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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20140326/22fb9f69/attachment-0001.html>
