Hi,

i made a media player in DirectShow that captures PCM audio frames from mp3 audio files.

I need to send this audio frames to PJSIP. Can anyone help me?

I have tried adding a passive_port to the conference bridge and it did not worked.

Now i am trying to use splitcomb. And i can send some noise.

Here's a code sample:

// 1º pjmedia_splitcom_create
pj_status_t status = pjmedia_splitcomb_create(pool_vars, nSamplesPerSec, nChannels, samplesPerFrame, wBitsPerSample, 0, &p_splitcomb);
// for each channel
for(unsigned int i = 0; i < nChannels; i++)  {
unsigned int p_slot;
pjmedia_port* p_port;

 // 2º pjmedia_splitcom_create_rev_channel
 status = pjmedia_splitcomb_create_rev_channel(pool_vars, p_splitcomb, i, 0, &p_port);

 // 3º pjmedia_conf_add_port
 status = pjmedia_conf_add_port(conference_bridge, pool_vars, p_port, NULL, &p_slot);

}

And then, when i want to send an audio frame i do the following:

pjmedia_port_put_frame(p_splitcomb, &frame);

What is wrong with my code?
Am i using splitcom correctly?

Any help would be appreciated...

Rafael Maia

Mais do que emails! Confira tudo o que Windows Live™ pode oferecer.
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How does APS implementation works in Symbian_ua?

Does g711,gsm,speex all works fine?

APS->G711->PCM->gsm

Or

APS->gsm??

Does it do the transcoding or it just only works for g711 and not gsm and
speex?

thanks

Thanks for your anwser Norman.

You are using a very diferent approach from mine. Could you provide some more information ( or some source code ) ?

My DirectShow sound reading routines are pushing audio frames that i need to send throw PJSIP.

So i need to call put_frame on a media port.
How did you created this "Passive" port ?

You also add it to the conference bridge, right ?

Thanks in advance...

Rafael Maia

From: norman@myasd.comTo: pjsip@lists.pjsip.orgDate: Fri, 9 Jan 2009 12:17:19 -0500Subject: Re: [pjsip] Sending my own audio framesI use pjmedia_port_info_init to create a player port. Something like this:

pool = pjsua_pool_create("MyPlayerPool", 4000, 4000);
pj_strdup2(pool, &base.info.name, "MyPlayer");

base.get_frame = &file_get_frame_get;
base.put_frame = &file_put_frame_get;
base.on_destroy = &file_on_destroy_get;

pjmedia_port_info_init(&base.info, &name, SIGNATURE, sampling_rate, channel_count, bits_per_sample, samples_per_frame);

This works well for playing files in a non-standard format that I need to support. You put your sample generation code in file_get_frame_get.

Norman Franke
Answering Service for Directors, Inc.
www.myasd.com

On Jan 9, 2009, at 9:24 AM, Rafael Maia wrote:

Hi, i made a media player in DirectShow that captures PCM audio frames from mp3 audio files. I need to send this audio frames to PJSIP. Can anyone help me? I have tried adding a passive_port to the conference bridge and it did not worked. Now i am trying to use splitcomb. And i can send some noise. Here's a code sample: // 1º pjmedia_splitcom_createpj_status_t status = pjmedia_splitcomb_create(pool_vars, nSamplesPerSec, nChannels, samplesPerFrame, wBitsPerSample, 0, &p_splitcomb);// for each channelfor(unsigned int i = 0; i < nChannels; i++)  {    unsigned int p_slot;    pjmedia_port* p_port;      // 2º pjmedia_splitcom_create_rev_channel    status = pjmedia_splitcomb_create_rev_channel(pool_vars, p_splitcomb, i, 0, &p_port);      // 3º pjmedia_conf_add_port    status = pjmedia_conf_add_port(conference_bridge, pool_vars, p_port, NULL, &p_slot);}  And then, when i want to send an audio frame i do the following: pjmedia_port_put_frame(p_splitcomb, &frame);  What is wrong with my code?Am i using splitcom correctly? Any help would be appreciated... Rafael Maia

O jeito mais fácil de manter a sua lista de amigos sempre em ordem! Organize seus contatos!_______________________________________________Visit our blog: http://blog.pjsip.orgpjsip mailing listpjsip@lists.pjsip.orghttp://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

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Well what I want to use my own implementation of G729. I am interested in APS only to use earpiece in place of loud speaker. Do I need to do anything else other than just enable APS? Does built in GSM and Speex of sumbian_ua working with APS?

