similar to: Quality setting Hard Min/Max Bitrate limits in SVN oggenc

Dear sir, I've a problem that the .wav file couldn't play after calling "sampleenc male.wav|sampledec male_speex_15.wav".I found that the new file male_speex_15.wav is smaller than the original file in size.I implemented the test on Linux system.The original file male.wav is 96044 bytes,while the new file male_speex_15.wav is 96000 bytes.I'm eager to know the reason.Thankyou!

I thought it would be useful for testing to include the bitrate in the output filename. I often screw it up and overwrite it by accident. That facility appears in the -n string as %B (uppercase to distinguish it from any comment field argument). I also implemented %F for input filename, for completeness and to save the bother of having to add a comment to a test file just so -n can be used.

To those who have reported having a problem selecting real number quality settings, could you kindly test the new compile you can d/l from: http://homepage.ntlworld.com/jfe1205/testXPd.ZIP and report whether this resolves the problem. No major changes, but I regularised a number of type casting discrepancies that could have been causing a problem. I'm sorry to have to ask for help on this

Stephen F. Booth, the author of a number of audio progs for OS X (sbooth.org), is a subscriber to this list too and he might be able to provide an OS X build of FLAC 1.3.0. However, I can understand the Xiph policy of only supplying officially blessed versions compiled by themselves. I guess we'll have to wait for 1.3.1... Michael On 9 Jul 2013, at 13:50, John Edwards <john.edwards33 at

Hi All, I am planning to implement Ogg Vorbis Decoder on a 24 bit DSP. I have following doubts: 1) What is min and max bitrate? 2)What is max Audio Packet size ? 3)What is the max size of setup header? 4)Any idea how to start for making a 24Bit fixed point reference code? Shall I start with floating point or Tremor or Tremor Low Mem Version? 5) What window sizes in Ogg Vorbis are normally

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

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

From: bumblebritches57 at gmail.com (Marcus Johnson) Date: Mon, 10 Jun 2013 10:03:52 -0400 Subject: [flac-dev] (no subject) Message-ID: <CAFWGNQW0aa6+dGzdSk120PWcTMv+vjPjmsVQmT-p4NN2N9kjVw at mail.gmail.com> Are there precompiled binaries for OS X? if not I could provide them. *** From: john.edwards33 at ntlworld.com (John Edwards) Date: Mon, 10 Jun 2013 15:06:39 +0100 Subject: [flac-dev]

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,

Are the ogg/vorbis programs updated to v1.0.1 ? OggDrop for example ? Regards, David ---------------------------------------------------------------------------- ----------- David Balazic mailto:david.balazic@hermes.si HERMES Softlab http://www.hermes-softlab.com Zolajeva 30 Phone: +386 2 450 8851 SI-2000 Maribor Slovenija

Hello, While tweaking the stream configuration interface in the Airtime broadcast automation tool (http://www.sourcefabric.org/en/airtime/) we noticed that Liquidsoap uses the zero to one scale for Vorbis quality, rather than the more familiar zero to ten scale used by oggenc. It seems I'm not the only person confused by this: http://ffmpeg.org/pipermail/ffmpeg-devel/2009-January/067964.html

Hello, In one of my recent assignments, I was asked to develop a ogg player. I am not able to find the right repository of the source code. There are few repositories on vorbis but I am not clear which one is the right one for Windows environment. If somebody has successfully compiled any ogg player ever, his ideas will be helpful to me. In one of the code set, I was able to compile the

Hi. I've some question about audio-stream of OGM. I know some encoding tools, such as headac3he or BeSweet, can encode 5.1ch AC3 files into 5.1ch vorbis files, but I couldn't find any decoding tools which can decode 5.1ch vorbis files properly. Current Tobias's Ogg Direct Show filter 0.9.9.1 (and its libraries) doesn't support decoding 5.1ch vorbis, right? And if so, how can I

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

The oggenc --help text says The bitrate option (--bitrate, -b) will choose the mode closest to the chosen bitrate. but judging by 'vorbis_encode_init', it's actually the mode with the least bitrate greater than or equal to the specified bitrate that's chosen. Here's a possible fix: The bitrate option (--bitrate, -b) will choose the mode with the least bitrate

...even worse, it *appears* to work, and produces crap files. When specifying a -b <foo> option for a batch encode, -b <foo> is only obeyed for the first file. All subsequent files encode with the equivalent of -b 0 (despite the chatty text output claiming otherwise) resulting in overly low bitrates that sound like shit. I don't know if this affects -q as well, but it a safe

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

Jean-Marc: I know you're seeing these - how about a response? --Randy Randy Yates <yates at ieee.org> writes: > If I understand these terms correctly, you cannot specify simultaneously > all three, e.g., high quality, low bitrate, and low complexity. > > It seems that specifying any two implies the third. For example, > specifying a high quality and low bitrate will

I built ffmpeg2theora from SVN, using libogg 1.1.4, libvorbis 1.2.3 and Thusnelda from SVN. Running ffmpeg2theora --help reports that ffmpeg2theora accepts a range of -2 to 10 for the --audioquality option and a bitrate of 32 to 500 for the --audiobitrate option, but it doesn't appear to actually accept any of the low end numbers. It appears that the lowest acceptable value for

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