CodeLogic
18th April 2005, 18:51
Hi,
I recently got a second sound device, the M-Audio Audiophile USB, and managed to get it to work (in linux using alsa). I have connected my receiver's digital out to the SPDIF input and have a few questions regarding its capture capabilities:
1. When I tried recording something (using arecord and the default
device), it recorded the data coming into the SPDIF port, which was AC3, as white noise. Now, is there any way to know if 'arecord' (the sound recording app) or the sound card is resampling the SPDIF data?
2. Is there any way to autodetect the sampling rate of the SPDIF stream? Any way to set the clock to sync with the stream on the Audiphile USB?
3. Does capturing of a SPDIF stream using a frequency that is not its
original frequency automatically result in resampling in the soundcard
or by 'arecord'? For that matter, does 'arecord' perform any resampling at all?
4. SPDIF samples or subframes are 32-bits in length. When captured using S16_LE PCM (16-bit little endian PCM), are these samples in its original order/structure? Are they padded/reordered and if so, anyone know how? I know BeSweet can convert these files to AC3 but I'd like to know how that conversion is done.
Thanks a lot for your time!
CodeLogic.
I recently got a second sound device, the M-Audio Audiophile USB, and managed to get it to work (in linux using alsa). I have connected my receiver's digital out to the SPDIF input and have a few questions regarding its capture capabilities:
1. When I tried recording something (using arecord and the default
device), it recorded the data coming into the SPDIF port, which was AC3, as white noise. Now, is there any way to know if 'arecord' (the sound recording app) or the sound card is resampling the SPDIF data?
2. Is there any way to autodetect the sampling rate of the SPDIF stream? Any way to set the clock to sync with the stream on the Audiphile USB?
3. Does capturing of a SPDIF stream using a frequency that is not its
original frequency automatically result in resampling in the soundcard
or by 'arecord'? For that matter, does 'arecord' perform any resampling at all?
4. SPDIF samples or subframes are 32-bits in length. When captured using S16_LE PCM (16-bit little endian PCM), are these samples in its original order/structure? Are they padded/reordered and if so, anyone know how? I know BeSweet can convert these files to AC3 but I'd like to know how that conversion is done.
Thanks a lot for your time!
CodeLogic.