From: pjsip-bounces@lists.pjsip.org [mailto:pjsip-bounces@lists.pjsip.org] On Behalf Of Pai Peng
Sent: Saturday, January 10, 2009 12:19 AM
To: pjsip list
Subject: Re: [pjsip] APS Implementation

Hi,

in source symbian_sound_aps.cpp:

void CPjAudioEngine::RecCb(TAPSCommBuffer &buffer)
{
...
buf[i] = pjmedia_alaw2linear(iBuffer[i]);
...
}

TInt CPjAudioEngine::StartStreamL()
{
....
iSession.SetEncoderMode(EALawOr20ms);
iSession.SetDecoderMode(EALawOr20ms);
...
}

audio is in G711-A Format.

The  APS supports the following codecs: AMR, G711, G729 and iLBC.

If you like use other codec for encoding or decoding, you should change the SetEncoderMode() and SetDecoderMode().

Pai

On Fri, Jan 9, 2009 at 5:54 PM, Rawshan Iajdani iajdani@provati.com wrote:

How does APS implementation works in Symbian_ua?

Does g711,gsm,speex all works fine?

APS->G711->PCM->gsm

Or

APS->gsm??

Does it do the transcoding or it just only works for g711 and not gsm and speex?

thanks

Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

Is there anyway I could test APS without a Publisher ID?

I don't think you can with this current wrapper.

So far I understood, the current wrapper just take PCM and convert that to
g.711 and pass it to APS and vice versa. You still need your own g.729
implementation. This wrapper just take PCM as input and output..

Correct me if I were wrong..

From: pjsip-bounces@lists.pjsip.org [mailto:pjsip-bounces@lists.pjsip.org]
On Behalf Of S. M. Nazmul Hasan (Opu)
Sent: Sunday, January 11, 2009 3:36 PM
To: pjsip list
Subject: Re: [pjsip] APS Implementation

Hi Pai,

Can you please tell me if i want to g729 from Aps what should i need to do?

in the line of encoder mode

iSession.SetEncoderMode(EALawOr20ms);

is for g711 but i saw no encoding mode for g729 . should i block those line
if i want to use g729?

Thanks

On Sat, Jan 10, 2009 at 12:19 AM, Pai Peng sipaipv6@gmail.com wrote:

Hi,

in source symbian_sound_aps.cpp:

void CPjAudioEngine::RecCb(TAPSCommBuffer &buffer)
{
...
buf[i] = pjmedia_alaw2linear(iBuffer[i]);
...
}

TInt CPjAudioEngine::StartStreamL()
{
....
iSession.SetEncoderMode(EALawOr20ms);
iSession.SetDecoderMode(EALawOr20ms);
...
}

audio is in G711-A Format.

The  APS supports the following codecs: AMR, G711, G729 and iLBC.

If you like use other codec for encoding or decoding, you should change the
SetEncoderMode() and SetDecoderMode().

Pai

On Fri, Jan 9, 2009 at 5:54 PM, Rawshan Iajdani iajdani@provati.com wrote:

How does APS implementation works in Symbian_ua?

Does g711,gsm,speex all works fine?

APS->G711->PCM->gsm

Or

APS->gsm??

Does it do the transcoding or it just only works for g711 and not gsm and
speex?

thanks

Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

--
S. M. Nazmul Hasan Opu
Software Engineer
R & D Application
Dhaka, Bangladesh
Mob: +880 1712 901 764

Well its nothing to do with APS. I was talking about the wrapper. If you
want to implement the  g729 you need to rewrite the wrapper itself and some
changes should be made in the PJMEDIA codec itself.

This wrapper does not pass g.711 directly to the PJMEDIA rather convert that
to PCM and pass that. If you even use g729 of APS, you still need to convert
that to PCM before passing that to PJMEDIA. PJMEDIA will again have to
convert this PCM to g.729.

So what is the use????

From: pjsip-bounces@lists.pjsip.org [mailto:pjsip-bounces@lists.pjsip.org]
On Behalf Of S. M. Nazmul Hasan (Opu)
Sent: Sunday, January 11, 2009 7:01 PM
To: pjsip list
Subject: Re: [pjsip] APS Implementation

I dont think APS implementation(G729) is so hard as you think. APS already
has G729 codec support that is if i use APS i must get the implementation of
g729 innerly . i just have to change the procedure to use g729 codec....like
some parameter and encoding decoding technique. ...

