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[jamescowdery]

I have an application where I need fast gapless switching between tracks when playing out over HDMI. JRiver attracted me because there are so many options available to change the track transitional behavior. When the sampling rate is the same the transition is pretty quick, as you would expect. When the sampling rate changes (48 to 192kHz and back) there is delay of about .85s to 1s. This is in addition to any downstream device delay due to locking to the new clock which I have compensated out of the measurement. I've tried to reduce this by playing with options but without any luck. I was wondering if anyone had any ideas?

I'm playing wave files in a playlist using WASAPI.

Cheers,
James

[kr4]

Set MC to output the same format all the time.

[AndrewFG]

Try applying a DSP resampling effect so that all source formats are re sampled to the same output sample rate..

[jamescowdery]

Thanks. Makes sense. I failed to mention that I also need bit perfect audio because I am carrying encoded audio within wav files. This means that I can't do the resampling which I think both of you are suggesting.

Any ideas that preserve the bit transparency (support for bit-perfect audio is the reason I picked jriver).
I understand output sampling rate changes are challenging and can't happen instantly but any tips on how to minimize the time would be appreciated. I changed everything I could see related to gaps to zero and reduced buffer sizes to minimum.

Thanks,
James

[blgentry]

Thanks. Makes sense. I failed to mention that I also need bit perfect audio because I am carrying encoded audio within wav files.

So DTS encoded CDs or something?  If you have files like this, you should convert them to FLAC.

If it's not DTS in a WAV container, what types of files are you talking about?

Brian.

[jamescowdery]

This is for a test application driving TVs and AVRs. The audio is Dolby Digital and Dolby Digital Plus encapsulated within a wavefile. Dolby Digital uses 48kHz whereas Dolby Digital Plus uses 192kHz hence the need for sampling rate changes.
Yes I know. I'm not your typical user.
James

[Hendrik]

Changing the sample rate (or channel count for that matter) requires draining all buffers first and then switching the device over. Its never going to be gapless (especially over HDMI, where even the driver can take a bit to switch rates), sorry.
You can possibly reduce the delay slightly if you reduce all buffers to the minimum where playback is still flawless.

[jamescowdery]

Henrik,
Thanks. This all makes sense. I have a method where I can measure the delay and compensate for it by putting markers in the content. The delay seems fairly consistent which helps.
James

[jamescowdery]

@Henrik - I got round the delay problem through compensation and got quite far with my application. I've run into another problem which is that I need to change the audio/non-audio bit in the outgoing iec stream. Most devices (TVs/AVRS etc.) are insensitive to this bit so I got quite far with it set incorrectly but I've found some products that require this bit to set correctly to non-audio when sending AC-4 etc. In Linux/ALSA setting you can set this bit with the 'iecset audio 0' command. Obviously your application is assuming linear PCM and so there is no need to provide the option to set this bit.

Can you think of anyway round this? Obviously this has to be on a file by file basis so I'd have to set something in the wavefile header that flips this bit in the output stream. I've looked at the specs and the non-audio options are for a-law and u-law etc. which are not relevant here. This is a deal breaker for me so unfortunately I'm now considering modifying a custom player which is a lot of work  .
Any suggestions are appreciated.