16kHz .wav won't play in VLC player at the correct speed

[Aroema]

I have wav file i recorded with Wavosaur where I set audio configuration for 'Audio in' at 16000hz

play in VLC player and its slower and lower pitch than it should
play in Sound Recorder accessory its also slower and lower pitch than it should

play in wavosaur and its also slower and lower pitch than it should
but that's cos in Wavosaur audio configuration the 'Audio out' needs to set not to the default of 44100hz but 16000hz

How do I get 16000Hz wav file to play in VLC player or the Sound Recorder accessory to play at the correct speed or Hz?

[Lotesdelere]

Please open Tools -> Messages (set Verbosity to 2) before you start the playback and then paste the full resulting log here or on Pastebin.com if it's too long.

Also upload a short sample of a problematic file to either http://streams.videolan.org/upload/ or to EmbedUpload.com, the latter will upload the file for you to several other hosts and then post the link to the file here.

[Aroema]

Hi, I've uploaded the short wav file 199kb
input and output 16khz.wav
to http://streams.videolan.org/upload/
and set Tools -> Messages Verbosity to 2 (debug)

main debug: processing request item: input and output 16khz.wav, node: Playlist, skip: 0
main debug: resyncing on input and output 16khz.wav
main debug: input and output 16khz.wav is at 0
main debug: starting playback of the new playlist item
main debug: resyncing on input and output 16khz.wav
main debug: input and output 16khz.wav is at 0
main debug: creating new input thread
main debug: Creating an input for 'input and output 16khz.wav'
qt4 debug: IM: Setting an input
main debug: using timeshift granularity of 50 MiB, in path 'C:\DOCUME~1\user\LOCALS~1\Temp'
main debug: `file:///C:/Documents%20and%20Settings/user/My%20Documents/My%20Music/Wavosaur.1.0.7.0%28en%29/input%20and%20output%2016khz.wav' gives access `file' demux `' path `/C:/Documents%20and%20Settings/user/My%20Documents/My%20Music/Wavosaur.1.0.7.0%28en%29/input%20and%20output%2016khz.wav'
main debug: creating demux: access='file' demux='' location='/C:/Documents%20and%20Settings/user/My%20Documents/My%20Music/Wavosaur.1.0.7.0%28en%29/input%20and%20output%2016khz.wav' file='C:\Documents and Settings\user\My Documents\My Music\Wavosaur.1.0.7.0(en)\input and output 16khz.wav'
main debug: looking for access_demux module: 3 candidates
main debug: no access_demux module matching "file" could be loaded
main debug: TIMER module_need() : 0.495 ms - Total 0.495 ms / 1 intvls (Avg 0.495 ms)
main debug: creating access 'file' location='/C:/Documents%20and%20Settings/user/My%20Documents/My%20Music/Wavosaur.1.0.7.0%28en%29/input%20and%20output%2016khz.wav', path='C:\Documents and Settings\user\My Documents\My Music\Wavosaur.1.0.7.0(en)\input and output 16khz.wav'
main debug: looking for access module: 3 candidates
filesystem debug: opening file `C:\Documents and Settings\user\My Documents\My Music\Wavosaur.1.0.7.0(en)\input and output 16khz.wav'
main debug: using access module "filesystem"
main debug: TIMER module_need() : 0.294 ms - Total 0.294 ms / 1 intvls (Avg 0.294 ms)
main debug: Using stream method for AStream*
main debug: starting pre-buffering
main debug: received first data after 0 ms
main debug: pre-buffering done 1024 bytes in 0s - 14705 KiB/s
main debug: looking for stream_filter module: 4 candidates
main debug: no stream_filter module matching "any" could be loaded
main debug: TIMER module_need() : 0.218 ms - Total 0.218 ms / 1 intvls (Avg 0.218 ms)
main debug: looking for stream_filter module: 1 candidate
main debug: using stream_filter module "stream_filter_record"
main debug: TIMER module_need() : 0.163 ms - Total 0.163 ms / 1 intvls (Avg 0.163 ms)
main debug: creating demux: access='file' demux='' location='/C:/Documents%20and%20Settings/user/My%20Documents/My%20Music/Wavosaur.1.0.7.0%28en%29/input%20and%20output%2016khz.wav' file='C:\Documents and Settings\user\My Documents\My Music\Wavosaur.1.0.7.0(en)\input and output 16khz.wav'
main debug: looking for demux module: 55 candidates
wav debug: chunk: fcc=`fmt ` size=16
wav debug: format: 0x0001, fourcc: araw, channels: 1, freq: 16000 Hz, bitrate: 15Ko/s, blockalign: 1, bits/samples: 8, extra size: 0
wav debug: found Raw audio audio format
wav debug: chunk: fcc=`data` size=202752
main debug: selecting program id=0
main debug: using demux module "wav"
main debug: TIMER module_need() : 0.766 ms - Total 0.766 ms / 1 intvls (Avg 0.766 ms)
main debug: looking for a subtitle file in C:\Documents and Settings\user\My Documents\My Music\Wavosaur.1.0.7.0(en)\
main debug: looking for decoder module: 31 candidates
araw debug: samplerate:16000Hz channels:1 bits/sample:8
main debug: using decoder module "araw"
main debug: TIMER module_need() : 0.263 ms - Total 0.263 ms / 1 intvls (Avg 0.263 ms)
main debug: looking for meta reader module: 2 candidates
lua debug: Trying Lua scripts in C:\Documents and Settings\user\Application Data\vlc\lua\meta\reader
lua debug: Trying Lua scripts in C:\Program Files\VideoLAN\VLC 2.0.2\lua\meta\reader
lua debug: Trying Lua playlist script C:\Program Files\VideoLAN\VLC 2.0.2\lua\meta\reader\filename.luac