APS use 24byte data ...thats a point also...

On Sun, Jan 11, 2009 at 3:46 PM, Rawshan Iajdani iajdani@provati.com
wrote:

I don't think you can with this current wrapper.

So far I understood, the current wrapper just take PCM and convert that to
g.711 and pass it to APS and vice versa. You still need your own g.729
implementation. This wrapper just take PCM as input and output..

Correct me if I were wrong..

From: pjsip-bounces@lists.pjsip.org [mailto:pjsip-bounces@lists.pjsip.org]
On Behalf Of S. M. Nazmul Hasan (Opu)
Sent: Sunday, January 11, 2009 3:36 PM

To: pjsip list
Subject: Re: [pjsip] APS Implementation

Hi Pai,

Can you please tell me if i want to g729 from Aps what should i need to do?

in the line of encoder mode

iSession.SetEncoderMode(EALawOr20ms);

is for g711 but i saw no encoding mode for g729 . should i block those line
if i want to use g729?

Thanks

On Sat, Jan 10, 2009 at 12:19 AM, Pai Peng sipaipv6@gmail.com wrote:

Hi,

in source symbian_sound_aps.cpp:

void CPjAudioEngine::RecCb(TAPSCommBuffer &buffer)
{
...
buf[i] = pjmedia_alaw2linear(iBuffer[i]);
...
}

TInt CPjAudioEngine::StartStreamL()
{
....
iSession.SetEncoderMode(EALawOr20ms);
iSession.SetDecoderMode(EALawOr20ms);
...
}

audio is in G711-A Format.

The  APS supports the following codecs: AMR, G711, G729 and iLBC.

If you like use other codec for encoding or decoding, you should change the
SetEncoderMode() and SetDecoderMode().

Pai

On Fri, Jan 9, 2009 at 5:54 PM, Rawshan Iajdani iajdani@provati.com wrote:

How does APS implementation works in Symbian_ua?

Does g711,gsm,speex all works fine?

APS->G711->PCM->gsm

Or

APS->gsm??

Does it do the transcoding or it just only works for g711 and not gsm and
speex?

thanks

Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

--
S. M. Nazmul Hasan Opu
Software Engineer
R & D Application
Dhaka, Bangladesh
Mob: +880 1712 901 764

Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

--
S. M. Nazmul Hasan Opu
Software Engineer
R & D Application
Dhaka, Bangladesh
Mob: +880 1712 901 764

Thanks alot for your source code Norman, it has been a big boost to my code development...

I am having a diferent problem now.
I can ear the sound frames i added on my sound speakers, but i am not sending them in the RTP session.
I did the following:

pjmedia_conf_add_port(conference_bridge, pool_vars, &port, NULL, &slot);
pjmedia_conf_connect_port(conference_bridge, slot, inSlot, 0);

I have tested putting frames on the master_port directly, and the saound frames are sent throw RTP:

pjmedia_port* master = pjmedia_conf_get_master_port(conference_bridge);
pjmedia_port_put_frame(master, &frame);

...but i don't think this is the proper way to do it

Can anyone help?
How should i send the audio frames?

Thanks in advance for any help....

Rafael Maia

From: norman@myasd.comTo: pjsip@lists.pjsip.orgDate: Fri, 9 Jan 2009 18:48:10 -0500Subject: Re: [pjsip] Sending my own audio framesYou can use pjmedia_splitcomb_create_rev_channel to reverse the put/get semantics. I do that elsewhere and it works.

I did then add it to the conference bridge.

Sadly, my code isn't easy to extract and is very C++ using boost and a custom string library.

typedef bool (*GetSamplesPtr)(void * inBuf, uint32_t * ioByteSize, pj_timestamp * inTime);

PlayPort::PlayPort(
unsigned sampling_rate,
unsigned channel_count,
unsigned samples_per_frame,
unsigned bits_per_sample
)
: pool(NULL), mCallIndex(INVALID_CALL_INDEX)
{
const pj_str_t name = pj_str("file");

pool = pjsua_pool_create("asd_player", 4000, 4000);
assert(pool);

pj_strdup2(pool, &base.info.name, "asd_player");

base.get_frame = &file_get_frame_get;
base.put_frame = &file_put_frame_get;
base.on_destroy = &file_on_destroy_get;

pjmedia_port_info_init(&base.info, &name, SIGNATURE, sampling_rate, channel_count, bits_per_sample, samples_per_frame);

port_is_open = true;
options = 0;

