search for: 96000

...(opusdec does a bunch of extra things, plus for interactive use we care about both the encoder and decoder, and celt_pitch_xcorr gets used vastly more by the encoder than the decoder... I think the decoder only uses it for PLC). > > Something like > ./opus_demo restricted-lowdelay 48000 2 96000 comp48-stereo.sw /dev/null > > comp48-stereo.sw can be found here: https://people.xiph.org/~tterribe/opus/comp48-stereo.sw > > celt_pitch_xcorr also gets used by the SILK encoder (more in fixed-point than float, but the float one uses it, too). So it may be worth doing a run with the ap...

Hi, I'm experimenting with managed mode encoding with specifying maximum bitrate. I call: vorbis_encode_init( &vorbisInfo, 2, 44100, -1, 96000, 96000); to initialize the encoding. To my surprise, it seems the maxbitrate value of 96000 doesn't have an effect, the bitrate of the generated vorbis audio hovers around 96kb/s, going up to 105 kb/s every once in a while. If I call vorbis_encode_init( &vorbisInfo...

...xtra things, plus for interactive use we care about both the encoder and decoder, and celt_pitch_xcorr gets used vastly more by the encoder than the decoder... I think the decoder only uses it for PLC). >> > >> > Something like >> > ./opus_demo restricted-lowdelay 48000 2 96000 comp48-stereo.sw /dev/null >> > >> > comp48-stereo.sw can be found here: https://people.xiph.org/~tterribe/opus/comp48-stereo.sw >> > >> > celt_pitch_xcorr also gets used by the SILK encoder (more in fixed-point than float, but the float one uses it, too). So it...

...a bunch of extra things, plus for interactive use we care about both the encoder and decoder, and celt_pitch_xcorr gets used vastly more by the encoder than the decoder... I think the decoder only uses it for PLC). > > > > Something like > > ./opus_demo restricted-lowdelay 48000 2 96000 comp48-stereo.sw /dev/null > > > > comp48-stereo.sw can be found here: https://people.xiph.org/~tterribe/opus/comp48-stereo.sw > > > > celt_pitch_xcorr also gets used by the SILK encoder (more in fixed-point than float, but the float one uses it, too). So it may be worth doi...

...arametrization for the call vorbis_encode_init()? For rc2, it worked calling it the following way: vorbis_encode_init( &vorbisInfo, 2, // two channels 44100, // 44.1kHz -1, 96000, // 96kb/s -1 ); And this produced an Ogg Vorbis stream of about 80 kbps on average. But in rc3, the above call produces a circa 240 kbps Ogg Vorbis stream (as reported by xmms). I tried putting 96000 in place of the -1 values, and also putting 96 instread o...

...FALSE, no check for existance and read permission of the file will be performed. } } \details{ -All kinds of wav files are supported: mono / stereo, 8 / 16 bits per sample, 1000 to 48000 samples/second. +All kinds of wav files are supported: mono / stereo, 8 / 16 / 24 bits per sample, 1000 to 96000 samples/second, +but no compressed formats are supported. } \value{ the Sample object that is equivalent to the wav file. Only in sound/man: loadSample.Rd~ diff -ru sound-orig/man/nullSample.Rd sound/man/nullSample.Rd --- sound-orig/man/nullSample.Rd 2006-02-20 12:56:37.000000000 -0500 +++...

...plus for interactive use we care about both the encoder and decoder, and celt_pitch_xcorr gets used vastly more by the encoder than the decoder... I think the decoder only uses it for PLC). >>> > >>> > Something like >>> > ./opus_demo restricted-lowdelay 48000 2 96000 comp48-stereo.sw /dev/null >>> > >>> > comp48-stereo.sw can be found here: https://people.xiph.org/~tterribe/opus/comp48-stereo.sw >>> > >>> > celt_pitch_xcorr also gets used by the SILK encoder (more in fixed-point than float, but the float one uses...

Hi I noticed a crash issue when I passed the following values: celt_mode_create(96000, 258, &e); CELTMode->mdct.kfft[0] is not initialized after calling?clt_mdct_init() and when?celt_mode_destroy() is called it tries to dereference this value in kiss_fft_free(). -- Bjoern Here's the callstack: !kiss_fft_free(const kiss_fft_state * cfg=0x00000000) ?Line 650 + 0x3 bytes C!c...

hello, can I get out of wine more than 16 bit output? sound_file -> player -> wine_out, which is 24/32 to linux? somethink like that, I am audiophile :( thank you

hi All, I happened to login to one of my servers today and saw 96000 failed login attempts. shown below is the address its coming from. I added it to my firewall to drop. Failed password for root from 123.183.209.135 port 14299 ssh2 FYI - others might be seeing it also. Jerry