main debug: no meta reader module matching "any" could be loaded
main debug: TIMER module_need() : 1.716 ms - Total 1.716 ms / 1 intvls (Avg 1.716 ms)
main debug: `file:///C:/Documents%20and%20Settings/user/My%20Documents/My%20Music/Wavosaur.1.0.7.0%28en%29/input%20and%20output%2016khz.wav' successfully opened
main debug: Buffering 0%
main debug: recycling audio output
main debug: looking for audio output module: 2 candidates
aout_directx debug: Opening DirectSound Audio Output
aout_directx debug: found device: Primary Sound Driver
aout_directx debug: found device: Realtek AC97 Audio
main debug: Buffering 16%
main debug: Buffering 33%
main debug: Buffering 50%
main debug: Buffering 66%
main debug: Buffering 83%
main debug: Buffering 100%
main debug: Stream buffering done (350 ms in 5 ms)
aout_directx debug: device supports 2 channels (DEFAULT!)
aout_directx debug: device supports 1 channel
aout_directx debug: Windows says your SpeakerConfig is stereo
aout_directx debug: creating DirectSoundThread
aout_directx debug: DirectSoundThread ready
main debug: using audio output module "aout_directx"
main debug: TIMER module_need() : 117.754 ms - Total 117.754 ms / 1 intvls (Avg 117.754 ms)
main debug: output 's16l' 16000 Hz Stereo/Mono frame=1 samples/4 bytes
main debug: mixer 'f32l' 16000 Hz Stereo/Mono frame=1 samples/8 bytes
main debug: filter(s) 'f32l'->'s16l' 16000 Hz->16000 Hz Stereo/Mono->Stereo/Mono
main debug: looking for audio filter module: 14 candidates
audio_format debug: f32l->s16l, bits per sample: 32->16
main debug: using audio filter module "audio_format"
main debug: TIMER module_need() : 0.272 ms - Total 0.272 ms / 1 intvls (Avg 0.272 ms)
main debug: conversion pipeline completed
main debug: looking for audio mixer module: 2 candidates
main debug: using audio mixer module "float32_mixer"
main debug: TIMER module_need() : 0.134 ms - Total 0.134 ms / 1 intvls (Avg 0.134 ms)
main debug: input 'u8 ' 16000 Hz Mono frame=1 samples/1 bytes
main debug: looking for audio filter module: 1 candidate
scaletempo debug: format: 16000 rate, 2 nch, 4 bps, fl32
scaletempo debug: params: 30 stride, 0.200 overlap, 14 search
scaletempo debug: 1.000 scale, 480.000 stride_in, 480 stride_out, 384 standing, 96 overlap, 224 search, 800 queue, fl32 mode
main debug: using audio filter module "scaletempo"
main debug: TIMER module_need() : 0.216 ms - Total 0.216 ms / 1 intvls (Avg 0.216 ms)
main debug: filter(s) 'u8 '->'f32l' 16000 Hz->16000 Hz Mono->Stereo/Mono
main debug: looking for audio filter module: 14 candidates
main debug: no audio filter module matching "any" could be loaded
main debug: TIMER module_need() : 0.145 ms - Total 0.145 ms / 1 intvls (Avg 0.145 ms)
main debug: looking for audio filter module: 14 candidates
main debug: using audio filter module "trivial_channel_mixer"
main debug: TIMER module_need() : 0.140 ms - Total 0.140 ms / 1 intvls (Avg 0.140 ms)
main debug: looking for audio filter module: 14 candidates
audio_format debug: u8 ->f32l, bits per sample: 8->32
main debug: using audio filter module "audio_format"
main debug: TIMER module_need() : 0.163 ms - Total 0.163 ms / 1 intvls (Avg 0.163 ms)
main debug: conversion pipeline completed
main debug: filter(s) 'f32l'->'f32l' 16000 Hz->16000 Hz Stereo/Mono->Stereo/Mono
main debug: conversion pipeline completed
main debug: filter(s) 'f32l'->'f32l' 17600 Hz->16000 Hz Stereo/Mono->Stereo/Mono
main debug: looking for audio filter module: 14 candidates
main debug: using audio filter module "samplerate"
main debug: TIMER module_need() : 0.423 ms - Total 0.423 ms / 1 intvls (Avg 0.423 ms)
main debug: conversion pipeline completed
main debug: End of audio preroll
main debug: Decoder buffering done in 114 ms
main warning: PTS is out of range (-9619), dropping buffer
main warning: audio output out of sync, adjusting dates (60990 us)
main warning: not synchronized (60993 us), resampling
main debug: EOF reached
main debug: waiting decoder fifos to empty
main debug: waiting decoder fifos to empty
main debug: waiting decoder fifos to empty
main debug: waiting decoder fifos to empty
main debug: waiting decoder fifos to empty
main debug: finished input
main debug: removing module "araw"
main debug: killing decoder fourcc `u8 ', 0 PES in FIFO
main debug: removing module "audio_format"
main debug: removing module "trivial_channel_mixer"
main debug: removing module "scaletempo"
main debug: removing module "samplerate"
main debug: removing module "aout_directx"
aout_directx debug: closing audio device
aout_directx debug: DirectSoundThread exiting
main debug: removing module "audio_format"
main debug: removing module "float32_mixer"
main debug: releasing audio output
main debug: removing module "wav"
main debug: removing module "stream_filter_record"
main debug: removing module "filesystem"
main debug: Program doesn't contain anymore ES
main debug: dead input
qt4 debug: IM: Deleting the input
main debug: changing item without a request (current 0/1)
main debug: nothing to play
main debug: TIMER input launching for 'input and output 16khz.wav' : 31.612 ms - Total 31.612 ms / 1 intvls (Avg 31.612 ms)

