search for: bitspersampl

...ions 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; > duration = 2; > bitspersample = 24; > > samplerange = 2 ^ (bitspersample-1) - 1; > samplemidpoint = 2 ^ (bitspersample-1); > > pi = 4 * a(1); > > scale = 18; > obase = 16; > > for (ix = 0; ix < duration * samplerate; ++ix) { > ? sample = samplemidpoint + samplerange * s(2 * pi * ix / sam...

...ired 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, BitsPerSample, BytesPerSample; RMstatus ret; AFLACStreamPlayer *pThis = (AFLACStreamPlayer *)client_data; pThis = (AFLACStreamPlayer *)client_data; /* Query the m_AppPlayerPipe and check for commands from it. Act * accordingly */ pThis->QueryCommand (); Channels = FLAC__seekable_stream_deco...

...d to test for the presence of awk(1) ? It is specified as one of the standard commands in the LSB : http://refspecs.linuxfoundation.org/LSB_1.0.0/gLSB/command.html Earl ??? awk -- ' ??? BEGIN { ??????????? samplerate = 8000; ??????????? tone = 1000; ??????????? duration = 1; ??????????? bitspersample = 24; ??????????? samplemidpoint = lshift(1, (bitspersample-1)); ??????????? samplerange = samplemidpoint - 1; ??????????? pi = 4 * atan2(1,1); ??????????? for (ix = 0; ix < duration * samplerate; ++ix) { ??????????????????? sample = sin(2 * pi * tone * ix / samplerate); ???????????????????...

...e'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(); numChannels = this->get_channels(); // How big is our sample that we want to give to FLACC? // bitsPerSample is 8,16,24,32 // So 8 = no change for numFrames // 16 = half it // 24,32 =...

...osix to identify non-POSIX awk usages. - $AWK -- ' - BEGIN { - samplerate = '$1'; + $AWK ' + BEGIN { + tone = 1000; + duration = 1; + samplerate = '$1'; + pi = 4 * atan2(1,1); - tone = 1000; - duration = 1; - bitspersample = 24; + bitspersample = 24; + amplitude = 2^bitpersample - 1; - samplemidpoint = 1; - for (sps = 0 ; sps < bitspersample - 1 ; sps++) { - samplemidpoint *= 2; + for (s = 0; s < duration * samplerate; s++) { + sample = sin(2 * pi * tone * s / samplerate); + sample = i...

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

...ks, James Code snippet: =================================================== FlacEncoder flacCompressor; bool setValue = false; // set up regular parameters setValue = flacCompressor.set_channels (numChannels); setValue = flacCompressor.set_bits_per_sample (bitsPerSample); setValue = flacCompressor.set_sample_rate (sampleRate); setValue = flacCompressor.set_blocksize(4608); setValue = flacCompressor.set_qlp_coeff_precision (0); setValue = flacCompressor.set_min_residual_partition_order (3); setValue = flacCompressor.set_max_residual_partition_o...

.... > 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, BitsPerSample, BytesPerSample; > RMstatus ret; > > AFLACStreamPlayer *pThis = (AFLACStreamPlayer *)client_data; > pThis = (AFLACStreamPlayer *)client_data; > > /* Query the m_AppPlayerPipe and check for commands from it. Act > * accordingly */ > pThis->QueryCommand ();...

...other 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; duration = 2; bitspersample = 24; samplerange = 2 ^ (bitspersample-1) - 1; samplemidpoint = 2 ^ (bitspersample-1); pi = 4 * a(1); scale = 18; obase = 16; for (ix = 0; ix < duration * samplerate; ++ix) { ? sample = samplemidpoint + samplerange * s(2 * pi * ix / samplerate); ? s = scale; ? scale = 0; ? sample /= 1; ? sa...

...__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, bitsPerSample)); e = FLAC__stream_encoder_set_sample_rate (encoder, sampleRate); e = FLAC__stream_encoder_set_blocksize (encoder, 2048); e = FLAC__stream_encoder_set_do_escape_coding (encoder, true); e = FLAC__stream_encoder_set_write_callback (encoder, &encodeWriteCallback);...

...ify that harmonics can be processed correctly. tonegenerator () { - # When using GAWK, use --lint=posix to identify non-POSIX awk usages. - $AWK -- ' + awk -- ' BEGIN { samplerate = '$1'; tone = 1000; duration = 1; bitspersample = 24; - - samplemidpoint = 1; - for (sps = 0 ; sps < bitspersample - 1 ; sps++) { - samplemidpoint *= 2; - } - + samplemidpoint = 2^(bitspersample - 1); samplerange = samplemidpoint - 1; pi = 4 * atan2(1,1); @@ -127,7 +112,7 @@ tonegenerator ()...

