question about sampling rate

question about sampling rate

[Grace Baldonasa]

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Hi all,

I have a very simple question regarding the convertion of sampling rate to
usb full/high speed rate, is this the right mailing list to throw the
question?

any reply will be greatly appreciate.
Thanks.

grace

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Re: question about sampling rate

[Adrian McMenamin]

On Wed, July 12, 2006 9:05 am, Grace Baldonasa wrote:
> Hi all,
>
> I have a very simple question regarding the convertion of sampling rate to
> usb full/high speed rate, is this the right mailing list to throw the
> question?
>
> any reply will be greatly appreciate.
> Thanks.
>
You need to be a bit clearer about what it is you want to do. I assume you
are referring to a sound device.

ALSA will automatically handle the sample rate - converting between the
sample rate of the sound you want to play and the capabilities of your
device as specified by the driver.



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Re: question about sampling rate

[Grace Baldonasa]

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Hi,

Yes I am referring to the audio class driver. I actually would want to
understand about this:

/*
 * convert a sampling rate into USB high speed format (fs/8000 in Q16.16)
 * this will overflow at approx 4 MHz
 */
inline static unsigned get_usb_high_speed_rate(unsigned int rate)
{
 return ((rate << 10) + 62) / 125;
}

and

/*
 * convert a sampling rate into our full speed format (fs/1000 in Q16.16)
 * this will overflow at approx 524 kHz
 */
inline static unsigned get_usb_full_speed_rate(unsigned int rate)
{
 return ((rate << 13) + 62) / 125;
}

I have no clue as to why rate was shifted by 13 then added by 62 then
divided by 125.
I will be writing an embedded linux audio class driver and would like to
simplify the alsa since it is a bit complex and many features may not be
applicable for our use.

Thank you for any help extended.




On 7/12/06, Adrian McMenamin <adrian@newgolddream.dyndns.info> wrote:
>
> On Wed, July 12, 2006 9:05 am, Grace Baldonasa wrote:
> > Hi all,
> >
> > I have a very simple question regarding the convertion of sampling rate
> to
> > usb full/high speed rate, is this the right mailing list to throw the
> > question?
> >
> > any reply will be greatly appreciate.
> > Thanks.
> >
> You need to be a bit clearer about what it is you want to do. I assume you
> are referring to a sound device.
>
> ALSA will automatically handle the sample rate - converting between the
> sample rate of the sound you want to play and the capabilities of your
> device as specified by the driver.
>
>

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Re: question about sampling rate

[Clemens Ladisch]

Grace Baldonasa wrote:
> /*
> * convert a sampling rate into our full speed format (fs/1000 in Q16.16)
> * this will overflow at approx 524 kHz
> */
> inline static unsigned get_usb_full_speed_rate(unsigned int rate)
> {
> return ((rate << 13) + 62) / 125;
> }
> 
> I have no clue as to why rate was shifted by 13 then added by 62 then
> divided by 125.

The parameter is in Hz.  To convert to the number of samples per USB
frame, we have to divide by 1000.  To convert to a 16.16-fixed-point
number, we have to multiply by 2^16.  This would result in "rate *
0x10000 / 1000".  Factoring out 8 yields "rate * 0x2000 / 125".  The
multiplication by a power of two can be replaced by a bitwise left
shift.  The "+62" prevents the division from rounding down.


HTH
Clemens


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Re: question about sampling rate

[Grace Baldonasa]

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Hi,

Thank you so much. Really appreciate this. I have my last clarification to
be made.

After computing the rate, value could be of decimal format, is it rounded
off to either equal or greater than the whole number itself?

/>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
subs->freqmax = subs->freqn + (subs->freqn >> 2); /* max. allowed frequency
*/
Q: why is subs->freqn shifted by 2

 subs->phase = 0;
 /* calculate the max. size of packet */
 maxsize = ((subs->freqmax + 0xffff) * (frame_bits >> 3)) >> 16;

Q: why subs->freqmax is added with 0xffff, then frame_bits is shifted by 3
then the whole thing shifted by 16.