[Lotesdelere]

The file is playing slower in any player because it has been saved that way.

in Wavosaur audio configuration the 'Audio out' needs to set not to the default of 44100hz but 16000hz

I'm not used to Wavosaur but for instance in Audacity there are two rate settings: one is about the whole project and the other one about the sound wave you're working on. This is the latter you must set to 44.1 kHz to be able to save the file with the proper speed and pitch.

[Aroema]

When i set verbosity to 1 (warning)
main warning: PTS is out of range (-9529), dropping buffer
main warning: audio output out of sync, adjusting dates (60917 us)
main warning: not synchronized (60920 us), resampling

Do these warnings mean anything?

When verbosity is 2 debug it mentions 16000 a few times (as below) but not 44100 therefore it should play at 16khz. Is 44100 VLC and the Sound Recorder accessory default?

wav debug: format: 0x0001, fourcc: araw, channels: 1, freq: 16000 Hz, bitrate: 15Ko/s, blockalign: 1, bits/samples: 8, extra size: 0
araw debug: samplerate:16000Hz channels:1 bits/sample:8
main debug: output 's16l' 16000 Hz Stereo/Mono frame=1 samples/4 bytes
main debug: mixer 'f32l' 16000 Hz Stereo/Mono frame=1 samples/8 bytes
main debug: filter(s) 'f32l'->'s16l' 16000 Hz->16000 Hz Stereo/Mono->Stereo/Mono
main debug: input 'u8 ' 16000 Hz Mono frame=1 samples/1 bytes
scaletempo debug: format: 16000 rate, 2 nch, 4 bps, fl32
main debug: filter(s) 'u8 '->'f32l' 16000 Hz->16000 Hz Mono->Stereo/Mono
main debug: filter(s) 'f32l'->'f32l' 16000 Hz->16000 Hz Stereo/Mono->Stereo/Mon
main debug: filter(s) 'f32l'->'f32l' 17600 Hz->16000 Hz Stereo/Mono->Stereo/Mono

[Lotesdelere]

it should play at 16khz.

It does play at 16kHz but that's not what you want to hear.

Once again, your file has been badly made, here is how it should have been done:
http://www.embedupload.com/?d=8WNZBFERNA

The rate of the sound has been increased from 16 kHz to 44.1 kHz to achieve the correct playback speed but notice that the sample rate of the whole file is 48 kHz. IMO you are confusing the sound rate and the file sample rate.

[Aroema]

I click on the link but on the page I can't seem to find a link with just lets me download the file. It keeps wanting me to install something instead. Also saying I need to update my flash player when I just updated it on Saturday to 11.5.502!

[Aroema]

Now i remmember using the Amiga years ago. If i had a low Hz file say 16khz and then increase its Hz to 24khz it would
sound higher and shorter, yet it is the opposite with Wavosaur (and I'm guessing with perhaps the other two programs)!

I put "sound rate" into Google but comes up with albums!

Wavosaur also can resample. I could have decided just to record at 44.1khz but that kind of defeats the purpose - as in to reduce megabytes. I also have an mp3 version also at 16khz but also plays slow.
If i was to upsample or downsample I'd want it to be a multiple of the previous sample eg 16x2 so as to reduce aliasing.
What would happen if I recorded at 48khz or resampled to 48khz?