}

PlayPort::~PlayPort()
{
Close();

if (pool) {
    pj_pool_release(pool);
    pool = NULL;
}

}

pj_status_t PlayPort::Open(SInt32 inCallIndex, float inVolume, GetSamplesPtr inFunc)
{
pj_status_t status;

status = pjsua_conf_add_port(pool, &base, &slot);
if (status != PJ_SUCCESS) {
    pjmedia_port_destroy(&base);
    return status;
}

mCallIndex = inCallIndex;

status = pjsua_conf_adjust_rx_level(slot, inVolume);

port_func = inFunc;

return status;

}

void PlayPort::Close(void)
{
file_on_destroy_get(&base);
}

bool PlayPort::IsOpen(void)
{
return port_is_open && slot >= 0;
}

pj_status_t PlayPort::ConnectTo(int inSlot)
{
return pjsua_conf_connect(slot, inSlot);
}

pj_status_t PlayPort::DisconnectFrom(int inSlot)
{
return pjsua_conf_disconnect(slot, inSlot);
}

// Class static
pj_status_t PlayPort::file_on_destroy_get(pjmedia_port *this_port)
{
// As the first element in a non-virtual class...
PlayPort *fport = (PlayPort *) this_port;
boost::recursive_mutex::scoped_lock l(fport->m);

pj_assert(this_port->info.signature == SIGNATURE);

if (fport->port_is_open) {
 fport->port_is_open = false; // Prevents a recursive callback

    fport->port_func = NULL;
    pjsua_conf_remove_port(fport->slot); // If it's already removed, e.g. audio device changed, this crashes. 
    fport->slot = 0;
    pjmedia_port_destroy(&fport->base);
}

return PJ_SUCCESS;

}

// Class static
pj_status_t PlayPort::file_get_frame_get(pjmedia_port *this_port, pjmedia_frame *frame)
{
// As the first element in a non-virtual class...
PlayPort *fport = (PlayPort *) this_port;

pj_assert(this_port->info.signature == SIGNATURE);

if (frame->size == 0) return PJ_SUCCESS;
if (! fport->port_func) return PJ_EEOF;

pj_size_t frame_size = frame->size;
if ((*fport->port_func)(frame->buf, &frame_size, (PJSUA_Timestamp *) &frame->timestamp)) {
    if (frame_size < frame->size) {
        bzero((UInt8*)frame->buf + frame_size, frame->size - frame_size);
    }
    frame->timestamp.u64 = 0;
    frame->type = PJMEDIA_FRAME_TYPE_AUDIO;
    return PJ_SUCCESS;
}
file_on_destroy_get(this_port);
return PJ_EEOF;

}

// Class static
pj_status_t PlayPort::file_put_frame_get(pjmedia_port *this_port, const pjmedia_frame *frame)
{
return PJ_EINVALIDOP;
}

Norman Franke
Answering Service for Directors, Inc.
www.myasd.com

On Jan 9, 2009, at 1:27 PM, Rafael Maia wrote:

Thanks for your anwser Norman. You are using a very diferent approach from mine. Could you provide some more information ( or some source code ) ? My DirectShow sound reading routines are pushing audio frames that i need to send throw PJSIP. So i need to call put_frame on a media port.How did you created this "Passive" port ? You also add it to the conference bridge, right ? Thanks in advance... Rafael Maia

From: norman@myasd.comTo: pjsip@lists.pjsip.orgDate: Fri, 9 Jan 2009 12:17:19 -0500Subject: Re: [pjsip] Sending my own audio framesI use pjmedia_port_info_init to create a player port. Something like this:

pool = pjsua_pool_create("MyPlayerPool", 4000, 4000);
pj_strdup2(pool, &base.info.name, "MyPlayer");

base.get_frame = &file_get_frame_get;
base.put_frame = &file_put_frame_get;
base.on_destroy = &file_on_destroy_get;

pjmedia_port_info_init(&base.info, &name, SIGNATURE, sampling_rate, channel_count, bits_per_sample, samples_per_frame);

This works well for playing files in a non-standard format that I need to support. You put your sample generation code in file_get_frame_get.