<<<<<<<<<<<<<<<<<<<<<<<<<<<</

I understand that this value is used to determine the maxpacketsize, and I
want to understand why is this so?
I really appreciate any help from you guys.

regards,
grace


On 7/12/06, Clemens Ladisch <clemens@ladisch.de> wrote:
>
> Grace Baldonasa wrote:
> > /*
> > * convert a sampling rate into our full speed format (fs/1000 in Q16.16)
> > * this will overflow at approx 524 kHz
> > */
> > inline static unsigned get_usb_full_speed_rate(unsigned int rate)
> > {
> > return ((rate << 13) + 62) / 125;
> > }
> >
> > I have no clue as to why rate was shifted by 13 then added by 62 then
> > divided by 125.
>
> The parameter is in Hz.  To convert to the number of samples per USB
> frame, we have to divide by 1000.  To convert to a 16.16-fixed-point
> number, we have to multiply by 2^16.  This would result in "rate *
> 0x10000 / 1000".  Factoring out 8 yields "rate * 0x2000 / 125".  The
> multiplication by a power of two can be replaced by a bitwise left
> shift.  The "+62" prevents the division from rounding down.
>
>
> HTH
> Clemens
>

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Re: question about sampling rate

[Clemens Ladisch]

Grace Baldonasa wrote:
> After computing the rate, value could be of decimal format,

In the computer, all values are binary.

> is it rounded off to either equal or greater than the whole number
> itself?

There is no whole number, it is interpreted as a number with 16 binary
fractional digits.

> subs->freqmax = subs->freqn + (subs->freqn >> 2);
> Q: why is subs->freqn shifted by 2

It's just subs->freqn plus 25 percent, and that is just some random
value that should be safe enough for any real sample clock.

> subs->phase = 0;
> /* calculate the max. size of packet */
> maxsize = ((subs->freqmax + 0xffff) * (frame_bits >> 3)) >> 16;
> 
> Q: why subs->freqmax is added with 0xffff, then frame_bits is shifted by 3
> then the whole thing shifted by 16.

It's shifted by 16 to throw away the fractional digits; we calculate
whole bytes.
"frame_bits/8" is the number of bytes per frame.
Adding 0xffff ensures that despite the truncation by the final shift,
the value is rounded up.


HTH
Clemens


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Re: question about sampling rate

[Grace Baldonasa]

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Hi Clemens,

Really appreciate your time answering my question. I have already started my
streaming and is facing another problem now.

Fs = 11025 Hz
Channel  = 2
Bit resolution = 8

so I tried sending my data at (11025/1000) * 2 / ms, I could hear some music
played but with some noise.

I am using Philips Speaker with MaxPackSize = 56, channel = 2, sampling bits
=16, so I decided to change my framesize:

11025 /1000 * 2 * 16/8 / ms -> 44, this time, I cant understand the music
out. The endpoint should still be able to handle it since data rate is
lesser than maxpacksize.

Do you know what went wrong here? hope to receive your advise.
thank...

grace




On 7/15/06, Clemens Ladisch <clemens@ladisch.de> wrote:
>
> Grace Baldonasa wrote:
> > After computing the rate, value could be of decimal format,
>
> In the computer, all values are binary.
>
> > is it rounded off to either equal or greater than the whole number
> > itself?
>
> There is no whole number, it is interpreted as a number with 16 binary
> fractional digits.
>
> > subs->freqmax = subs->freqn + (subs->freqn >> 2);
> > Q: why is subs->freqn shifted by 2
>
> It's just subs->freqn plus 25 percent, and that is just some random
> value that should be safe enough for any real sample clock.
>
> > subs->phase = 0;
> > /* calculate the max. size of packet */
> > maxsize = ((subs->freqmax + 0xffff) * (frame_bits >> 3)) >> 16;
> >
> > Q: why subs->freqmax is added with 0xffff, then frame_bits is shifted by
> 3
> > then the whole thing shifted by 16.
>
> It's shifted by 16 to throw away the fractional digits; we calculate
> whole bytes.
> "frame_bits/8" is the number of bytes per frame.
> Adding 0xffff ensures that despite the truncation by the final shift,
> the value is rounded up.
>
>
> HTH
> Clemens
>

