similar to: Encoding with non-8/16/24 bits per sample with flac

Dear list, i'm doing a bit of analisys on flac's source code and i've run into something i can't quite grasp. flac version 1.2.1 flaclib C stream_encoder.c function "process_subframes_" line 2999 ++++++++++++++++++++++ if(do_mid_side) { FLAC__ASSERT(encoder->protected_->channels == 2); for(channel = 0; channel < 2; channel++) {

On Thu, Aug 14, 2014 at 12:34 PM, lvqcl <lvqcl.mail at gmail.com> wrote: > Jose Pablo Carballo <jose.carballo at ridgerun.com> wrote: > >> - channels = 2; >> - bps = 16; >> + channels = ((unsigned)buffer[23] << 8) | buffer[22]; >> + bps = ((unsigned)buffer[35] << 8) | buffer[34]; >> total_samples = (((((((unsigned)buffer[43] << 8)

On Sat, Mar 8, 2014 at 7:49 AM, ??????? ?????? <rodionkarimov at yandex.ru>wrote: > I create FLAC file decoding, processing and playing program and have the > following question : how to convert FLAC 16 bit file data to 32 bit float > buffer for CPU processing? I've already inplemented sound playing and > tested it with sine wave - it works without problems; I even made

Hello. I create FLAC file decoding, processing and playing program and have the following question : how to convert FLAC 16 bit file data to 32 bit float buffer for CPU processing? I've already inplemented sound playing and tested it with sine wave - it works without problems; I even made writing into decoding buffer values of sine wave, instead of decoded FLAC file data, and it also works

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

On May 26, 2017, at 9:52 PM, Erik de Castro Lopo <mle+la at mega-nerd.com> wrote: > Mark H. David wrote: >> Notice the warning was given only for the WAVE file produced by the >> flac reference executable. > > I agree. The flac decoder should not generate WAV files which it > warns about when encoding. > >> Should I file a bug for this? > > No need.

Thank you for writing and pointing out that one can have WAVE files without getting the warning, and that your library does it. I found out Goldwave (Windows software) writes 24-bit WAVE files that do not get the warning. (Good for them!) However, the flac executable from the flac reference software apparently produces WAVE files that do get the warning. Example annotated log: Encode a 24-bit

Hello, I've tried encoding a FLAC file from a WAVE file saved with 24-bits-per-sample, using the flac executable. I get this warning: WARNING: legacy WAVE file has format type 1 but bits-per-sample=24 The code is currently here on line 373: src/flac/encode.c where the comment says let these slide with a warning since they're unambiguous I do not really understand the warning

I changed the condition in *_precompute_partition_info_sums_*() functions from if(bps <= 16) to if(FLAC__bitmath_ilog2(default_partition_samples) + bps < 32) and then changed the 'subframe_bps' argument of find_best_partition_order_() in evaluate_fixed_subframe_() and evaluate_lpc_subframe_() as follows: evaluate_fixed_subframe_(): evaluate_lpc_subframe_(): 1)

lvqcl wrote: > I changed the condition in *_precompute_partition_info_sums_*() > functions from > if(bps <= 16) > to > if(FLAC__bitmath_ilog2(default_partition_samples) + bps < 32) > > and then changed the 'subframe_bps' argument of find_best_partition_order_() > in evaluate_fixed_subframe_() and evaluate_lpc_subframe_() as follows: > >

When decoding a FLAC file with bits per sample that is not a multiple of 8 and outputting in raw format, the decoder gets as far as the write callback before realising that it doesn't know how to write the data. It then fails with an unhelpful message (or trips an assert() in debug mode). The attached patch performs the check earlier, and gives a more helpful error message. Kind

I found a bug with FLAC v0.6 raw encoding. It appears that the file pointer in the source file is not reset after seeking to the end for checking the size. I've attached a patch. I'm excited about FLAC!! I've been looking for a good GPL lossless RAW audio compressor for use with sound fonts. Sound font files contain 16 bit samples that are word aligned, so just treating it as raw

----- Forwarded message from Don Armstrong <don@donarmstrong.com> ----- Date: Thu, 30 Sep 2004 16:19:41 -0700 From: Don Armstrong <don@donarmstrong.com> Resent-From: Don Armstrong <don@donarmstrong.com> To: submit@bugs.debian.org Subject: Bug#274301: libflac4 segfaults on corrupt flac files Severity: normal Package: libflac4 Version: 1.1.0-11 Running ogg123 on

--- Ingo Ralf Blum <ingoralfblum@gmx.de> wrote: > Hi, > > these two values can be stored in the stream info metadata block as > well as > in the header of each frame. This leads to some questions. > > - Are these values supposed to change during playback, e.g. first > comes a > frame with 2 channels, then with four and the next switches the bit > count >

I'm guessing here, but it seems that some encoders might only be able to process samples in 4-bit or 8-bit groups, maybe even 16-bit multiples, so this flag would allow the decoder to remove any padding that might have been necessary on the encoder side. For example, if someone were to encode 12-bit samples, but the encoder was forced to use 16-bit registers for calculations, then

2010/4/27 Brian Willoughby <brianw at sounds.wa.com> > i.e. I wouldn't be surprised if there are no FLAC files with this flag > set. > If you look for lossyWAV you'll find that it is actually being used -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/flac-dev/attachments/20100427/12821b2a/attachment.htm

How can i determine how many bits per sample are in .ogg files? Thanks. -- Wayne Davis - wfdavis@seas.upenn.edu - PGP Key Available --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'vorbis-request@xiph.org' containing only the word 'unsubscribe' in the body. No subject

Hi, these two values can be stored in the stream info metadata block as well as in the header of each frame. This leads to some questions. - Are these values supposed to change during playback, e.g. first comes a frame with 2 channels, then with four and the next switches the bit count from 16 to 8 and so on? - There are methods to retrieve these values from the stream decoder. Unfortunately

--- Erik de Castro Lopo <erikd-flac@mega-nerd.com> wrote: > Hi all, > > I'm trying to use the FLAC C libraries to encode audio. > > I'm doing something like: > > FLAC__seekable_stream_encoder_set_channels(pflac->fse, 1); > FLAC__seekable_stream_encoder_set_sample_rate(pflac->fse, 11025); >

Hi Guys, I've just faced a wrong warning trying to encode a 24 bits WAV file if(wFormatTag == 1) { if(bps != 8 && bps != 16) { if(bps == 24 || bps == 32) { /* let these slide with a warning since they're unambiguous */ flac__utils_printf(stderr, 1, "%s: WARNING: legacy WAVE file has