Norman Franke
Answering Service for Directors, Inc.
www.myasd.com

On Jan 9, 2009, at 9:24 AM, Rafael Maia wrote:

Hi, i made a media player in DirectShow that captures PCM audio frames from mp3 audio files. I need to send this audio frames to PJSIP. Can anyone help me? I have tried adding a passive_port to the conference bridge and it did not worked. Now i am trying to use splitcomb. And i can send some noise. Here's a code sample: // 1º pjmedia_splitcom_createpj_status_t status = pjmedia_splitcomb_create(pool_vars, nSamplesPerSec, nChannels, samplesPerFrame, wBitsPerSample, 0, &p_splitcomb);// for each channelfor(unsigned int i = 0; i < nChannels; i++)  {    unsigned int p_slot;    pjmedia_port* p_port;      // 2º pjmedia_splitcom_create_rev_channel    status = pjmedia_splitcomb_create_rev_channel(pool_vars, p_splitcomb, i, 0, &p_port);      // 3º pjmedia_conf_add_port    status = pjmedia_conf_add_port(conference_bridge, pool_vars, p_port, NULL, &p_slot);}  And then, when i want to send an audio frame i do the following: pjmedia_port_put_frame(p_splitcomb, &frame);  What is wrong with my code?Am i using splitcom correctly? Any help would be appreciated... Rafael Maia

O jeito mais fácil de manter a sua lista de amigos sempre em ordem! Organize seus contatos!_______________________________________________Visit our blog: http://blog.pjsip.orgpjsip mailing listpjsip@lists.pjsip.orghttp://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

Notícias direto do New York Times, gols do Lance, videocassetadas e muitos outros vídeos no MSN Videos! Confira já! _______________________________________________Visit our blog: http://blog.pjsip.orgpjsip mailing listpjsip@lists.pjsip.orghttp://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

Mais do que emails! Confira tudo o que Windows Live™ pode oferecer.
http://www.microsoft.com/windows/windowslive/

Opu,

You live Dhaka??

From: pjsip-bounces@lists.pjsip.org [mailto:pjsip-bounces@lists.pjsip.org]
On Behalf Of S. M. Nazmul Hasan (Opu)
Sent: Sunday, January 11, 2009 7:01 PM
To: pjsip list
Subject: Re: [pjsip] APS Implementation

I dont think APS implementation(G729) is so hard as you think. APS already
has G729 codec support that is if i use APS i must get the implementation of
g729 innerly . i just have to change the procedure to use g729 codec....like
some parameter and encoding decoding technique. ...

APS use 24byte data ...thats a point also...

On Sun, Jan 11, 2009 at 3:46 PM, Rawshan Iajdani iajdani@provati.com
wrote:

I don't think you can with this current wrapper.

So far I understood, the current wrapper just take PCM and convert that to
g.711 and pass it to APS and vice versa. You still need your own g.729
implementation. This wrapper just take PCM as input and output..

Correct me if I were wrong..

From: pjsip-bounces@lists.pjsip.org [mailto:pjsip-bounces@lists.pjsip.org]
On Behalf Of S. M. Nazmul Hasan (Opu)
Sent: Sunday, January 11, 2009 3:36 PM

To: pjsip list
Subject: Re: [pjsip] APS Implementation

Hi Pai,

Can you please tell me if i want to g729 from Aps what should i need to do?

in the line of encoder mode

iSession.SetEncoderMode(EALawOr20ms);

is for g711 but i saw no encoding mode for g729 . should i block those line
if i want to use g729?

Thanks

On Sat, Jan 10, 2009 at 12:19 AM, Pai Peng sipaipv6@gmail.com wrote:

Hi,

in source symbian_sound_aps.cpp:

void CPjAudioEngine::RecCb(TAPSCommBuffer &buffer)
{
...
buf[i] = pjmedia_alaw2linear(iBuffer[i]);
...
}

TInt CPjAudioEngine::StartStreamL()
{
....
iSession.SetEncoderMode(EALawOr20ms);
iSession.SetDecoderMode(EALawOr20ms);
...
}

audio is in G711-A Format.

The  APS supports the following codecs: AMR, G711, G729 and iLBC.

If you like use other codec for encoding or decoding, you should change the
SetEncoderMode() and SetDecoderMode().

Pai

On Fri, Jan 9, 2009 at 5:54 PM, Rawshan Iajdani iajdani@provati.com wrote:

How does APS implementation works in Symbian_ua?

Does g711,gsm,speex all works fine?

APS->G711->PCM->gsm

Or

APS->gsm??

