You could run one WASAPI thread and service both playback and capture events using WaitForMultipleObjects. You should implement audio capture for the obvious reason to make the microphone work, plus not providing pjsip with capture buffers will cause lots of underflow messages. Bill On 3/26/2014 12:48 PM, Nuno Centeio wrote: > *@Bill,* > > But the record is not made in the thread that gets the frame (my > WASAPI thread), right? Or do I need to implement my capture engine in > order for the render buffer to be correct? > > *@Abdelrahman*, > > I think that the only problem will be the WinSocks. However, I'm not > targeting for ARM devices, where WinSocks does not exist. In x86 and > x64 processors there is WinSocks. But, however, I'm already prepared > to be rejected a lot of times :( > > > On 26 March 2014 16:40, Abdelrahman Salah > <abdelrahman.salah.cs at gmail.com > <mailto:abdelrahman.salah.cs at gmail.com>> wrote: > > Dear nuno, > > This is a side hint, please bay attention to the dependencies > required, because most of the dependencies required are not > allowed in windows store and more than likely it will fail the > certificate test. > So take that into consideration before spending time and effort. > > Abdelrahman Salah > Sent From My Note II > > On 26 Mar 2014 18:25, "Nuno Centeio" <nuno.r.centeio at gmail.com > <mailto:nuno.r.centeio at gmail.com>> wrote: > > Hi guys, > > I'm trying to implement the Windows Audio Session API (WASAPI) > in order to make PJSip compliant with Windows Store apps. > > I think that I'm almost done but I can get the sound correctly > (it's just that!). > > Here is what I have: > > (pb_client is a IAudioRenderClient) > hr = strm->pb_client->GetBuffer(frame_to_render, &cur_pb_buf); > > pjmedia_frame frame; > void* destBuffer = > (void*)malloc(strm->bytes_per_frame*frame_to_render*sizeof(pj_uint16_t)); > /* PCM mode */ > frame.type = PJMEDIA_FRAME_TYPE_AUDIO; > frame.size = strm->bytes_per_frame; > frame.timestamp.u64 = strm->pb_timestamp.u64; > frame.bit_info = 0; > frame.buf = destBuffer; > > status = (*strm->pb_cb)(strm->user_data, &frame); > > /* Because PJSip will give me PCM 16 bits, 2 channels and > WASAPI needs > IEEE Floating point 32-bits, 2 channels > */ > CAudioConverter* conv = new CAudioConverter(); > conv->ConvI16ToF32((BYTE*)destBuffer, cur_pb_buf, > frame_to_render); > > /* Write to the device. */ > hr = strm->pb_client->ReleaseBuffer(frame_to_render, 0); > > While running I have the following prints: > 3/26/2014 4:00:01 PM V SIP - Underflow, buf_cnt=0, > will generate 1 frame > 3/26/2014 4:00:01 PM V SIP - Underflow, buf_cnt=0, > will generate 1 frame > 3/26/2014 4:00:01 PM V SIP - 698 samples reduced, > buf_cnt=6092 > 3/26/2014 4:00:01 PM V SIP - Buffer size adjusted > from 6790 to 6092 (eff_cnt=5760) > 3/26/2014 4:00:01 PM V SIP - Underflow, buf_cnt=0, > will generate 1 frame > 3/26/2014 4:00:02 PM V SIP - 320 samples reduced, > buf_cnt=6412 > 3/26/2014 4:00:02 PM V SIP - Buffer size adjusted > from 6732 to 6412 (eff_cnt=5760) > > Any hints on what I might be doing wrong? > > Thanks > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org <mailto:pjsip at lists.pjsip.org> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org <mailto:pjsip at lists.pjsip.org> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20140326/1d8026e7/attachment-0001.html>
