Tim Deveaux 11-Sep-16 16:54 11-Sep-16 16:54 Excellent article! I'm wondering if you have in your travels seen any information on how to use the MFPKEY_WMRESAMP_LOWPASS_BANDWIDTH property of the resampler transform. I've been trying to resample 48000 down to 44100 with pretty noisy results, and that seems to be the only thing left to poke. The actual need is to go from 48000 or 44100 up/down to/from 8k 16 bit (phone quality) and results there might be passable (I realize 48000 to 44100 is a tough one) but would be nice to understand this a bit better. Heto Power Dry Manual. Any clues that don't involve a treatise on decimation and interpolation appreciated Failing that, would you think there's any advantage in going 48000 to/from 8000 rather than 44100 - in terms of processing time?

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Seems to be a better ratio (?) Thanks. Mikhail0 16-May-15 11:01 16-May-15 11:01 Hi Mark, I thought I'd rather separate the issues I have an NAudio client, which gets a WaveStream as input (based on the type returned by e.g. I want to be able to set volume and balance for the output. I probably could do it by combining two StereoToMono converters (each one producing output for one channel, then somehow mix them), although I'm not sure how exactly this is to be done. Install Activex Webrec.cab. Another way I see is to create my own ISoundProvider for the purpose, but here I am not clear how I go from WaveStream to ISoundProvider and then back to WaveStream (as I use e.g.

WaveStream.CurrentTime property, which does not exit in ISoundProvider). Could you give me a hint on the implementation? This should be a trivial task, so I probably just got lost. Lg Gp08 Linux Driver. Member 10770565 23-Apr-14 14:08 23-Apr-14 14:08 Matt, I've been using NAudio to convert WAV files to MP3 with some success. However, I've come across a problem that I hope you can help me with.

I have some test audio files that when I use NAudio to convert to MP3 it seems not all of the compressed audio is written out. In my tests, it shows that I lose a second of audio according to my player and I can indeed hear that the WAV is slightly longer. What I've tried so far: I.

Inside MediaFoundationEncoder.ConvertOneBuffer I detect when a 'partial' buffer will be used to generate a sample and pad the remaining buffer with 0's. It seems to write out the entirety of the audio data plus a section of silence. I can confirm with an audible test as well as an inspection of the binary data against the 'truncated' MP3 (I've seen up to 892 extra non-zero bytes get written out when I pad).