Wav 8khz 16bit Mono |TOP| Download
Andrew Henson
Using Windows XP, Aud 2.0.2, Rode NT2A mic, Avid Fast Track Solo pre-amp. Project is for music on hold with announcements. Importing mp3 audio and resampling to 8khz and reduce to stereo to mono. Playback is poor, as expected with reduced sampling rate.
G.722 is more complex than that, and will have lower quality than 16 bit 16kHz signed linear. Asterisk supports files in raw 16 bit, 16kHz, little endian, mono, signed linear, using the filename extension .sln16, so if you record to that format, you will get no additional losses from transcoding to the variant of G.722 that Asterisk supports, beyond those inherent in using G.722.
It would be more common practice to say that this is an 8 bit 8 kHz mono file. But kbps is a valid way of saying it, although kbps alone is not sufficient, as 64 kbps could also be a 16 bit 4 kHz mono file (16 x 4 x 1 = 64) or even an 8 bit 4 kHz stereo file (8 x 4 x 2). So providing bitdepth and sample frequency is more specific. And, as stated, it is not common practice to give the resolution of PCM files in kbps.
If your phone requires WAV files with slightly different characteristics than these you can adjust the instructions below appropriately. It's assumed you have already made the track mono as per "Convert stereo to mono".
If your phone has these same requirements this should also work for you; if your phone requires MP3 files with slightly different characteristics you can adjust the tutorial instructions below appropriately. It's assumed you have already made the track mono as per "Convert stereo to mono".
The global phone network standardizes on the telephony format (PCM 8 Khz 16 bit, mono). Thus all audio files must be converted to the telephony format in order to be played over the phone. If you have control over the source of the audio recording, it is much better to save the audio to the telephony format directly. This avoids audio conversion, which normally will distort the quality to some degree.
So you use the pulldown at the end of the line to select the format, and then click on 'change' to select the type of recording you want (8k 8-bit mono), and in the 'format settings' box use Change to select something appropriate (which is slightly wrong in that picture!), and then put the extension you want on the end of the file name.
I have an application where I have a digital mic/codec (maxim codec chip) combination transmitting PCM 16-bit samples over I2S to an STM32F217. I am trying to use DMA to capture this audio stream, however I only wish to capture one single channel on I2S. I am using mono mode.
The third main variation on PCM is the number of channels. This is usually either 1 (mono) or 2 (stereo), but you can of course have more (such as 5.1 which is common for movie sound-tracks). The samples for each channel are stored interleaved one after the other, and a pair or group of samples is sometimes referred to as a "frame".
Every decoder has a single preferred PCM output format for a given input type. For example, your MP3 file may natively decode to 44.1kHz stereo 16 bit, and a G.711 file will decode to 8kHz mono 16 bit. If you want floating point output, or 32kHz your decoder might be willing to oblige, but often you have to do that as a separate stage yourself.
Likewise, your encoder is not likely to accept any type of PCM as its input. It will have specific constraints. Usually both mono and stereo are supported, and most codecs are flexible about sample rate. But bit depth will almost always need to be 16 bit. You should also never attempt to change the input format to an encoder midway through encoding a file. Whilst some file formats (e.g. MP3) technically allow sample-rate and channel count to change in the middle of a file, this makes life very difficult for anyone who is trying to play that file.
Probably the easiest change to PCM is modifying the number of channels. To go from mono to stero, you just need to repeat every sample. So for example, if we have a byte array called input, containing 16 bit mono samples, and we want to convert it to stereo, all we need to do is:
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