...tonegenerator () > { > - # When using GAWK, use --lint=posix to identify non-POSIX awk usages. > - $AWK -- ' > + awk -- ' > BEGIN { > samplerate = '$1'; > > tone = 1000; > duration = 1; > bitspersample = 24; > - > - samplemidpoint = 1; > - for (sps = 0 ; sps < bitspersample - 1 ; sps++) { > - samplemidpoint *= 2; > - } > - > + samplemidpoint = 2^(bitspersample - 1); > samplerange = samplemidpoint - 1; > > pi = 4 *...

...? +# Replay gain tests - Test the rates which have specific filter table entries +# and verify that harmonics can be processed correctly. + +tonegenerator () +{ +??? awk -- ' +??? BEGIN { +??????????? samplerate = '$1'; + +??????????? tone = 1000; +??????????? duration = 1; +??????????? bitspersample = 24; + +??????????? samplemidpoint = lshift(1, (bitspersample-1)); +??????????? samplerange = samplemidpoint - 1; + +??????????? pi = 4 * atan2(1,1); + +??????????? for (ix = 0; ix < duration * samplerate; ++ix) { +??????????????????? sample = sin(2 * pi * tone * ix / samplerate); +???????????...

...ify that harmonics can be processed correctly. tonegenerator () { - # When using GAWK, use --lint=posix to identify non-POSIX awk usages. - $AWK -- ' + awk -- ' BEGIN { samplerate = '$1'; tone = 1000; duration = 1; bitspersample = 24; - - samplemidpoint = 1; - for (sps = 0 ; sps < bitspersample - 1 ; sps++) { - samplemidpoint *= 2; - } - + samplemidpoint = 2^(bitspersample - 1); samplerange = samplemidpoint - 1; pi = 4 * atan2(1,1); @@ -127,7 +112,7 @@ tonegenerator ()...

...But the resulting files are remarkable different between my application and the FLAC frontend (although using the same settings). for example: FLAC frontend (quality = 8) -------------------------------- FLAC_vendor = reference libFLAC 1.1.1 20041001 bitrate = 1047 samplerate = 44100 channels = 2 bitspersample = 16 codec = FLAC ---------- 4831008 samples @ 44100Hz File size: 14 334 137 bytes My Application (quality = 8) ------------------------------- FLAC_vendor = reference libFLAC 1.1.1 20041001 bitrate = 1314 samplerate = 44100 channels = 2 bitspersample = 16 codec = FLAC ---------- 4831008 sa...

Can the bitsPerSample be set any higher than 16? I try to set it to 20 or 24 and get this error... DarkIce: LameLibEncoder.h:122: specified bits per sample not supported [24] teststream# <p>When I start Darkice I get the following error for about 5 seconds and then it stops with "broken pipe". 08:57...

Actually...I think it's because I set it to mono instead of stereo...anyway, is there a way to set the bitsPerSample higher? Matt <p><p>>>> Matt@cmitech.com 8/28/02 9:09:03 AM >>> Hmmm...I raised my sample rate to 44100 from 22050 and now it seems to be more stable??? >>> Matt@cmitech.com 8/28/02 9:04:47 AM >>> Can the bitsPerSample be set any higher than 16? I...

...seems to be working I get a really slow input stream from darkice. I think the stream must have double speed for a correct playback. This is the current section of my darkice.cfg Code: [general] duration = 0 bufferSecs = 1 [input] device = /dev/dsp sampleRate = 22050 bitsPerSample = 16 channel = 1 [icecast-0] bitrateMode = abr quality = 1 format = mp3 bitrate = 16 server = localhost port = 8000 password = XXXXXXXXXXXXXXX mountPoint = stream When I start the stream I hear it on a client. But its to slow...

...ter: arecord -l you should see: card 1: CODEC [USB Audio CODEC], device 0: USB Audio [USB Audio] that means all is fine so far. in darkice.cfg, try this for the input section: # this section describes the audio input that will be streamed [input] device = hw:1,0 sampleRate = 44100 bitsPerSample = 16 channel = 2 [icecast2-0] your settings for icecast here... that works, tested it with a line-in source 5 min ago with my 302. u. Kristoffer Gustafsson: > Hi. > I didn't Think of saying thank you. I Had so much to do. > I have beenup very late and tried...

...================================================ > FlacEncoder flacCompressor; > bool setValue = false; > > // set up regular parameters > setValue = flacCompressor.set_channels (numChannels); > setValue = flacCompressor.set_bits_per_sample (bitsPerSample); > setValue = flacCompressor.set_sample_rate (sampleRate); > setValue = flacCompressor.set_blocksize(4608); > setValue = flacCompressor.set_qlp_coeff_precision (0); > setValue = flacCompressor.set_min_residual_partition_order (3); > setValue = flacCompressor.set...