8khz wav file

Cancel Copy to Clipboard. In this line. The problem might be that 16 should be something wavread returns, not something you have to specify. Even though the error message does not seem to be related with that. It is just an idea,try. I still met the same problrm after modify the code, Carols Even I copy the wav file to the same folder with the source code, the error message still the same. And I have tried to use another wav file from my friend he could use it in his code base , still see the error.

Thanks for all your kindly help I will try to find some command to modify it later. Walter Roberson on 14 Mar What does size y and sie Fs report? Answers 1. Wayne King on 14 Mar Vote 1. Carlos is correct that you should not have not have an actual scalar value as the target of an assignment.

It should be nbits or some similarly named output. Actually I am very surprised that you did not get the error:. Find centralized, trusted content and collaborate around the technologies you use most. Connect and share knowledge within a single location that is structured and easy to search.

I'm currently extracting mel features from my baby cry sound dataset and the wav files' sampling rate is 8kHz, 16bit, mono and about 7 sec. But as you can see, whenever I extract features with different sampling rates sr , the values of the mel-spectrogram change. I thought that since the wav file's sampling rate is 8kHz, if I set the sampling rate to over 16kHz the value of Hertz must be same. First you are asking it to create an FFT-based spectrogram over the possible range.

So the possible frequency range for a signal sampled at Contrary to a regular FT-based spectrogram, a mel spectrogram, does not have a linear frequency scale, but an almost logarithmic scale.

To map the FT-based spectrogram to the logarithmic scale, all available data is mapped to a specific number of logarithmically spaced bins. This range is mapped onto logarithmically spaced bands. If your signal is sampled at 16 kHz, you can represent a range from 0 to 8 Hz. This range is mapped onto logarithmically spaced bands, i. This must lead to different results. Stack Overflow for Teams — Collaborate and share knowledge with a private group.

Create a free Team What is Teams? Collectives on Stack Overflow. Learn more. Asked 2 years, 6 months ago. In case the user desires to utilize other formats supported by an installed sound card, this can be done with the Sound Recorder application, which is part of all Windows packages. The conversion dialog is started by manipulating the file properties or as part of a Save As operation.

The reader may want to consult the check sheets of two-way conversions between the formats made with Sound Recorder and the G. The memory dumps demonstrate the benefit of using the program's codecs compared to the ones provided by Microsoft. The application opens a Windows G. Similarly, raw data from an ISDN recording may be stuffed into a standard wave file and thus integrated into any multimedia oriented Windows application.

This is a program under construction and it doesn't yet have much of a user interface. However, the tool can also be used as a console oriented command line utility, e. Please consult the G. The conversions can be made without a sound card. In case the user does have a card, the application can be used to play telephone sounds and to perform some simple editing and recording.

The application is being developed and tested on a Windows NT 4. Find out which file formats are appropriate for your phone system, and what differentiates one audio file format from another. With dozens of manufacturers in existence and new platforms being released every day, there are a multitude of options for business phone systems that support professional IVR and MOH Systems.

Knowing what type of audio file formats your phone system can support is important. Some of the most prevalent file container formats in the telephony ecosystem are. But many platforms only support certain audio encoding methods, and the encoding methods may be distinguished by different formatting standards. Wav files using this format will commonly be referred to by their bit depth, then sampling rate and encoding method — the most common PCM format for phone systems being a 16 bit, 8 kHz.

Adaptive Differential Pulse Code Modulation ADPCM is an audio encoding method that adapts the size of the compression sampling, which results in a smaller file size with acceptable quality in many types of call environments.

Occasionally, other containers such as. The sample rate is commonly set at 8 kHz, but may occasionally be seen at 6 kHz as a requirement on some phone systems. The container for these files is.