similar to: FLAC__stream_decoder_skip_single_frame() // how is it used?

On Fri, Jan 7, 2011 at 7:36 PM, Brian Willoughby <brianw at sounds.wa.com> wrote: > This thread has raised several good topics. ?It's surprising that the > FLAC-Dev list has been silent for years, and now suddenly there are several > good ideas to discuss. I'll take credit for this, toot toot toot :D > > On Jan 7, 2011, at 15:04, David Richards wrote: >>

The issue is that silent frames compress to a very small size, and the Ogg packeting layer can put more than one FLAC frame into a page. So if you have an extended period of silence with a live or rate-limited input stream, the client buffers may exhaust themselves before a new page can be put together and transmitted. > This thread has raised several good topics. It's surprising that

Not that many people are streaming FLAC, and if they are, they are, in my mind, undoubtedly experiencing problems. Probably the most prized kind of software problem of all, the intermittent kind. I am streaming FLAC. 24bit no less, and its fantastic. The future I would like to live in, is one where lossless music is streamed and heard by people other than myself. It is generally not possible to

This thread has raised several good topics. It's surprising that the FLAC-Dev list has been silent for years, and now suddenly there are several good ideas to discuss. On Jan 7, 2011, at 15:04, David Richards wrote: > I am interested in streaming lossless audio, FLAC is probably the best > option for that. Currently the OggFLAC way of doing it mostly works > with a few hacks in

On Jan 7, 2011, at 16:48, Ben Allison wrote: > The issue is that silent frames compress to a very small size, and > the Ogg > packeting layer can put more than one FLAC frame into a page. So > if you > have an extended period of silence with a live or rate-limited input > stream, the client buffers may exhaust themselves before a new page > can be > put together and

Thanks Brian. I converted everything to libFLAC and got the same results. Here is some debug output encoder: [34.270050] FLAC encoder set succeeded [34.271183] write_callback, frame: 0, samples: 0 [34.271282] write_callback, frame: 0, samples: 0 [34.271313] write_callback, frame: 0, samples: 0 [34.271351] FLAC encoder initialization succeeded [34.356251] write_callback, frame: 0, samples: 4096

hi thx, exactly the explanation i wanted to hear! i want to try to damage some flac files to hear the effects on the audio, but does anybody knows if there is a program (like hex editor, ...) that could be used to visually 'search' the different blocks in the file, assuming this is even possible. if it's not, what would be a good way to damage the file in some places to be able to

On Fri, Nov 08, 2002 at 07:12:35PM -0800, Josh Coalson wrote: > Yeah, it's useful, so now there is a > FLAC__seekable_stream_decoder_get_decode_position() and > FLAC__file_decoder_get_decode_position(). I haven't documented > them yet but you can see an example in > src/metaflac/operations_shorthand_seektable.c where I use it > during seektable creation. Ok, here is

Hi Jesper, Thanks for reaching out. The links page on the FLAC website needs some pruning. I'm afraid there are a lot of dead links there. I can include your effort here as a library wrapper though: https://github.com/ietf-wg-cellar/flac-specification/wiki/Implementations Anyway, nice to see that libFLAC can be used for Java. As far as I am aware, libFLAC was only available for Java through

To clarify, I am one of the REAPER developers. To say that FLAC itself is not broken is sort of meaningless--FLAC itself is just a specification. So yes, it is just libFLAC that is broken. I have spent QUITE a bit of time trying to sort out why it fails on 24 bit mode.. If there are some specific alignment requirements of samples, please point me to the documentation of them, but we have

On Jul 20, 2010, at 22:28, Dmitry Kichenko wrote: > I'm not sure whether this is the best place to send the question, > but since I'm mostly interested in FLAC and there didn't appear to > be a better place for it.. here goes. I think flac-dev is the perfect place for questions about building flac. But the answers are probably going to be geared towards the FLAC

Cristian Adam wrote: > On Wed, Feb 25, 2009 at 2:23 PM, Brian Willoughby <brianw at sounds.wa.com>wrote: > >> A better suggestion might be to start with libFLAC, optimize as >> needed, and then submit the optimizations back to the FLAC project >> where they will be more widely useful. >> >> But that's just my opinion. >> > > I agree. This

Hello, I joined because this list is the preferred method of communication according to web site, and I would like to tell you about something I've been working with that perhaps you would like to include among the links on the FLAC web site? Since 2014, I have a system that runs as a service on a computer connected to my stereo to play FLAC files, exposing a web interface for playback

The problem is really as I wrote: 1. Metaflac is no option for me, I use libFLAC.dll 2. There is no way (at least how I read the code) to avoid saving comment with libFLAC; I would appreciate an extra option to avoid it, which can default to old behavior if compatibility is important. 3. I have a high speed application, where re-initializing an encoder is really significant. On corner cases it

Hi Justin, I am using flac on the Mac OS X command-line with AIFF input files. In fact, you can use my Mac installer for flac to get everything you need for working 24-bit files in FLAC. You can also build flac from the open-source project to get the same thing. It sounds like you're using libFLAC, and the REAPER programmers are unable to get it working with 24-bit. Have you tried

I wasn’t suggesting that you run metaflac, but that you examine its source to see how it creates new FLAC files without the Vorbis comment. As far as I know, metaflac uses the standard libFLAC and creates files without the Vorbis overhead. My quick review of the source seemed to indicate that calling FLAC__metadata_object_new(FLAC__METADATA_TYPE_VORBIS_COMMENT) will create the comment, but I

Thanks for the tip, Brian. I did have a version that does everything in memory, but only had enough memory to get six threads going. (Although my system has 16GB, I have not taken the time to create 64-bit libraries for wxWidgets, TagLib, libFLAC, etc.) Based on the direction you are pointing me in, I assume that encoding from within the decoder is not permitted. I really like your idea of the

2007/9/8, Brian Willoughby <brianw@sounds.wa.com>: > > Ralph, > > The problem is that there is no clear advantage, at least in terms of > multiple cores, to the approach you're asking about. In order to > allow each stage of the codec to overlap, you need smart buffering > between each stage. That adds code and complexity which isn't there > currently. So

On Tue, Jan 17, 2012 at 09:44:25AM +0100, rs at noveltech.de wrote: > We are currently try to add Gapless support on our device? If we rip an CD > with our device, we can find out, that one track follow after another so we > can > recognize, that the tracks are gapless or not. Are you aiming for maximum compatibility with how the source CD would be played on an average CD audio

No, streams should stay. Audio is NOT a file based process -- it's a stream. You can't listen to an entire song simultaneously. You organize it into files for later use, but you listen and record from a stream. Stream-based storage is practically REQUIRED for an audio codec. It's not random access, it's sequential. You can put wrappers around it to make it convenient for file