G723 Codec Audio Problem

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One more thing,
since coded frame size calculation (used for frame size allocation in
jitter buffer, etc) involves only integer that is hurting of decimal
truncation, it is important to change this as well:
attr->info.avg_bps = 6300;
to an adequate bigger value like 6400.

Regards,
nanang


On 17/03/2008, Nanang Izzuddin <nanang.izzuddin at gmail.com> wrote:
> Please change:
>
> attr->info.pcm_bits_per_sample = 24;
>
> with
>  attr->info.pcm_bits_per_sample = 16;
>
>  and please make sure the active bitrate used by your g723.1 codec
>  implementation is really 6.3 kbps.
>
>  Regards,
>  nanang
>
>
>
>  On 17/03/2008, sre kdkjf <kk_kksri at yahoo.com> wrote:
>  > Hi All
>  >
>  > i am trying to include G723 codec in PJSIP Stack.
>  >
>  > i had included G723 codec details in the stack, and in rtp also.
>  >
>  > the problem i am getting is when i am in call, in the rtp also G723.1 is
>  > flowing from one end to end.  But there is lot of noise disturbence. i.e.
>  > lot of distortion is coming.so that both users not able to hear the audio
>  > properly.
>  >
>  > the default attribute which i have included for G723 codec is as given
>  > below.
>  >
>  >
>  >   PJ_UNUSED_ARG(factory);
>  >     PJ_UNUSED_ARG(id);
>  >     pj_bzero(attr, sizeof(pjmedia_codec_param));
>  >     attr->info.clock_rate = 8000;
>  >     attr->info.channel_cnt = 1;
>  >     attr->info.avg_bps = 6300;
>  >     attr->info.pcm_bits_per_sample = 24;
>  >     attr->info.frm_ptime = 30;
>  >     attr->info.pt = PJMEDIA_RTP_PT_G723;
>  >     attr->setting.frm_per_pkt = 1;
>  >     attr->setting.vad = 1;
>  > #if !PLC_DISABLED
>  >     attr->setting.plc = 1;
>  > #endif
>  >
>  > so is there any changes i have to make in the source.
>  >
>  > so can you please tell me how to reduce the distortion in audio i.e. how to
>  > reduce noise in audio, so that both parties can communicate can hear audio
>  > clearly.
>  >
>  > if any body solved this problem it would be great helpful  for me.
>  >
>  > Thankyou.
>  >
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>
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