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 at lists.pjsip.orgDate: Fri, 9 Jan 2009 18:48:10 -0500Subject: Re: 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 at lists.pjsip.orgDate: Fri, 9 Jan 2009 12:17:19 -0500Subject: Re: 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: // 1? 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; // 2? pjmedia_splitcom_create_rev_channel status = pjmedia_splitcomb_create_rev_channel(pool_vars, p_splitcomb, i, 0, &p_port); // 3? 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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