similar to: Getting subframe type=verbatim on 16 bit files

Here's how I set up the data for processing: // For moving data into 32 bit shape uint8_t *buffer8 = NULL; uint16_t *buffer16 = NULL; uint32_t *buffer32 = NULL; unsigned sample32; unsigned sample, channel; uint32_t bitsPerSample = this->get_bits_per_sample(); numFrames = inData.GetSize();

looks fine, I would suspect how the PCM sample are formatted and sent to process(), could you show that part of the code? Josh --- James Smith <jsmith@landmarkdigital.com> wrote: > > I'm using libFLACC++ and libFLAC and I think that I'm using the calls > in the > typical order (see code below). But every monoe or stereo file that > I send > thru I get files

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,

Hi Flac developers! I'm using the seekable_stream_decoder API of libFLAC 1.03 and I'm stuck with a mysterious problem: my FLAC plugin works nicely as long as the seek_absolute() function is never used, but as soon as I seek around in a sample (ie. call seek_absolute() with an offset != 0), I cannot detect the end of a file anymore. decoder_get_state() never returns

Hi All, I m having a strange problem with the Ogg-Vorbis writting code. The code I m using to write is skipping some samples at the end of the file. For example I m converting the 10000 sample .wav file ( 441000 sample rate , 16 bit depth , stereo ) to ogg format. But while reading the ogg file I only find 5824 samples in the ogg file. Can any one suggest what could be wrong in the code. Is

Hi All, I m having a strange problem with the Ogg-Vorbis writting code. The code I m using to write is skipping some samples at the end of the file. For example I m converting the 10000 sample .wav file ( 441000 sample rate , 16 bit depth , stereo ) to ogg format. But while reading the ogg file I only find 5824 samples in the ogg file. Can any one suggest what could be wrong in the code. Is

I am deeply sorry about the corrupt message just being sent; there seems to have been a compatibility issue with my mailer and my browser. This is an identical copy of the previous message: Hello Martin, Vorbis encoders are lossy, which is in a sense equivalent to converting the sample size of the raw PCM stream into something that would result in the desired bitrate. The "sample size"

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 Ruud van Asseldonk, > On Jun 11, 2017, at 7:24 AM, Ruud van Asseldonk <dev at veniogames.com> wrote: > >>> I'm jumping in on this thread to make a few remarks about the spec. I >>> implemented a FLAC decoder by only looking at the spec, and I have a few >>> notes that would have saved me a lot of time if the spec had mentioned >>> them. They

>> I'm jumping in on this thread to make a few remarks about the spec. I >> implemented a FLAC decoder by only looking at the spec, and I have a few >> notes that would have saved me a lot of time if the spec had mentioned >> them. They are obvious in hindsight, of course. >> >> * If the channel assignment includes a difference channel, then the >>

Earl Chew wrote: > I'm a little reluctant to introduce another compiled program when there are > so many other options that will work well enough out of the box. > > Here are two ideas: > > 1. Use bc(1) to compute the raw samples > 2. Use perl(1) to compute the raw samples > > To generate raw unsigned samples using bc(1) for example: > > samplerate = 1000;

Thanks for the advice. With complexity=1, bit rates 5950,8000 and 15000 (CBR only), I'm still getting 30-50 seconds encoding time for a 10-second file. To anyone who has made this work in ARMv4, or knows how to, can I get some advice on the settings? FIXED_POINT is already defined in all relevant files. My source code is patterned after sampleenc and I haven't tried using Ogg.. my source

Brian Willoughby wrote: > What about using the C library sin() and cos() functions to generate > the test audio instead of sox? I did not see a description of how > the test files are generated, so maybe this is easy or maybe it is > hard. The benefit of shipping the test audio generation source code > around with the FLAC sources is that the tests won't break when

I'm using seekable_stream_decoder, And., this is my write_callback. I'm not getting the required output. The PCM i get is not the proper music. Am I doing something wrong here? FLAC__StreamDecoderWriteStatus AFLACStreamPlayer::StreamWriteCb ( const FLAC__SeekableStreamDecoder *decoder, const FLAC__Frame *frame, const FLAC__int32 * const buffer[], void *client_data) { int Channels,

On Nov 26 22:39:27, hans at stare.cz wrote: > ./test_replaygain.sh fails for me in tonegenerator(), saying: > > ./test_replaygain.sh[91]: mawk: not found > Testing FLAC replaygain 8000 (8000 x 1) ... -: ERROR: got partial sample > > Apparently, the tone-generating awk script does not work with > my system's awk, which is "awk version 20110810" as distributed

--- Roland Rabien <Roland.Rabien@mackie.com> wrote: > 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 =

Hi all, I try to enable FEC in the encoder using the macro OPUS_SET_INBAND_FEC and I set the packet loss percentage to a constant value of 30%, using the macro OPUS_SET_PACKET_LOSS_PERC. Please find my encoder settings below: opus: encoder fmtp (maxplaybackrate=8000;maxaveragebitrate=24000;sprop-stereo=1;cbr=1;useinbandfec=1;usedtx=1) opus: encode bw=narrow bitrate=24000 fch=auto vbr=0 fec=1

Hello, I have been tinkering with Wine on Solaris 8 x86. I downloaded the source (20010305) and built wine. The installation went very well until the last part where I asked to build a registry using regapi. Here is the error I get when trying to run regapi. bash-2.03# ./regapi setValue < /usr/local/software/wine-20010305/winedefault.reg Could not stat /mnt/fd0 (No such file or

Hi , guys: I have a problem on the openldap Manager account login : The server is running CentOS 5.3 i386 and I have phpldapadmin-1.0.1-1.el5.noarch.rpm installed . It's running apache 2.2.3 with php 5.1.6 and openldap 2.3.34. I believe I have slapd setup correctly but I'm not completely sure. My /etc/openldap/slapd.conf file has... access to * by

hi all, i'm new to speex and i'm stuck on a problem. I have a buffer that holds 5 seconds of audio data, sampled at 16000 hz and quantitized with 16 bits pcm. I only want to encode it with speex and then decode it and write it back in the same buffer. no compiling errors and no errors during runtime, but from the sample of 5 seconds only a short noise is all I get. I hope you have any