...e couldn't play after calling "sampleenc male.wav|sampledec male_speex_15.wav".I found that the new file male_speex_15.wav is smaller than the original file in size.I implemented the test on Linux system.The original file male.wav is 96044 bytes,while the new file male_speex_15.wav is 96000 bytes.I'm eager to know the reason.Thankyou! sincerely, Weiqin Bao -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20070712/a54e8550/attachment.htm

...bis_encode_init()? > >For rc2, it worked calling it the following way: > >vorbis_encode_init( &vorbisInfo, > 2, // two channels > 44100, // 44.1kHz > -1, > 96000, // 96kb/s > -1 ); > >And this produced an Ogg Vorbis stream of about 80 kbps on average. > >But in rc3, the above call produces a circa 240 kbps Ogg Vorbis stream >(as reported by xmms). I tried putting 96000 in place of the -1 values, >an...

Hi, I am trying to encode PCM to OGG based on example code in OggVorbis SDK Win 1.0. There is no problem with "vorbis_encode_init_vbr(&vi, 1, 44100, 0.4f);" setting. However, using "vorbis_encode_init(&vi, 1, 44100, 96000, 96000, 96000);" setting gives such a warning by ogginfo: "Warning: EOS not set on stream 1" I have modified ogginfo to display exact playback length. The original PCM data I am trying to encode is about 89.310s. OggInfo displays "89,309683s" for OGG encoded with VBR setti...

...PM Subject: problem with encoding cbr <p><p>Hi, I am trying to encode PCM to OGG based on example code in OggVorbis SDK Win 1.0. There is no problem with "vorbis_encode_init_vbr(&vi, 1, 44100, 0.4f);" setting. However, using "vorbis_encode_init(&vi, 1, 44100, 96000, 96000, 96000);" setting gives such a warning by ogginfo: "Warning: EOS not set on stream 1" I have modified ogginfo to display exact playback length. The original PCM data I am trying to encode is about 89.310s. OggInfo displays "89,309683s" for OGG encoded with VBR setti...

...;s the preferred method for avoiding `unexpected' bits of scientific notation in write.table() output? I've found several inelegant workarounds, but I'm sure I'm overlooking an obvious answer A simple example.. . > df_data.frame(df=95000+1000*0:10); df df 1 95000 2 96000 3 97000 4 98000 5 99000 6 100000 7 101000 8 102000 9 103000 10 104000 11 105000 > write.table(df,'c:\\df.txt',quote=F,sep='\t',row.names=F,col.names=F) .. . creates a file with contents .. . 95000 96000 97000 98000 99000 1e+05 101000 102000 103000 104000 105000 .....

Hi All, Need Help ! I am interested testing opus codec encoding decoding qaulity. for this have complied opus code codec from souce. After compiling i got opus_demo app. for Encoding i followed below steps: 1) Reference file used music_orig.wav (http://www.opus-codec.org/examples/samples/music_orig.wav) Number of samples : 4358219 (90.8 s) 2015-04-13 10:40:07 UTC Sampling

Here are some minor things I noticed when updating the vorbis module for GoldWave: Bitwise.c, line 175 and 207 Warning: Negative unsigned value ret=-1UL; Info.c, line 385 Warning: Unreachable code break; vorbisfile.c, line 1407 Warning: Call to function with no prototype int host_endian = host_is_big_endian(); fix: add 'void' to line 1339: static int host_is_big_endian( void )

...CEA-861D or below edid_version [0x3] CEA-861-B, C or D manufacture_id 0x2d4c product_id 0x69b port_id 0x0 support_hdcp 0 support_ai 1 audio_sync_delay 0 speakers [0xf] FL/FR LFE FC RL/RR sad_count 4 sad0_coding_type [0x1] LPCM sad0_channels 2 sad0_rates [0x6e0] 32000 44100 48000 88200 96000 sad0_bits [0xe0000] 16 20 24 sad1_coding_type [0x1] LPCM sad1_channels 6 sad1_rates [0x6e0] 32000 44100 48000 88200 96000 sad1_bits [0xe0000] 16 20 24 sad2_coding_type [0x2] AC-3 sad2_channels 6 sad2_rates [0xe0] 32000 44100 48000 sad2_max_bitrate 640000 sad3_coding_type [0x7] DTS sad...

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 how should I pack the data to feed the encoder. Any guidance regarding the use of libFLAC, the understanding of the format, or anything else that you can notice where I'm complete lost,...

Hello all, I'm using gentoo, with kernel 3.17.0-p1-pf and at some point, a patch was included in this branch of the gentoo kernel that broke hdmi audio. I've checked with the latest 3.19 vanilla kernel, and I still have the same problem. I cannot output multichannel sound over hdmi. After some investigations, I've narrowed down the issue to the following lines in the file