Does it do the transcoding or it just only works for g711 and not gsm and
speex?

thanks

Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

--
S. M. Nazmul Hasan Opu
Software Engineer
R & D Application
Dhaka, Bangladesh
Mob: +880 1712 901 764

Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

--
S. M. Nazmul Hasan Opu
Software Engineer
R & D Application
Dhaka, Bangladesh
Mob: +880 1712 901 764

Great. I am currently in Dhaka. Which operator are you using to connect to
your server? Grameen or Banglalink?

Seems like both of them not working for me. They are using some sort of 10
series IP and I detect NAT? Did you face this?

From: pjsip-bounces@lists.pjsip.org [mailto:pjsip-bounces@lists.pjsip.org]
On Behalf Of S. M. Nazmul Hasan (Opu)
Sent: Tuesday, January 13, 2009 3:21 PM
To: pjsip list
Subject: Re: [pjsip] APS Implementation

yaap..

On Tue, Jan 13, 2009 at 3:47 AM, Rawshan Iajdani iajdani@provati.com
wrote:

Opu,

You live Dhaka??

From: pjsip-bounces@lists.pjsip.org [mailto:pjsip-bounces@lists.pjsip.org]
On Behalf Of S. M. Nazmul Hasan (Opu)
Sent: Sunday, January 11, 2009 7:01 PM

To: pjsip list
Subject: Re: [pjsip] APS Implementation

I dont think APS implementation(G729) is so hard as you think. APS already
has G729 codec support that is if i use APS i must get the implementation of
g729 innerly . i just have to change the procedure to use g729 codec....like
some parameter and encoding decoding technique. ...

APS use 24byte data ...thats a point also...

On Sun, Jan 11, 2009 at 3:46 PM, Rawshan Iajdani iajdani@provati.com
wrote:

I don't think you can with this current wrapper.

So far I understood, the current wrapper just take PCM and convert that to
g.711 and pass it to APS and vice versa. You still need your own g.729
implementation. This wrapper just take PCM as input and output..

Correct me if I were wrong..

From: pjsip-bounces@lists.pjsip.org [mailto:pjsip-bounces@lists.pjsip.org]
On Behalf Of S. M. Nazmul Hasan (Opu)
Sent: Sunday, January 11, 2009 3:36 PM

To: pjsip list
Subject: Re: [pjsip] APS Implementation

Hi Pai,

Can you please tell me if i want to g729 from Aps what should i need to do?

in the line of encoder mode

iSession.SetEncoderMode(EALawOr20ms);

is for g711 but i saw no encoding mode for g729 . should i block those line
if i want to use g729?

Thanks

On Sat, Jan 10, 2009 at 12:19 AM, Pai Peng sipaipv6@gmail.com wrote:

Hi,

in source symbian_sound_aps.cpp:

void CPjAudioEngine::RecCb(TAPSCommBuffer &buffer)
{
...
buf[i] = pjmedia_alaw2linear(iBuffer[i]);
...
}

TInt CPjAudioEngine::StartStreamL()
{
....
iSession.SetEncoderMode(EALawOr20ms);
iSession.SetDecoderMode(EALawOr20ms);
...
}

audio is in G711-A Format.

The  APS supports the following codecs: AMR, G711, G729 and iLBC.

If you like use other codec for encoding or decoding, you should change the
SetEncoderMode() and SetDecoderMode().

Pai

On Fri, Jan 9, 2009 at 5:54 PM, Rawshan Iajdani iajdani@provati.com wrote:

How does APS implementation works in Symbian_ua?

Does g711,gsm,speex all works fine?

APS->G711->PCM->gsm

Or

APS->gsm??

Does it do the transcoding or it just only works for g711 and not gsm and
speex?

thanks

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It is finaly working. Thanks alot Norman for your help.

To anyone who is interested in sending PCM audio frames colected outside of pjsip:

// Create a pjmedia_port
pjmedia_port port;

// init it's info
pjmedia_port_info_init(&port.info, &name, signature, samplingRate, channelCount, bitsPerSample, samplesPerFrame);

// Set it's callback functions
port.get_frame = &get_frame;
port.put_frame = &put_frame;
port.on_destroy = &on_destroy;

// Add port to conference bridge
pjmedia_conf_add_port(conference_bridge, pool_vars, &port, NULL, &slot);

// Connect the port to the call's conference slot
pjmedia_conf_connect_port(SIPEngine::instance->conference_bridge, slot, call_port, 0);

