search for: monophon

...nnels. part1: All channels to be coded are summed, each one effected by a coefficient related to its sound power importance for the listener. The sum of all channels is devided by the square root of the sum of coefficients, in order to provide an overall channel providing the user the feeling of a monophonic coding from all channels. This overall channel is stored in a backward compatible bitstream like if it was a monophonic recording. Additional information to reconstruct all the joint coded channels will be put in ancillary data. This way, old decoders are able to provide an acurate monophonic ove...

Hi, there is an interesting case when the FLAC encoder (using 1.2.0) is given simple waves. Simple waves means: I have a list of {frequency, duration, pause} tuples that define the monophonic tune. In other words, exactly one frequency is played at a time. This is the original dataset from 1989 (driving a PC speaker back then): $ ls -l ihold.sd -rw-r--r-- 1 jengelh users 20616 Aug 14 00:57 ihold.sd Since driving the PC speaker is mostly a privileged operation these days, one would...

When I encode a monophonic wav file, I would expect the resulting ogg file to be at about half the bit rate specified on the command line, as stated in the "oggenc -h" help text: "The 6 modes are approximately 112, 128, 160, 192, 256, and 350 kbps (for stereo 44.1kHz input. Halve these numbers for mono input...

Hi, I have the encoder running on an ARM7 at 66mHz and it takes about 2-times real-time to encode a monophonic pcm file. Both quality and complexity are set to 3. I am using the ADS 1.2 tool-chain which seems to optimize C pretty well. I cannot use the inline ARM assy code because the operations used are only available for the V5 and up core. I would be grateful for any thoughts or suggestions. Tha...

Op 03-12-14 om 15:49 schreef Olivier Tristan: > [...] > If you want to check this on a single piano note, I would be > happy to know if this improves the monophonic use case as well. This sample is indeed a case where the retuning of FLAC 1.3.1 shows a severe regression. It seems the patch does work quite well in this case, but it does not fully fix the regression. FLAC 1.3.0 -5: 1066kB FLAC 1.3.0 -8: 874kB FLAC 1.3.1 -5: 1066kB FLAC 1.3.1 -8: 1062kB Patc...

Hi, This patch improves compression a very tiny bit on average, but up to 0.1 percentage point for classical music. I haven't found any tracks that show worsening compression with this patch. -------------- next part -------------- A non-text attachment was scrubbed... Name: 0001-Improve-LPC-order-guess.patch Type: text/x-patch Size: 0 bytes Desc: not available Url :

--- Jan Engelhardt <jengelh@computergmbh.de> wrote: > Hi, > > > there is an interesting case when the FLAC encoder (using 1.2.0) is > given simple waves. Simple waves means: I have a list of {frequency, > duration, pause} tuples that define the monophonic tune. In other > words, exactly one frequency is played at a time. > > This is the original dataset from 1989 (driving a PC speaker back > then): > > $ ls -l ihold.sd > -rw-r--r-- 1 jengelh users 20616 Aug 14 00:57 ihold.sd > > Since driving the PC speaker is mostly...

Dear all, I'm trying to finding the fundamental harmonics (ie. peaks in a periodogram) from a time series (extracted from an mp3). For example, if I look at spectrum(fdeaths, spans = c(3,3)) I'd say the fundamental harmonics are about 1, 2, 3.5 and 4.5 - but how can I extract these automatically? (preferably with some heuristic for choosing the smoothing spans too) I'm aware of

...ce of another. This is the Elec 301 project of Justin Chen, Matthew Hutchinson, Gina Upperman, and Brian VanOsdol. http://cnx.org/content/col10252/latest/ (is Jean-Marc in Vorbis-dev list? This one should have some use to him) Musical Instrument Recognition To detect the pitch and instrument of a monophonic signal. To decompose polyphonic signals into their component pitches and instruments by analyzing the waveform and spectra of each instrument. Elec 301 Project Fall 2005. http://cnx.org/content/col10313/latest/ Cool stuff. -Ivo

...ave been misled by a common misconception in the > consumer audio industry. ... > The reason why most consumer electronics experts get this wrong is > because of the standard techniques use in studio recording. Most > music is recorded as multiple channels, e.g., 16, that are each > monophonic. These channels are played back through a mixing console, > and a simple pan pot is used to artificially place them in a > location. Because the pan pot only effects the volume, not the phase > difference or time delay, this means that a studio recording is going > to have no direct...

I see :( That what I would call a good struct size optimisation. Please tell me there was another reason behind this being only 3 instead of 8 or 16 bits, right ? 2017-01-25 18:30 GMT+01:00 Tor-Einar Jarnbjo <tor-einar at jarnbjo.name>: > Hello Olivier, > > the limitation is in the file format itself, as the number of channels is > encoded in a 3 bit field in the streaminfo

Brian Willoughby wrote: > The ckSize field can be odd to represent the size of the valid data. > > However, the chunk itself must always be an even size. This requires a padding byte at the end of a chunk before the next chunk can begin, or before the end of file. The latter case is the one that most often occurs in buggy RIFF writing programs - the last chunk will have an odd ckSize and

Hi FLACers I'm studying music production and am currently doing an analysis of the FLAC format. If anyone has the knowledge and a minute to explain i would greatly appreciate any help. One thing that i cannot make sense from in the FLAC documentation (a thing that is hard to find info on in general) is how the decorrelation fase works exactly. I understand that the two channels in a stereo

Cool, thanks for all the great comments. I think we agree now on that the "find mp3 before encoding" feature would not be a good idea to implement in the flac core. As Brian pointed out, it might be a better idea to create a program that automatically checks if a flac might have been an mp3 source. My first suggestion was to use FFT, because I know that 128kbps mp3 have a low-pass

Hi, Searching this list archives, I couldn't find a definitive answer to my question : how to send SMS to Gigaset phones ? My goal is to send Alert SMS such as "This phone system will be stopped in 5mn for maintenance" to every terminal (SIP phones and Gigaset DECT phones). (So at the moment, I'm not looking for way to send SMS from handsets). I could successfully send 1 short