similar to: float to PCM packing in libvorbisfile

I've been experimenting with the ideas of Replay Gain[1] and find that ogg123 doesn't have a way of specifying the scaling applied to replayed samples (like -f in mpg123). Looking at libvorbisfile, I see no function exactly matching this possibly desirable behaviour. ov_read() scales by either 128 (byte output) or 32768 (word output), but there's nothing in between. ov_read_float()

Hi, I need to adjust the output volume from ogg123 the same way that mpg123 has the "-f n change scalefactor [32768]" option. I need to do this within ogg123, instead of alternatives that involve pipes, such as piping the output to sox. I do not believe that this is currently possible in ogg123, and I'm looking for some suggestions as to how it can be accomplished by modifying

Hi, I need to adjust the output volume from ogg123 the same way that mpg123 has the "-f n change scalefactor [32768]" option. I need to do this within ogg123, instead of alternatives that involve pipes, such as piping the output to sox. I do not believe that this is currently possible in ogg123, and I'm looking for some suggestions as to how it can be accomplished by modifying

Hello. I noticed that when reading data with ov_read_float(), you can get values outside [-1..1] when the stream is encoded at lower quality, but with higher quality, the values trim down to inside [-1..1]. Looking at the plot from -q10, the data from ov_read_float seems clipped. I've made some plots of encoding the start of Rammsteins Feuer from the xXx soundtrack (it's a pretty loud

Here's a patch to vorbisfile.c in libvorbis that splits the float to integer pcm packing out of ov_read into it's own function. The good stuff: - ov_pack_pcm can be used by folks who decode to float pcm with ov_read_float to do various sound manipulations (eg, preamp, fades, replaygain). No more need to roll your own, which will really begin to pay off when 24 bit output gets

Greetings, I'm having some sort of problem using ov_read_float(). Everything looks good to me, but I'm trashing memory somehow, so clearly I'm screwing something up. What confuses me is why it takes a ***float for the buffer. I call it like so: float **buffer bytes_read = ov_read_float(&vf, &buffer, 0) memcpy (b, *buffer, 0); This compiles fine, but after a few reads,

Hi, in ov_read_float.html there is a small mistake: long ov_read(OggVorbis_File *vf, float ***pcm_channels, int *bitstream); hould read: long ov_read_float(OggVorbis_File *vf, float ***pcm_channels, int samples, int *bitstream); It seems this function has been changed (compared to the RC3). Anything else worked fine for me with the 1.0 release. Olaf <p>--- >8 ---- List archives:

Hello, Am I missing something? float** pcm_channels; int bitstream; int where = ov_pcm_tell(&vf); -> The Result is "0"; ov_read_float(&vf, &pcm_channels, 1, &bitstream); -> pcm_channels[0][0] is "0.000338580" ov_pcm_seek(&vf, 0); char* buffer = new char[2]; ov_read(&vf, buffer, sizeof(buffer), 0, 2, 1, &bitstream); short temp =

Hi there, I've been grappling with this for some time, so I'm finally breaking down and trying this list. I'm trying to integrate libvorbisfile and libao to create a simple sound file player, a la ogg123. To do this, I borrowed heavily from ao_example.c and vorbisfile_example.c. I feed the buffer from ov_read into ao_play in a simple loop. It usually works, but occasionally all it

Hi! I solved the problem using ov_read_float instead ov_read. Still do not know why ov_read returned garbage, but ov_read_float works fine! Thanks! El 17 de julio de 2011 10:41, Juan Miguel Mart?n Mu?oz <jmcubo at gmail.com>escribi?: > Hi, thanks for your replies! > > ogg.k, I'm sure MACOSX is defined. When I'm debugging line by line the > program runs into ov_read.

Hi Gunter, I've solved the sound glitch. Since it was hard to reproduce, it took me a while, but I eventually figured it out. What I needed to do was fill the buffer with more data before handing it off to ao_play. It requires lots of bookkeeping, pointer arithmetic, and a sufficiently large buffer. First, the bigger buffer. I just pulled this code from ogg123: #define PRIMAGIC

Hi Gunter, I think this problem started happening when I upgraded from Debian Wheezy to Debian Jessie. If nothing looks amiss in my code, it probably is a sound driver problem. But since it works 100% of the time in ogg123, I feel I must have missed some corner case. My audio driver is almost always active. I usually have my music player going in the background when I do my testing. The problem

Hi all, Perhaps a stupid question, but I just need pieces of audio from a file. I know I have to use vorbisfile, and the example compiles, but I don't know how to read the buffer. In fact, it's not even documented (or I'm looking at the wrong place). I assumed it was a interleaved buffer containing unsigned shorts, but that didn't work. I also tried the normal decode example,

How can this problem be reproduced? If you have an specific .ogg file that gives problems it would be helpful to have them so we can try to reproduce them. Kind regards, Gunter. On So, Jun 14, 2015 at 7:10 , Marshall Mason <marshallmason2 at gmail.com> wrote: > Hi there, > I've been grappling with this for some time, so I'm finally breaking > down and trying this

Hi, I'm new to writing any kind of sound program and somewhat new to C. I am on linux and I've been trying to get the example code (http://www.xiph.org/ogg/vorbis/doc/vorbisfile/example.html)for the decoder using libvorbisfile to work. I think my main problem is output to /dev/dsp. If I use the example where the only change I made were to open a file instead of using stdin and I ouput

First off, I don't even know if this is the right place to ask these kind of questions, but I haven't been able to find answers anywhere else, so 1. I have found that "pcmTotal * vorbisInfo->channels * 2" gives the uncompressed size of every ogg vorbis file I have used. What is a more robust way of getting the full uncompressed file size? 2. How do I read a certain amount of

Hi Gunter, Thank you for your quick response. As I mentioned in the email, I cannot dependably reproduce it. It comes and goes, and I just have to wait for it to come up before I can try to debug it. I've found it tends to happen when I've been taxing my sound driver, but this isn't consistent either. My inability to consistently reproduce this has made it a nasty bug for me to

I have a wav file which has been converted to an ogg file with libvorbis via ffmpeg. However I've tested a few files just to make sure. I run a standard while loop to using ov_read to pull all the data out of ogg file. Using 4096 as the size to read each call. Watching my debug logs it is reading 2048 bytes each call. When I load the wav file the data chunk for the pcm data is 167680

Hi, I have this code to decode ogg data: unsigned long PSS_OggStream::DecodeOggVorbis(OggVorbis_File *psOggVorbisFile, char *pDecodeBuffer, unsigned long ulBufferSize, unsigned long ulChannels) { int current_section; long lDecodeSize; unsigned long ulSamples; short *pSamples; unsigned long ulBytesDone = 0; while (true) { #ifdef WIN32 lDecodeSize = ov_read(psOggVorbisFile,

Hello, hope you're fine. And hopefully someone can help me. I wrote a short demonstration to show you my problem (see below). I'm asking always for 64 samples. When "ov_read_float()" is finished, I want to seek back to sample 1, and ask immediately for the next 64 samples. But when I run this small app, I get an totally high CPU load. Is there a way to it speed up? Or do I