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Re: question about sampling rate

[Clemens Ladisch]

Grace Baldonasa wrote:
> I am using Philips Speaker with MaxPackSize = 56, channel = 2,
> sampling bits =16, so I decided to change my framesize:
> 
> 11025 /1000 * 2 * 16/8 / ms -> 44,

44.1

> this time, I cant understand the music out.

How did you set the sample format?  Are there endpoint-specific
controls?

> The endpoint should still be able to handle it since data rate is
> lesser than maxpacksize.

How do you handle the fractional part of the frame size?

Is this an adaptive endpoint, or what synchronization type does it have?


Regards,
Clemens

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Re: question about sampling rate

[Grace Baldonasa]

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Hi Clemens,

I am not yet handling the fractional part of frame size, would it be cause
of the distorted sound? I was expecting to hear music with some noise
though, but nothing good is coming this time.

The endpoint says, adaptive synchrnonization, but no address is defined in
the syncaddress field of the endpoint descriptor.

Do i have to consider the synchronization before I could  hear some music in
my streamed data?
Thanks for all the help.

regards,

grace

On 7/18/06, Clemens Ladisch <clemens@ladisch.de> wrote:
>
> Grace Baldonasa wrote:
> > I am using Philips Speaker with MaxPackSize = 56, channel = 2,
> > sampling bits =16, so I decided to change my framesize:
> >
> > 11025 /1000 * 2 * 16/8 / ms -> 44,
>
> 44.1


yes you're right.

> this time, I cant understand the music out.
>
> How did you set the sample format?  Are there endpoint-specific
> controls?


no endpoint specific control is set. sample format is based on the max
sampling frequency from the format descriptor.

> The endpoint should still be able to handle it since data rate is
> > lesser than maxpacksize.
>
> How do you handle the fractional part of the frame size?
>
> Is this an adaptive endpoint, or what synchronization type does it have?
>
>
> Regards,
> Clemens
>

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Re: question about sampling rate

[Clemens Ladisch]

Grace Baldonasa wrote:
> I am not yet handling the fractional part of frame size, would it be cause
> of the distorted sound?

There shouldn't be too much noise in this case.

> The endpoint says, adaptive synchrnonization,

Then it should be able to handle a frequency that is 0.25% off.

> >How did you set the sample format?  Are there endpoint-specific
> >controls?
> 
> no endpoint specific control is set. sample format is based on the max
> sampling frequency from the format descriptor.

Did you set the altsetting, or how does the device know about the
current sample format?


Regards,
Clemens

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Re: question about sampling rate

[Grace Baldonasa]

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Hi Clemens,

yes, i did set the alternate setting. The currrent endpoint should be able
to handle the stream data since the maxpacketsize is 56, the max sampling
rate it support is 55kHz, but as I mentioned in my previous email, it's only
noise that I can hear from the speaker.

I used a converted raw data in hex format, I am not really sure what is the
sampling bit, sampling rate or number of channels used when conversion was
made to the raw data, I am in doubt the sampling bit / channels does  not
fit with my experimentations.

Can you recommend some application that will allow me to convert small wav
to hex as array of characters that I could use to stream.

