search for: group__flac__stream__encod

On Jan 2, 2007, at 5:15 AM, Arek Korbik wrote: > The binary frameworks distribution, as found on: > http://xiph.org/quicktime/download.html , hasn't really been tested > much outside the XiphQT project. I never had any problems with them, > but any improvement suggestions or help is welcome. Well, ok, then if the framework is intended to work, then let me describe closer the

I do the following to init flac: encoder = FLAC__stream_encoder_new(); e = FLAC__stream_encoder_set_do_mid_side_stereo (encoder, numChannels == 2); e = FLAC__stream_encoder_set_loose_mid_side_stereo (encoder, numChannels == 2); e = FLAC__stream_encoder_set_channels (encoder, numChannels); e = FLAC__stream_encoder_set_bits_per_sample (encoder, jmin (24,

...erleaved(). if you could send > your code where you are doing that, it would help. > > note that samples going in to the process calls must be converted > to signed 32-bit integers (this is lossless) regardless of the > initial format. see also: > http://flac.sourceforge.net/api/group__flac__stream__encoder.html#ga63 > > from your description it sounds like you might be sending packed > 16-bit samples somehow which could cause every other sample to get > encoded. OK, now I'm totally stumped. I can confirm that I'm sending the encoder 32-bit signed integers, and that they are...

Hi. I'm have a little problem with setting compression level since 1.1.3 releases. I'm using : FE = FLAC__stream_encoder_new(); FLAC__stream_encoder_set_channels( FE, channels); FLAC__stream_encoder_set_bits_per_sample( FE, bitsample); FLAC__stream_encoder_set_sample_rate( FE, samplerate); unsigned int level; if (formatdatalen>0) level = *(char*)(formatdata); else level = 7; if

...C__stream_encoder_process_interleaved(). if you could send your code where you are doing that, it would help. note that samples going in to the process calls must be converted to signed 32-bit integers (this is lossless) regardless of the initial format. see also: http://flac.sourceforge.net/api/group__flac__stream__encoder.html#ga63 from your description it sounds like you might be sending packed 16-bit samples somehow which could cause every other sample to get encoded. Josh __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http:/...

...c I am using, it is fairly old, but I > don't see any version numbers in the headers. Is this a known bug > that has been fixed? 88200Hz is not in the FLAC subset http://flac.sourceforge.net/format.html#subset so first you need to turn off subset checking http://flac.sourceforge.net/api/group__flac__stream__encoder.html#ga16 Josh ____________________________________________________________________________________ Need Mail bonding? Go to the Yahoo! Mail Q&A for great tips from Yahoo! Answers users. http://answers.yahoo.com/dir/?link=list&sid=396546091

Hi All, It is possible to make Opus/CELT a lossless coder if I allow a sufficiently high bit rate? We considered using FLAC, but FLAC's latency is well beyond the acceptable range. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/opus/attachments/20120822/60c69469/attachment.htm

Hello all, I recently upgraded the libFLAC used in my application Max (http:// sbooth.org/Max/) to 1.1.3 and added preliminary support for album art. During the upgrade I evidently made some coding mistakes with interesting results. I've combed everything over and can't quite see the problems. I've become somewhat frustrated because my code didn't really change

Hi, In the last days I've been taking as reference the example found in examples/c/encode/file/main.c. With it I've been able to encode a 2ch, 16 bps, 44100 sample rate input WAV file to a FLAC file. Now I've been trying to modify this example to encode a 2ch, 24 bps, 96000 sample rate WAV file. I have to say I'm a bit lost on how I should read the input file in this case, and

Greetings, I'm playing aroung with the library and try to encode some PCM samples out of a WAVE file. I recognized that my own encoding is bigger than the encoding with the command line tool and I can't figure out why it is so. I tried some API functions to get a better result but I missed. minimal test code: http://stinfwww.informatik.uni-leipzig.de/~mai02cal/test.c example wav:

Hi, I'm sorry to say that I've found the libFLAC++ interface and documentation pretty slim and baffling. What I need to do is add some application-specific metadata to a FLAC file (i'm using the FLAC++ fileencoder). It is currently just a string of characters, which doesn't need to be null-terminated. I've looked through the docs, i've looked through the tests, and

...e of 1 sample, then increase until you trigger the problem. a couple other things that are probably not related: 1. you can replace a lot of the settings more reliably and get better compression using FLAC__stream_encoder_set_compression_level() instead, see http://flac.sourceforge.net/api/group__flac__stream__encoder.html#ga20 2. libFLAC does not support 32 bps yet > The second issue is likely more serious. When I attempt to add album > art to one of the FLAC files I've created (using the level 2 metadata > interface), the files are becoming corrupt (in a sense). Testing > with flac-t ind...

Hi, We are using the libFLAC++ library to encode FLAC files with Vorbis comments in them and have had the FLAC 1.1.2 version fail consistently (access violation or segmentation fault) on both Windows and Linux. We tracked this down to the metadata code in the C++ API. The problem is in src/libFLAC++/file_encoder.cpp. There are two problems - firstly, a C-style cast is hiding a bad conversion