similar to: Incomplete format description?

> -----Ursprungliche Nachricht----- > Von: Josh Coalson > > --- Tor-Einar Jarnbjo <Tor-Einar_Jarnbjo@grosch-link.de> wrote: > > Hi, > > > > have I just overseen a link, or is the format description on > > http://flac.sourceforge.net/format.html lacking information on how to > > actually decode the residual partitions to PCM sample data? > > you

Torsdag, 23 januar 2003, skrev Miroslav Lichvar <lichvarm@phoenix. inf.upol.cz>: >If input is 16 bit, side channel will be 17 bit (16bit - 16bit is >17bit number). And warmup samples will be (17 - wasted_bits) bit. Voila, this was the source of all my frustration, sync problems and who knows what. My decoder now works correctly for files encoded with the default settings. Are

Torsdag, 23 januar 2003, skrev Tor-Einar Jarnbjo <Tor-Einar_Jarnbjo@grosch- link.de>: >According to the format description, the coding method has to be 0. > >I've been using libFLAC 1.0.4 to encode the stream. I've checked my interpretation against "flac -a" and it seems to read 17 bits for each warmup sample. Here is its output: frame=168 blocksize=4608

> -----Ursprungliche Nachricht----- > Von: Josh Coalson > > Miroslav's suggestion is the best right now. I probably should > hand-craft a stream that exercises a decoder as much as possible. I've already tested my decoder with the options -0 to -8 and it works, but the streams doesn't contain any verbatim or constant subframes. I guess this depends heavily on the input

--- Tor-Einar Jarnbjo <Tor-Einar_Jarnbjo@grosch-link.de> wrote: > > -----Ursprungliche Nachricht----- > > Von: Josh Coalson > > > > --- Tor-Einar Jarnbjo <Tor-Einar_Jarnbjo@grosch-link.de> wrote: > > > Hi, > > > > > > have I just overseen a link, or is the format description on > > > http://flac.sourceforge.net/format.html

> -----Ursprungliche Nachricht----- > Von: Josh Coalson > > yes, I will probably get to it soon after the release. > the encoding side is pretty convoluted but for decoding, > src/libFLAC/stream_decoder.c:read_residual_partitioned_rice_() > should be pretty straightforward once you ignore the > FLAC__SYMMETRIC_RICE stuff (which is not used). feel free to > ask

Mandag, 2 juni 2003, skrev du: >On Mon, 02 Jun 2003 21:22, Vitaly Nechyos wrote: >This is what people usually want when they say 'normalization'. AFAIK, you >can't easily tinker with the vorbis frames to add a scale factor in the same >way you can with MP3, but I could be wrong. I could be wrong too, but if I haven't overseen anything, it should be possible to

The IETF passed the application/ogg MIME type some days ago. I couldn't bring you the news earlier because of the MS SQL worm that has been wreaking havoc all over my local networks. The IETF wants some clarifications in Silivias draft for the Ogg stream format, but apart from that I think it will also be passed soon. The RFC and IANA registration of this mimetype will probably be published

Hi everyone, after attempting to correct Ivo's statement about Theora and Vorbis being the "default codecs" in the HTML 5 draft, I just got the following mail off list: "Why are you still here? And by here I mean the Xiph lists. I've seen you speaking ill of our formats and of free software more than in one occasion. So, for what purpose do you still stick around? What

Hi everybody, after hearing about the new Sound functionality in Flash 10, Arek's performance gain using the new Vector type instead of Array and not at least actually seeing the performance boost he achieved compared to my old Vorbis decoder, I found a few hours yesterday to migrate my old code. First of all: the new PCM playback API for Flash is rather restricted and only allows 44.1

Isn't the FLAC encoder 'tuned' for the compression of audio data at common sample rates anyway? Does it make sense to use FLAC to compress arbitrary analog data at very high sample rates as opposed to other general purpose compression algorithms? Tor Am 25.06.2020 um 14:49 schrieb Martijn van Beurden: > Op di 2 jun. 2020 om 05:59 schreef Con Kolivas <kernel at kolivas.org

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

Torsdag, 23 januar 2003, skrev "Tor-Einar Jarnbjo" <Tor-Einar_Jarnbjo@grosch- link.de>: >- My implementation seems to decode fixed subframes now, but the lpc >subframes contains nothing but noise, and the bitstream is out of sync after >reading one. The source code I'm using for decoding the lpc subframe is >available here:

--- Tor-Einar Jarnbjo <Tor-Einar_Jarnbjo@grosch-link.de> wrote: > Torsdag, 23 januar 2003, skrev Miroslav Lichvar <lichvarm@phoenix. > inf.upol.cz>: > > >If input is 16 bit, side channel will be 17 bit (16bit - 16bit is > >17bit number). And warmup samples will be (17 - wasted_bits) bit. > > Voila, this was the source of all my frustration, sync problems

Hi Guys, I know that FLAC format is currently limited to 8 channels but I was wondering if this hard limitation of the format or if it can be easily circumvented if the flac library is compiled with other settings and/or the software using it don't mind it Thanks ! -- Olivier Tristan Research & Development www.uvi.net

Hi All, I am getting this error when i try to compile the "Linphone" package by typing----- make. please help me i am feeling very frustrated with this error pasdt from 7 days i am getting this error. please help me. speexec.c: In function `speex_ec_process': speexec.c:112: `spx_int32_t' undeclared (first use in this function) speexec.c:112: (Each undeclared identifier is

Hi, There is something I don't understand about the page_segment and segment_table values in the documentation. As I understand it, the segment table consist of as many bytes as specified in page_segment, PLUS one trailing byte with a value between 0 and 254. I've looked at some files that has a Comment tag so large, that it is spread over two pages. Here, the first Page Header (not the

Hi! I don't find any detailed information about this in the specification, so I try a question here. If two plain, unmultiplexed Ogg/Vorbis-files are chained into one file, you end up with an Ogg-file with two logical Vorbis streams. When playing this file, would it be correct for a decoder to synchronize the two streams according to the granule position of the Ogg pages (playing them

Conrad Parker schrieb: > so perhaps patching the build to include libspeex is not technically > difficult. Of course it'd make more sense to access libspeex through > VoIP apps than just through the generic MediaPlayer interface. > Even if they included Speex support, the Android media API is very limited, as it can only access audio from a file or an HTTP URL. I raised a bug

I think the software is ripe enough for a broader audience now, so I would just like to inform you about the Java library J-Ogg, available from http://www.j-ogg.de Features: - Open source & free. - Decoding of Ogg-encapsulated Vorbis and FLAC streams. - Support for local files, single files over HTTP, icecast streams over HTTP. - Plugins for JMF (Java Media Framework). - J2ME (Java 2 Micro