Thank you very much once again,

grace


On 7/18/06, Clemens Ladisch <clemens@ladisch.de> wrote:
>
> Grace Baldonasa wrote:
> > I am not yet handling the fractional part of frame size, would it be
> cause
> > of the distorted sound?
>
> There shouldn't be too much noise in this case.
>
> > The endpoint says, adaptive synchrnonization,
>
> Then it should be able to handle a frequency that is 0.25% off.
>
> > >How did you set the sample format?  Are there endpoint-specific
> > >controls?
> >
> > no endpoint specific control is set. sample format is based on the max
> > sampling frequency from the format descriptor.
>
> Did you set the altsetting, or how does the device know about the
> current sample format?
>
>
> Regards,
> Clemens
>

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Re: question about sampling rate

[Clemens Ladisch]

Grace Baldonasa wrote:
> yes, i did set the alternate setting. The currrent endpoint should be able
> to handle the stream data since the maxpacketsize is 56, the max sampling
> rate it support is 55kHz, but as I mentioned in my previous email, it's only
> noise that I can hear from the speaker.

What is the output of "lsusb -v" for this device?

> I used a converted raw data in hex format, I am not really sure what is the
> sampling bit, sampling rate or number of channels used when conversion was
> made to the raw data, I am in doubt the sampling bit / channels does  not
> fit with my experimentations.
> 
> Can you recommend some application that will allow me to convert small wav
> to hex as array of characters that I could use to stream.

Some Python script like this:

f = open("something.wav", "rb")
while 1:
	c = f.read(1)
	if not c:
		break
	print "0x%02x," % ord(c)
f.close()


HTH
Clemens

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Re: question about sampling rate

[Grace Baldonasa]

[-- Attachment #1.1: Type: text/plain, Size: 1980 bytes --]

Hi Clemens,

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

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
polling interval = 1 ms
refresh = 00
syncaddress = 0

My understanding is no sync pipe is defined  by the device, therefore I dont
need to schedule any sync data, am I correct?
I hope I can  be given some enlightenment in this issue. Any help will be
greatly appreciated.
Thank you very much.

regards,

Grace



On 7/19/06, Clemens Ladisch <clemens@ladisch.de> wrote:
>
> Grace Baldonasa wrote:
> > yes, i did set the alternate setting. The currrent endpoint should be
> able
> > to handle the stream data since the maxpacketsize is 56, the max
> sampling
> > rate it support is 55kHz, but as I mentioned in my previous email, it's
> only
> > noise that I can hear from the speaker.
>
> What is the output of "lsusb -v" for this device?
>
> > I used a converted raw data in hex format, I am not really sure what is
> the
> > sampling bit, sampling rate or number of channels used when conversion
> was
> > made to the raw data, I am in doubt the sampling bit / channels
> does  not
> > fit with my experimentations.
> >
> > Can you recommend some application that will allow me to convert small
> wav
> > to hex as array of characters that I could use to stream.
>
> Some Python script like this:
>
> f = open("something.wav", "rb")
> while 1:
>        c = f.read(1)
>        if not c:
>                break
>        print "0x%02x," % ord(c)
> f.close()
>
>
> HTH
> Clemens
>

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Re: question about sampling rate

[Clemens Ladisch]

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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Re: question about sampling rate

[Grace Baldonasa]

[-- Attachment #1.1: Type: text/plain, Size: 2056 bytes --]

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@fastmail.net> 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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Re: question about sampling rate

[Clemens Ladisch]

Grace Baldonasa wrote:
> I am dividing the buffer size into 20 ms, in the given example, I am
> sending 176 bytes / packet.

This would correspond to a sample rate of 44000 Hz.

You cannot use a constant number of bytes per packet at 44100 Hz.

> 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.

See section 5.12 of the USB 2.0 specification.

> 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.

Apparently, Windows has converted the data to 48 kHz.


Regards,
Clemens

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Re: question about sampling rate

[Grace Baldonasa]

[-- Attachment #1.1: Type: text/plain, Size: 1537 bytes --]

Hi Clemens,

What do you mean when you said Windows had converted it to 48000 Hz, you
mean the audio data itself is converted or just the sampling rate factor in
the computation.