// Implement the get_frame...the method called by pjsip to get new audio frames to send
pj_status_t get_frame(struct pjmedia_port* this_port, pjmedia_frame* frame) {

 .... code to get an audio frame to be sent

 return PJ_SUCCESS;

}

// The put_frame method is a simple one...because we only send frames
pj_status_t put_frame(struct pjmedia_port* this_port, const pjmedia_frame* frame) {
return PJ_EINVALIDOP;
}
// The on_destroy has to have the following lines of code
pj_status_t on_destroy(struct pjmedia_port* this_port) {

 ....

 pjmedia_conf_remove_port(conference_bridge, slot);

 ...

 pjmedia_port_destroy(port);

 ...

 return PJ_SUCCESS;

}

I hope this helps anyone who add the same problems has i did...

Rafael Maia

From: norman@myasd.comTo: pjsip@lists.pjsip.orgDate: Mon, 12 Jan 2009 16:09:50 -0500Subject: Re: [pjsip] Sending my own audio framesYou need to connect it to the port associated with the active call as well. PJ is very flexible with audio routing, so you need to explicitly connect up everything that should "hear" your source. Your player should connect to the user's speakers and the call or calls in progress.

Norman Franke
Answering Service for Directors, Inc.
www.myasd.com

On Jan 12, 2009, at 2:15 PM, Rafael Maia wrote:

Thanks alot for your source code Norman, it has been a big boost to my code development...  I am having a diferent problem now.I can ear the sound frames i added on my sound speakers, but i am not sending them in the RTP session.I did the following: pjmedia_conf_add_port(conference_bridge, pool_vars, &port, NULL, &slot);pjmedia_conf_connect_port(conference_bridge, slot, inSlot, 0);  I have tested putting frames on the master_port directly, and the saound frames are sent throw RTP: pjmedia_port* master = pjmedia_conf_get_master_port(conference_bridge);pjmedia_port_put_frame(master, &frame);  ...but i don't think this is the proper way to do it Can anyone help?How should i send the audio frames? Thanks in advance for any help....  Rafael Maia

From: norman@myasd.comTo: pjsip@lists.pjsip.orgDate: Fri, 9 Jan 2009 18:48:10 -0500Subject: Re: [pjsip] Sending my own audio framesYou can use pjmedia_splitcomb_create_rev_channel to reverse the put/get semantics. I do that elsewhere and it works.

I did then add it to the conference bridge.

Sadly, my code isn't easy to extract and is very C++ using boost and a custom string library.

typedef bool (*GetSamplesPtr)(void * inBuf, uint32_t * ioByteSize, pj_timestamp * inTime);

PlayPort::PlayPort(
unsigned sampling_rate,
unsigned channel_count,
unsigned samples_per_frame,
unsigned bits_per_sample
)
: pool(NULL), mCallIndex(INVALID_CALL_INDEX)
{
const pj_str_t name = pj_str("file");

pool = pjsua_pool_create("asd_player", 4000, 4000);
assert(pool);

pj_strdup2(pool, &base.info.name, "asd_player");

base.get_frame = &file_get_frame_get;
base.put_frame = &file_put_frame_get;
base.on_destroy = &file_on_destroy_get;

pjmedia_port_info_init(&base.info, &name, SIGNATURE, sampling_rate, channel_count, bits_per_sample, samples_per_frame);

port_is_open = true;
options = 0;

}

PlayPort::~PlayPort()
{
Close();

if (pool) {
    pj_pool_release(pool);
    pool = NULL;
}

}

pj_status_t PlayPort::Open(SInt32 inCallIndex, float inVolume, GetSamplesPtr inFunc)
{
pj_status_t status;

status = pjsua_conf_add_port(pool, &base, &slot);
if (status != PJ_SUCCESS) {
    pjmedia_port_destroy(&base);
    return status;
}

mCallIndex = inCallIndex;

status = pjsua_conf_adjust_rx_level(slot, inVolume);

port_func = inFunc;

return status;

}

void PlayPort::Close(void)
{
file_on_destroy_get(&base);
}

bool PlayPort::IsOpen(void)
{
return port_is_open && slot >= 0;
}

pj_status_t PlayPort::ConnectTo(int inSlot)
{
return pjsua_conf_connect(slot, inSlot);
}

pj_status_t PlayPort::DisconnectFrom(int inSlot)
{
return pjsua_conf_disconnect(slot, inSlot);
}

