quality 0 is roughly equivalent to 64kbps average, 5 is roughly 160kbps, and 10 gives about 400kbps. Most people seeking very-near-CD-quality audio encode at a quality

I'm a bit confused about VBR with Opus. I see that the default is VBR, but I don't see any way to specify a quality setting. I can set a target bitrate, but that's definitely not what I want; I want a constant quality level, like "-q" in oggenc, and for the encoder to select bitrates based on the desired quality. For example, for playback through earbuds or laptop

Jean-Marc Valin <jean-marc.valin at usherbrooke.ca> writes: > Quality and bit-rate are different ways of specifying exactly the same > thing. Complexity is orthogonal. That clears it up - thanks. Is this in the manual? --Randy > > Jean-Marc > > On 2010-04-03 08:28, Randy Yates wrote: >> Jean-Marc: I know you're seeing these - how about a response? >>

Hie, I just don't understand one point in the speex's man page about the quality and bitrate settings. Are these settings's behavior same as Vorbis settings ? Quality option set variable bitrate in order to maintain constant quality during file and bitrate option set a defined bitrate , isn't ? I'm not sure, so I'm pleased if you could say that I'm right. A good idea is

Hi, all. I'm new to this list, and I know that my question touches on a FAQ. I'm just hoping to get more information than "it's a priority item, but it's not done yet." (Basically what the FAQ says.) I've been encoding about 20 minutes of mono audio each week to as small a file as I can, so it can be served via html. As long as the result is understandable,

This mail depends upon the fact that I don't have a couple of good earphones ;-) I read in the site that q=6 is a very high quality, but does it contain perceivable differencies from the original? (for 95% of people, of course). I also found q=6 to produce files slightly bigger (1/10 bigger) than those produced with lame VBR q=2 (about 192 bps on average). I always thought LAME VBR q=2

Hi all, I used vorbis several years ago because of its freeness and VBR, but after playing ha.org and getting an mp3-capable discman, I switched to APS and API mp3 encoding. Now, I am liking musepack a lot, but the flexibility is more "there" for the vorbis format, including players, tagging comments (lyrics in the comments - killer idea!), etc.... However, I am a bit worried

Michael, > See the examples. If you initialise a managed mode (which this is), you > MUST use vorbis_bitrate_addblock() and vorbis_bitrate_flushpacket(). Thanks for the tip. I added the calls, and it works now. > You SHOULD also give an option to set min/max, since they're now used, > and to use a VBR mode rather than the managed modes here. But when streaming, IMHO it is

I'm sorry if this question has been asked before; I've looked through the archives and haven't seen anything. The problem I'm seeing is that oggenc's VBR encoding doesn't seem to pay attention to any sort of bitrate limitation, either the -m or bitrate_hard_min settings. It isn't that it temporarily dips below the minimum; the average for the whole (in this case,

On 2010-04-11 10:09, Randy Yates wrote: > If I specify VBR mode via > > speex_encoder_ctl(pSpeexEncoder, SPEEX_SET_VBR,&isTrue); Try: speex_encoder_ctl(pSpeexEncoder, SPEEX_SET_ABR, &desiredRate);

I've just done some rudimentary testing to see how Vorbis degrades at absurdly low bitrates without downsampling. In summary, don't hope for anything decent below -q 0 for now. I tried oggenc -b <bitrate> -M <bitrate> for the below and a few in between: 24k - spectral energy "floor" captured decently, but many pure-tone blips (think old computer movie sound effects)

Hi! I'm very impressed with the progress Ogg Vorbis has made, ever since getting that T-shirt at LinuxWorld Expo :-) Using oggenc, I've run into a problem, though: "Mode initialisation failed: invalid parameters for bitrate" This happens when 2 conditions are true: 1) the input file has a sampling rate less than 44100, and 2) the -M parameter is used to specify a maximum

Hi! I have some MP3 audio material which is basically speech with some background noises, essentially > 120Hz and < 5kHz. I had the idea to reduce the file size by recoding the material to Opus at 56kbps. Unfortunately the result is a file sampled at 48kHz much larger than the original. I hope you agree that it does not make sense to create a file larger than the original (MP3). Of course

Hi, I'm quite familiar w/ mp3 cbr/abr/vbr encoding, as well as mpeg4 (cbr/vbr,etc). And I can always calc the bit rate for a given file size with: file size * 8000 / length in seconds = kbits/sec Works great w/ mpeg4 + mp3. BUT FOR THE LIFE OF ME: I cannot get oggenc (1.0x version) to give me the file size I want. I calc. it with the above formula, and nothing comes out right. Then I do

On 27.11.2017 02:00, YIRAN LI wrote: > I'm using libvorbis in my program and need to encode to target bitrate. I > know libvorbis prefer to use quality scale but I can't use it. > > I've found something at faq http://vorbis.com/faq/#quality > > *For now, quality 0 is roughly equivalent to 64kbps average, 5 is roughly > 160kbps, and 10 gives about 400kbps. Most

Hi, I'm experimenting with IceCast2, using DarkIce to generate the stream. I have found some peculiarities with the vorbis bitrates. In DarkIce, I call vorbis_encode_init() with about the following values: vorbis_encode_init( &vorbisInfo, 2, 44100, 96, 96, 96); which by all reasons should generate a 96 kb/s stream, as all max_bitrate, nominal_bitrate and min_bitrate are set to 96.

I had expected that the default bitrate for the encoder would be the same as setting it to OPUS_AUTO, but I'm getting difference results: >opusenc --comp 4 sample.wav sample.opus Encoding using libopus 1.3-rc2 (audio) ----------------------------------------------------- Input: 8 kHz, 1 channel Output: 1 channel (1 uncoupled) 20ms packets, 25 kbit/s VBR Preskip: 312

The problem is that the mp3 dewdz will be used to looking at bitrates to determine quality, so the solution (as several here say) seems to me to just rip out anything mentioning bitrate from the simple tools. But, they will still see the bitrate in players like winamp. To solve this, why not store the quality setting used when encoding into the ogg file itself ? Either use a field unused when

In testing, I have found that both MP3 and RealAudio 3(!) are much better sounding at bitrates from 8 to 40-something kbps than Ogg Vorbis. This definatly needs drastic improvents and I know that 1.1 will be focusing on improvements in those areas but that seems pretty far off(a few months minimum) which leaves low-bitrate streams sounding pretty bad. --- >8 ---- List archives:

I created some test samples and transcoded to FDK AAC and libopus at fairly low bitrates - I cannot recreate what bothered me about Opus & noisy music previously. It also seems I cannot tease ffmpeg into encoding FDK's AAC with VBR. As it stands, Opus clearly wins in this scenario.* Q: Is it possible to stream in variable bitrate? * ffmpeg -i "$track" -vn -ac 2 -c:a libfdk_aac