similar to: [Cellar] FLAC Markdown

Hi Ruud van Asseldonk, > On Jun 11, 2017, at 7:24 AM, Ruud van Asseldonk <dev at veniogames.com> wrote: > >>> I'm jumping in on this thread to make a few remarks about the spec. I >>> implemented a FLAC decoder by only looking at the spec, and I have a few >>> notes that would have saved me a lot of time if the spec had mentioned >>> them. They

>> I'm jumping in on this thread to make a few remarks about the spec. I >> implemented a FLAC decoder by only looking at the spec, and I have a few >> notes that would have saved me a lot of time if the spec had mentioned >> them. They are obvious in hindsight, of course. >> >> * If the channel assignment includes a difference channel, then the >>

Hello all! (cc-ing the flac-dev list) I would like to give an update as to the recent CELLAR work on the FLAC specification. • Work has been done to make internal and external links more accurate and reliable. • 'Rice Coding' has been clarified as 'Exponential Golomb Coding.' • Clarifications have been made for binary representation. • Typos and other small changes have been

Hi all, And cc'ing flac-dev. > On May 10, 2017, at 12:15 PM, Dave Rice <dave at dericed.com> wrote: > > Hi Andrew, > >> On May 10, 2017, at 11:19 AM, Andrew James Weaver <weevz at uw.edu <mailto:weevz at uw.edu>> wrote: >> >> Hello all! >> >> In a previous discussions on this list about people interested in working on the FLAC

Hi all, > On Jun 5, 2017, at 11:52 PM, Andrew James Weaver <weevz at uw.edu> wrote: > > Hello all! > (cc-ing the flac-dev list) > > I would like to give an update as to the recent CELLAR work on the FLAC specification. > > • Work has been done to make internal and external links more accurate and reliable. > • 'Rice Coding' has been clarified as

> On May 12, 2017, at 1:05 PM, Dave Rice <dave at dericed.com> wrote: > > Hi all, > And cc'ing flac-dev. > >> On May 10, 2017, at 12:15 PM, Dave Rice <dave at dericed.com <mailto:dave at dericed.com>> wrote: >> >> Hi Andrew, >> >>> On May 10, 2017, at 11:19 AM, Andrew James Weaver <weevz at uw.edu <mailto:weevz at

On Thu, Jun 19, 2014 at 03:30:22PM +0400, lvqcl wrote: > BTW, what can you say about the following place in stream_decoder.c > in read_subframe_lpc_() function: > > /*@@@@@@ technically not pessimistic enough, should be more like > if( (FLAC__uint64)order * ((((FLAC__uint64)1)<<bps)-1) * ((1<<subframe->qlp_coeff_precision)-1) < (((FLAC__uint64)-1)

I would like to know if my vision of things is correct about frame structure. -------------------------------------------------------- Frame header -------------------------------------------------------- Subframe1 header|Subframe1 content| -------------------------------------------------------- Subframe2 header|Subframe2 content| --------------------------------------------------------

> -----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

I don't understand the comment in src/libFLAC/stream_decoder.c: /*@@@@@@ technically not pessimistic enough, should be more like if( (FLAC__uint64)order * ((((FLAC__uint64)1)<<bps)-1) * ((1<<subframe->qlp_coeff_precision)-1) < (((FLAC__uint64)-1) << 32) ) */ if(bps + subframe->qlp_coeff_precision + FLAC__bitmath_ilog2(order) <= 32) see

Hello, I've set up and have been reading through the FLAC reference implementation source code on Windows and stepping through it in the debugger.   I've been trying to understand how the protocol knows where the subframe and frame boundaries are. Is there a good tutorial that discusses the ins and outs of the flac protocol? Also, is there a piece of the reference code that shows how

Jean-Marc, >> > It's odd that it "almost" works with the fixed_generic.h. The easiest >> > thing would be to gradually replace routines and see which one causes >> > problem. It's most likely (though I'm not 100% sure) that somewhere in >> > the code, I have a 16-bit value that gets sent to a function/macro that >> > expects a

To avoid crash caused by an unbound LPC decoding when predictor order is larger than blocksize, the sanity check needs to be moved to the subframe decoding functions. --- src/libFLAC/stream_decoder.c | 30 ++++++++++++------------------ 1 file changed, 12 insertions(+), 18 deletions(-) diff --git a/src/libFLAC/stream_decoder.c b/src/libFLAC/stream_decoder.c index d13b23b..211b4db 100644 ---

>> Ok I will give it a try and see what I can come up with. > Is the most interesting data the [frame] and [subframe] info? > Is the qlp_coeff[], warmpup[] and parameter[] data for each frame/subframe important? Yes, it would be nice if MetaFLAC will show minimum and maximum values of parameters (if are not constant/fixed) for the stream. And it would be VERY NICE if you implement

Hi all, We (WebRTC/Google) would like to extend Opus to natively support 120 ms encoding instead of relying on repacketization as a post processing step. This is to ensure that a valid 120 ms packet is always available. I've attached a couple of patches to add this to opus_encoder(), based on the internal repacketization process carried out by 60 ms CELT. We intend to extend this later for

On Mon, Dec 1, 2014 at 10:40 PM, ???? ?????? <bart.gopnik at gmail.com> wrote: > Please help me solve the following problem: > > FLAC tools have a lots of warnings about non-subset files during > encoding, but unfortunately don't have easy way to check/test existing > FLAC stream for subset compliance. > > "flac -a" generates the big text file that has

Hi all, Thanks for the really great help I've received on this board in the past. I have a very particular graph that I'm trying to plot, and I'm not really sure how to do it. I think I should be able to use ggplot for this, but I'm not really sure how. I have a data.frame which contains fifty sub frames containing one hundred data points each. I can do a histogram of each of

Hi, Is there any way to determine frame size in bits or bytes of encoded frame in WB mode for given quality? That is I need to know how many bits encoded frame will occupy in RTP packet. I see SPEEX_SUBMODE_BITS_PER_FRAME request, but it works like this only for NB mode, and for WB mode it return size of WB "subframe" (?) only. -- Regards, Alexander Chemeris. SIPez LLC. SIP VoIP, IM

Hi Felicia, I still don't quite understand why you need to make 120 ms a special case, rather than extend the code that already handles 40 ms and 60 ms. Cheers, Jean-Marc On 06/01/2016 12:58 PM, Felicia Lim wrote: > Hi all, > > I've just realized that there's a better and simpler way of doing this > which ensures that analysis and selection of the mode/bandwidth etc

On 06/01/2016 02:06 PM, Felicia Lim wrote: > That was my intention with refactoring out the subframe encoding and > repacketizing bit. Or do you mean I should merge the explicit check for > 120 ms frame and the existing checks for 40/60 ms wideband? What I mean is that this line in opus_encoder.c: if (frame_size > st->Fs/50 && (st->mode == MODE_CELT_ONLY ||