[pjsip] A couple of questions.

huali at cis.strath.ac.uk huali at cis.strath.ac.uk
Sat Apr 12 19:07:42 EDT 2008


Hi Nanang,

Firstly thanks a lot for all the help you've given us - got the latest
build from svn and it worked first time.  For now at least, the conference
bridge seems to be the way to go.  In fact, it actually works better than
we expected, so thumbs up.

About the gmail thing, there definitely is a problem.  I tried replying to
your last email from my gmail account and it didn't work.  Both of us
signed up to the list using our gmail accounts in exactly the same way
that hussain did with this account so I don't know what's wrong.  We can
both receive from the list just not send.

Anyway, it's not a major problem - just a bit annoying.

Cheers,
Adam

> Hi Hussain,
>
> It seems you set the wav player port setting independently from stream
> port.
>
> Master port requires both ports to have same clock rate and ptime
> (packet time/frame length in millisecond unit).  In this case, you
> have put resample to adjust the clock rate of wav player port, but you
> haven't synchronized the ptime. Since usually codecs (managed by
> stream port) are strict on ptime, so the wav player should follow the
> stream port's ptime.
>
> Please take a look at this quote from streamutil.c:
> ---------
> 	wav_ptime = stream_port->info.samples_per_frame * 1000 /
> 		    stream_port->info.clock_rate;
> 	status = pjmedia_wav_player_port_create(pool, play_file, wav_ptime,
> 						0, -1, &play_file_port);
> ---------
>
> As alternative, you can utilize the conference bridge for this purpose
> and it provides convenience by automatic adjustment on the clock rate
> and ptime (and also mixing signal, etc) for all ports plugged in to
> it.
>
> Btw, about the wrong wav format, it may be caused by compressed wav
> data, currently wav player port only supports uncompressed data.
>
> Regards,
> nanang
>
>
> On 10/04/2008, huali at cis.strath.ac.uk <huali at cis.strath.ac.uk> wrote:
>> I am having some trouble with WAV file compatibility.
>>
>>  Basically, I have a PCM 44.1 kHz wav file that I am trying to stream
>> using
>>  RTP.  Having looked at some samples, it looks to me like I should be
>>  setting up a number of ports like so:
>>
>>  wav_player_port->master_port->stream_port
>>
>>  The problem I have been having is to do with the sample rate and the
>>  number of samples per frame.  I tried adding a resample port between
>> the
>>  wav_player_port and the master_port, which successfully resamples the
>>  audio to 8kHz, but leaves us with 320 samples per frame, which the
>>  master_port does not like because the stream_port has around 160
>> samples
>>  per frame (can't remember if the numbers are correct, but you get the
>>  idea).
>>
>>  I tried resampling the wav file using nero wav editor, but all that
>>  happens is that the wav_player_port then complains that the file is in
>> the
>>  wrong format.
>>
>>  Can anyone suggest a way around this?
>>
>>  Thanks alot,
>>
>>  Hussain
>>
>>
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>>
>
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