similar to: Bits per Sample, Channel Count

--- 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 >

Hi, The FLAC__stream_decoder_flush doesn't change the state, but when I play a stream and reach the end I have to do a FLAC__stream_decoder_reset and reread the metadata instead of simply a FLAC__stream_decoder_flush. This is because the stream state, which indicates the end of the stream, is not changed. Is there a reason, for the state not being reset to

Hello everyone, I was just browsing through some video capturing websites/forums and saw some people complain that there was no lossless codec available to encode audio in AVI files. I thought that FLAC would have one but it doesn't seem to be the case. As it's a popular demand (now that hard drives are inexpensive, people can record hours of video/audio without any loss and then

Hi, The stream decoder follows a push model, however, I need a pull model decoder, like, for example, implemented by the Vorbis libs. For this I simply save the pointer to the audio data in the read callback and acces the data lateron after the call to FLAC__stream_decoder_process_one_frame returned. Is this safe? Can I assume that the internal buffer is not freed, when

Hi, currently flac sets all granulepos members of the ogg pages to 0, when embedding a FLAC stream into an Ogg file. Is it possible to change this behaviour to use a more reasonable value, e.g. the bytes written so far. So when the first packets is 1000 bytes long it gets the granulepos 1000 and when the second packet is 1234 bytes long it gets the granulepos 2234 and so on. This would make it

--- Ingo Ralf Blum <ingoralfblum@gmx.de> wrote: > > currently flac sets all granulepos members of the ogg pages to 0, > when > > embedding a FLAC stream into an Ogg file. Is it possible to change > this > > behaviour to use a more reasonable value, e.g. the bytes written so > far. > So > > when the first packets is 1000 bytes long it gets the granulepos

Is there a reason for having the FLAC__stream_decoder_get_input_bytes_unconsumed function private. This function is required now by two of my filters for seeking and buffer managment, when not all data is available from the beginning. Regards, Ingo

Hi, I have a problem when I try to open a file, which is not a FLAC file. When I open a non-flac file with the stream decoder API, one of the first things called is FLAC__stream_decoder_process_metadata, which itself calls stream_decoder_find_metadata_. Unfortunately the non-flac file contains some content, which leads to the state set to FLAC__STREAM_DECODER_READ_FRAME. However in

HI Josh, I'm starting now to compile the ACM codec. I don't know how long it will take, since I do it in my free time, but I guess it will not take to long. About Mark Powell's comment, with an ACM codec there will be a lot of Windows GUIs for FLAC. I'm writing this message to talk about the FLAC format. The GUIs for the Audio Codec Manager, including Winamp and Windows Media

I have few notes: It seems there is changed API in CVS again. So, what about adding function like unsigned FLAC__format_frame_size(const FLAC__Frame *frame) which returns size of the frame in bytes. This can be useful, for example, in xmms plugin to display current bitrate like vorbis plugin does. If 'PERFORMER' field is missing in vorbis comment, it would be nice to display

I am getting large, corrupted native FLAC files when trying to call the encoder from inside the decoder. The metadata in the output file is OK. Is initializing and calling an encoder from inside a libFLAC decoder write callback supported? The encoder is initialized on the first write callback. Environment is Win 7 Ult, 64-bit, VS 2008, libFLAC 1.2.1. Here is some context. The static

Hi (again), I started to work on an ACM (Audio Compression Manager) for FLAC. I currently only have the canvas. The encoding/decoding is not done yet. So it's completely useless. But since the canvas is the harder thing to start, here it is. (just in case I don't have time to continue) http://mukoli.free.fr/flac-ACM.v0.1.0.zip

Hi, >That's what I wanted to say a month ago. Mr. Blum, I'm really sorry for misunderstanding you. My mistake. I choose an ACM codec because it's the only option that works on every Windows Platform, and since the DirectShow has a generic wrapper that allows it to use ACM codecs, it appeared a good idea. I didn't note that the file format was going to be a problem. But I think

Just out today: Zorannt has released test versions of some Vorbis Direct Show filters, courtesy of Info Ralf Blum. http://sourceforge.net/project/showfiles.php?group_id=24067 --- >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-dev-request@xiph.org' containing only the word

Just out today: Zorannt has released test versions of some Vorbis Direct Show filters, courtesy of Info Ralf Blum. http://sourceforge.net/project/showfiles.php?group_id=24067 --- >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-dev-request@xiph.org' containing only the word

We have FLAC support on our AudioReQuest products, and we allow streaming from our built-in web server for MP3s, but can not get Flac working. Does anyone know a Windows based player that will play FLAC streams? I have tried the Winamp plug-in for FLAC (even updated it to 1.1.0), but streaming is not working. We use embedded id3 tags instead of Flac comments if this means anything. Thanks

For instance the write_callback thing: client_data ? can it be a pointer to just about anything? For instance a pointer to a two-dimensional array? Are those FLAC-WAV and WAV-FLAC examples the only examples available? What would be the best approach to read a FLAC file to an array of some kind? Passing a pointer to the array as ?client_data?? I'd like to use a two-dimensional array, but

OK, back to this one... --- Josh Coalson <xflac@yahoo.com> wrote: > --- Ingo Ralf Blum <ingoralfblum@gmx.de> wrote: > > The FLAC__stream_decoder_flush doesn't change the state, but when I > > play a > > stream and reach the end I have to do a FLAC__stream_decoder_reset > > and reread > > the metadata instead of simply a FLAC__stream_decoder_flush.

En r?ponse ? Ingo Ralf Blum <ingoralfblum@gmx.de>: > Hi, > > > I saw on the website that the ACM was a work in progress, but have > seen no > other > > mention anywhere else. So if someone is working on that I'm ready to > help, > > otherwise I'm ready to start it. > > I just read on the FLAC main website, that the MediaXW project contains

First, Thank you for your answers. I using the following code to try simply decode a flac file and write the decoded data raw PCM file. The resulting file is just noise and pops, so is the decoded data in a different format than PCM? struct flacData { FILE *inputFile; FILE *outputFile; char *filename; }; FLAC__StreamDecoderReadStatus flac_DecoderReadCallback(const FLAC__StreamDecoder