// Class static
pj_status_t PlayPort::file_on_destroy_get(pjmedia_port *this_port)
{
// As the first element in a non-virtual class...
PlayPort *fport = (PlayPort *) this_port;
boost::recursive_mutex::scoped_lock l(fport->m);

pj_assert(this_port->info.signature == SIGNATURE);

if (fport->port_is_open) {
 fport->port_is_open = false; // Prevents a recursive callback

    fport->port_func = NULL;
    pjsua_conf_remove_port(fport->slot); // If it's already removed, e.g. audio device changed, this crashes. 
    fport->slot = 0;
    pjmedia_port_destroy(&fport->base);
}

return PJ_SUCCESS;

}

// Class static
pj_status_t PlayPort::file_get_frame_get(pjmedia_port *this_port, pjmedia_frame *frame)
{
// As the first element in a non-virtual class...
PlayPort *fport = (PlayPort *) this_port;

pj_assert(this_port->info.signature == SIGNATURE);

if (frame->size == 0) return PJ_SUCCESS;
if (! fport->port_func) return PJ_EEOF;

pj_size_t frame_size = frame->size;
if ((*fport->port_func)(frame->buf, &frame_size, (PJSUA_Timestamp *) &frame->timestamp)) {
    if (frame_size < frame->size) {
        bzero((UInt8*)frame->buf + frame_size, frame->size - frame_size);
    }
    frame->timestamp.u64 = 0;
    frame->type = PJMEDIA_FRAME_TYPE_AUDIO;
    return PJ_SUCCESS;
}
file_on_destroy_get(this_port);
return PJ_EEOF;

}

// Class static
pj_status_t PlayPort::file_put_frame_get(pjmedia_port *this_port, const pjmedia_frame *frame)
{
return PJ_EINVALIDOP;
}

Norman Franke
Answering Service for Directors, Inc.
www.myasd.com

On Jan 9, 2009, at 1:27 PM, Rafael Maia wrote:

Thanks for your anwser Norman. You are using a very diferent approach from mine. Could you provide some more information ( or some source code ) ? My DirectShow sound reading routines are pushing audio frames that i need to send throw PJSIP. So i need to call put_frame on a media port.How did you created this "Passive" port ? You also add it to the conference bridge, right ? Thanks in advance... Rafael Maia

From: norman@myasd.comTo: pjsip@lists.pjsip.orgDate: Fri, 9 Jan 2009 12:17:19 -0500Subject: Re: [pjsip] Sending my own audio framesI use pjmedia_port_info_init to create a player port. Something like this:

pool = pjsua_pool_create("MyPlayerPool", 4000, 4000);
pj_strdup2(pool, &base.info.name, "MyPlayer");

base.get_frame = &file_get_frame_get;
base.put_frame = &file_put_frame_get;
base.on_destroy = &file_on_destroy_get;

pjmedia_port_info_init(&base.info, &name, SIGNATURE, sampling_rate, channel_count, bits_per_sample, samples_per_frame);

This works well for playing files in a non-standard format that I need to support. You put your sample generation code in file_get_frame_get.

Norman Franke
Answering Service for Directors, Inc.
www.myasd.com

On Jan 9, 2009, at 9:24 AM, Rafael Maia wrote:

Hi, i made a media player in DirectShow that captures PCM audio frames from mp3 audio files. I need to send this audio frames to PJSIP. Can anyone help me? I have tried adding a passive_port to the conference bridge and it did not worked. Now i am trying to use splitcomb. And i can send some noise. Here's a code sample: // 1º pjmedia_splitcom_createpj_status_t status = pjmedia_splitcomb_create(pool_vars, nSamplesPerSec, nChannels, samplesPerFrame, wBitsPerSample, 0, &p_splitcomb);// for each channelfor(unsigned int i = 0; i < nChannels; i++)  {    unsigned int p_slot;    pjmedia_port* p_port;      // 2º pjmedia_splitcom_create_rev_channel    status = pjmedia_splitcomb_create_rev_channel(pool_vars, p_splitcomb, i, 0, &p_port);      // 3º pjmedia_conf_add_port    status = pjmedia_conf_add_port(conference_bridge, pool_vars, p_port, NULL, &p_slot);}  And then, when i want to send an audio frame i do the following: pjmedia_port_put_frame(p_splitcomb, &frame);  What is wrong with my code?Am i using splitcom correctly? Any help would be appreciated... Rafael Maia

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