Hi Clemens,
I am dividing the buffer size into 20 ms, in the given example, I am sending 176 bytes / packet.
I have here another scenario and I hope you can give me some idea again.
I have converted a song using this configuration:
sampling rate = 32000
bit resolution = 16
channel = stereo
after this computation (32000*16*2) * 2 -> 128/packet
I used a speaker which is supporting a minimum frequency of 44100 and maximum of 48000. How can I adjust my streaming size now that the streaming configuration far below than the minimum supported by the speaker.
I attached this speaker in windows and then played the converted file, it is sending 192bytes/packet, and the total bytes transffered had almost trippled the original size of the converted file. Looks like the windows class driver had padded the data, but I dont know how to arrived to the exact number of bytes to be padded to the computed one. I hope I was able to explain my problem well.
Thanks for all the help.
regards,
grace
On 8/12/06, Clemens Ladisch <cladisch@xxxxxxxxxxxx> wrote:
Grace Baldonasa wrote:
> I am half way my development. I can stream music using dynamic settings (any
> speaker plugged in to my host be heard of some music when my application is
> streaming some audio data). However, there are still background noise I am
> hearing. Here is how I computed my streaming size:
>
> sampling rate = 44100
> channel = 2
> bit resolution = 16
> period length = 20 ms
> period size = 3528
bytes
> This period size is used as my buffer size, and this is sliced as it is
> streamed to my device.
> I have not considered any sychronization yet. Here is the speaker
> configuration:
> Synchronization type = adaptive sync
>
> My understanding is no sync pipe is defined by the device, therefore I dont
> need to schedule any sync data, am I correct?
Yes. An adaptive endpoint should be able to work with any sample rate,
as long as it's near 44100.
How do you slice your buffer?
Regards,
Clemens
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