When I took a CATC trace using ALSA version, it is also using a fixed 176
bytes / packet, I am missing something here?  I hope you can help me sort
out my confusion.

Thanks & Best regards,

Grace


On 8/31/06, Clemens Ladisch <cladisch@fastmail.net> wrote:
>
> Grace Baldonasa wrote:
> > I am dividing the buffer size into 20 ms, in the given example, I am
> > sending 176 bytes / packet.
>
> This would correspond to a sample rate of 44000 Hz.
>
> You cannot use a constant number of bytes per packet at 44100 Hz.
>
> > 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.
>
> See section 5.12 of the USB 2.0 specification.
>
> > 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.
>
> Apparently, Windows has converted the data to 48 kHz.
>
>
> Regards,
> Clemens
>

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Re: question about sampling rate

[Clemens Ladisch]

Grace Baldonasa wrote:
> What do you mean when you said Windows had converted it to 48000 Hz, you
> mean the audio data itself is converted or just the sampling rate factor
> in the computation.

Windows likes to convert the data to the highest sample rate supported
by the device so that the audio quality doesn't suffer when a stream
with a higher sample rater must be mixed in later.

> When I took a CATC trace using ALSA version, it is also using a fixed 176
> bytes / packet, I am missing something here?

For a 44.1 kHz stream, snd-usb-audio uses packets with both 176 and 180
bytes.


Regards,
Clemens

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Re: question about sampling rate

[Grace Baldonasa]

[-- Attachment #1.1: Type: text/plain, Size: 762 bytes --]

OK!
Now I understand. Thanks for all the info.

grace


On 8/31/06, Clemens Ladisch <cladisch@fastmail.net> wrote:
>
> Grace Baldonasa wrote:
> > What do you mean when you said Windows had converted it to 48000 Hz, you
> > mean the audio data itself is converted or just the sampling rate factor
> > in the computation.
>
> Windows likes to convert the data to the highest sample rate supported
> by the device so that the audio quality doesn't suffer when a stream
> with a higher sample rater must be mixed in later.
>
> > When I took a CATC trace using ALSA version, it is also using a fixed
> 176
> > bytes / packet, I am missing something here?
>
> For a 44.1 kHz stream, snd-usb-audio uses packets with both 176 and 180
> bytes.
>
>
> Regards,
> Clemens
>

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Re: question about sampling rate

[Grace Baldonasa]

[-- Attachment #1.1: Type: text/plain, Size: 1225 bytes --]

Hi Clemens,

Got another question, following to my previos emails, do I have to adjust my
streaming size since the sampling freq used when audio data was converted is
not supported by the speaker? if I do adjust the streaming size, I am
compromising the bit resolution and the sampling rate already?

Thanks in advance.

regards,
grace


On 8/31/06, Grace Baldonasa <gbaldona@gmail.com> wrote:
>
>  OK!
> Now I understand. Thanks for all the info.
>
> grace
>
>
>  On 8/31/06, Clemens Ladisch <cladisch@fastmail.net> wrote:
>
> > Grace Baldonasa wrote:
> > > What do you mean when you said Windows had converted it to 48000 Hz,
> > you
> > > mean the audio data itself is converted or just the sampling rate
> > factor
> > > in the computation.
> >
> > Windows likes to convert the data to the highest sample rate supported
> > by the device so that the audio quality doesn't suffer when a stream
> > with a higher sample rater must be mixed in later.
> >
> > > When I took a CATC trace using ALSA version, it is also using a fixed
> > 176
> > > bytes / packet, I am missing something here?
> >
> > For a 44.1 kHz stream, snd-usb-audio uses packets with both 176 and 180
> > bytes.
> >
> >
> > Regards,
> > Clemens
> >
>
>
>

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Re: question about sampling rate

[Clemens Ladisch]

Grace Baldonasa wrote:
> Got another question, following to my previos emails, do I have to adjust my
> streaming size since the sampling freq used when audio data was converted is
> not supported by the speaker?

ALSA will allow only those sample rates/formats that the driver says it
supports (in the snd_pcm_hardware structure, and with any constraints
set in the open callback).


HTH
Clemens

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Re: question about sampling rate

[Grace Baldonasa]

[-- Attachment #1.1: Type: text/plain, Size: 1208 bytes --]

Hi,

Do you know of any means how to test the streaming IN raw data. I am
attaching a headset and could only test the speaker or streaming OUT. I need
to test as well the streaming IN.

I am writing every single byte I received into a buffer, and after some
period,  I am streaming it out again, but I can only hear noise and no
clarity in the quality of the audio captured.

My captured settings are as follows:
Sampling Freq = 48000
Channel = 1
Sampling Bit = 16

when i streamed it out, the playback settings are as follows:
 Sampling Freq = 48000
Channel = 2
Sampling Bit = 16

I forcefully change the channel number to 2, would this cause any problem?
Any advise can be deeply appreciated.
Thanks.

regards,

Grace



On 8/31/06, Clemens Ladisch <cladisch@fastmail.net> wrote:
>
> Grace Baldonasa wrote:
> > Got another question, following to my previos emails, do I have to
> adjust my
> > streaming size since the sampling freq used when audio data was
> converted is
> > not supported by the speaker?
>
> ALSA will allow only those sample rates/formats that the driver says it
> supports (in the snd_pcm_hardware structure, and with any constraints
> set in the open callback).
>
>
> HTH
> Clemens
>

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Re: question about sampling rate

[Grace Baldonasa]

[-- Attachment #1.1: Type: text/plain, Size: 1660 bytes --]

Hi Clemens,

Can I directly stream the recorded data as playback without going for any
conversion or anything to the captured data?

I am hearing pure  noise if I am streaming back directly the recorded sound
that I captured. Is this an expected behaviour?

Really appreciate any advice.

Thanks,

grace


On 9/19/06, Grace Baldonasa <gbaldona@gmail.com> wrote:
>
> Hi,
>
> Do you know of any means how to test the streaming IN raw data. I am
> attaching a headset and could only test the speaker or streaming OUT. I need
> to test as well the streaming IN.
>
> I am writing every single byte I received into a buffer, and after some
> period,  I am streaming it out again, but I can only hear noise and no
> clarity in the quality of the audio captured.
>
> My captured settings are as follows:
> Sampling Freq = 48000
> Channel = 1
> Sampling Bit = 16
>
> when i streamed it out, the playback settings are as follows:
>  Sampling Freq = 48000
> Channel = 2
> Sampling Bit = 16
>
> I forcefully change the channel number to 2, would this cause any problem?
> Any advise can be deeply appreciated.
> Thanks.
>
> regards,
>
> Grace
>
>
>
> On 8/31/06, Clemens Ladisch <cladisch@fastmail.net> wrote:
> >
> > Grace Baldonasa wrote:
> > > Got another question, following to my previos emails, do I have to
> > adjust my
> > > streaming size since the sampling freq used when audio data was
> > converted is
> > > not supported by the speaker?
> >
> > ALSA will allow only those sample rates/formats that the driver says it
> > supports (in the snd_pcm_hardware structure, and with any constraints
> > set in the open callback).
> >
> >
> > HTH
> > Clemens
> >
>
>

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Re: question about sampling rate

[Clemens Ladisch]

Grace Baldonasa wrote:
> Can I directly stream the recorded data as playback without going for any
> conversion or anything to the captured data?

Only if both streams use exactly the same sample format/rate.

> I am hearing pure  noise if I am streaming back directly the recorded sound
> that I captured. Is this an expected behaviour?

That may be a different sample format, or possibly just wrong data.

How does the recorded data look like?  Do you have a file or a hexdump?


Regards,
Clemens

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