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Trying to write a program to record phone calls and act as answering machine. I'm initializing TAPI, opening the cellular line, then trying to use lineGetID to get the "wave/in" and "wave/out" devices id's, this last function call always returns the same error "the operation failed".
Any thoughts???
as I've wrote before it is impossible to make an answering machine. You can record phone calls, but cannot play anything to the line. It is a software limitation in CE 4.20
HOW can you record phone calls
(sounds handy when certain callcentre agents make promises)
built-in recorder can record calls. You may decompile it to find how it is working. I think that code used to record calls is the same as recording from a device microphone when the call is not active.
How would you use LineGetId in order to recieve data calls through TAPI ?? I'm struggling udnerstanding where this function should be implemented and what it actually does.
Thanks for all your help !!!
After you catch CONNECT event (media type is datamodem) you must get handler of opened serial device for data transmition
1)LPVARSTRING lpVarStr;
2)Allocate memory for lpVarStr;
3)Call lineGetID
for example: lineGetID(0, 0, hCall, LINECALLSELECT_CALL, lpVarStr, TEXT("comm/datamodem"))
4-th parameter means how call will be selected
LINECALLSELECT_CALL - select by specified call
also you can select by deviceID, address, line etc
last parameter specifies device class (look msdn for possible values)
4)check lpVarStr->dwNeededSize
5)Realloc lpVarStr
6)call lineGetID
7)get handle of serial device *(LPHANDLE)((BYTE*)lpVarStr + lpVarStr->dwStringOffset))
I think you will find that the ability to record the incoming audio is only as a result of the microphone picking up the audio from the speaker. I have tested everything I can think of to do this and hit a brick wall. firstly if you check the linedevcaps for each line device, non support the "wave/***" classes needed. Secondly the linegetid will always fail because of the way the line is being opened. To open a line correctly to use "wave/***" in linegetid, the dwmediamodes flag of lineopen needs to be set to LINEMEDIAMODE_AUTOMATEDVOICE. If you did not an error of LINEERR_INVALMEDIAMODE returns. If your error is LINEERR_OPERATIONFAILED, I think it could be coming from your timing of when you call linegetid. This will not work on any device I have because the hardware will not support it.
There is a physical seperation between the hardware that picks up and sends voice, and the computing hardware. I first encountered this while using an audiovox rtm8000 card in a toshiba e570. The modem built into the card can communicate data over the port but dose nothing with audio. That may sound irrelivant exept that later audiovox started selling the hardware combination built together under their brand as a pda phone to compeat with xda. They modified it by hooking the system speaker and mic up to the card with extra wires internally. All pda phones are arranged essentially like this. the fact that you cant see a seperate "phone card" makes it deceptive.
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loose end. How dose my bluetooth wireless earpeice get the audio from my xdaII. Explore that, I am.
Having installed SJPhone 3.20b for Windows Mobile 5.0 (download from: http://www.sjphone.org/preview/ce/) I experienced very choppy inbound audio.
The solution is changing SJPhone's advanced audio settings as follows:
buffer size: 80 ms
input queue length: 10
output queue length: 5
RTP jitter queue length: 5
Audio quality through WLAN is quite okay now. As soon I'm in a place with 3G coverage I'll try to place a VoIP call over 3G, if O2 Germany hasn't blocked VoIP ports yet.
I am getting better resutls with these settings:
Driver buffer size, msec: 100
Driver input queue length: 10
Driver output queue length: 10
RTP jitter queue length: 32
Btw, x-lite ce 1.01 has better quality than SJphone, at least on my TyTN, download here:
http://forum.xda-developers.com/viewtopic.php?t=58154
Also, calls must be done using the USB headset otherwise you get horrybly echo.
I do have the sound coming from the rear speaker...is that what you have?
Further, the sound with SJphone is absolutely horrible...whatever the audio settings....
I tried X-lite..gives same effects...
It s really unusable.
When I call the same correspondant with my PC, it s seemless..
Any clues?
Thx
use the usb headset
No sound...
Hi. I am using HTC X01HT (Softbank) which is supposed to be equivalent to TyTN. I installed SJphone but for some reason no sound is coming. Seemingly, even though I don't hear any sound, the recipient phone is normally ringing. Of course, even if the recipient picks up the phone, still silence on both sides. Any solution? Thank you in advance.
Check your SIP and NAT settings. You'll have problems if your internet connection or the voip server you are connecting to are behind a NAT.
You should use a STUN-server!
@Poof
Hi Poof,
Sorry for the trivial question but I geuss that answer isn't that obvious as it seems. YOu've mentioned that you're using X-Lite with some specific settings. I've downloaded the software using the link you had provided. I did manage to specify the SIP proxy and other SIP related settings during the first run of X-Light.
My question is: HOW DO YOU CHANGE OTHER SETTINGS? In case I need to change SIP IP HOW DO I DO THAT? I didn't succeed with finding searate file responsible for configuration nor I cold find settings changes option in the application itself.
I'd appreciate if you could advice me on these mattares.
TyTN and SJPhone
Hi,
I get horrible sound delays with SJPhone + TyTN. I tried all the settings from this forum and many others and although the sound quality seems OK (I didn't see any significant improvement because of the USB headsed) the delay is 2 to 3 times bigger (up to 1.5-2 s) than on my Wizard.
Please let me know if the above mentioned settings don't give you any delay. If there's no delay, could you please let me know the ROM (and any possible improvements) that you're using?
Please follow this thread also for news:
http://forum.labs.softjoys.com/viewtopic.php?t=409&postdays=0&postorder=asc&start=20
As for the X-Lite application - I get also delays and bad sound quality. As for its interface - it makes you go nuts, and even more. I never managed to find out how to change the settings for one SIP-account. (through the registry ? this is a joke, right?) Even the PC version is made as if that guy wanted to make a patience testing quiz, not a softphone.
Thank you all.
USB headset
It seems it has something to do wih the USB headset. If I use it - SJPhone gets stuck. If I don't use it - SJPhone runs fine and even the sound quality is acceptable
realn said:
Hi,
I get horrible sound delays with SJPhone + TyTN. I tried all the settings from this forum and many others and although the sound quality seems OK (I didn't see any significant improvement because of the USB headsed) the delay is 2 to 3 times bigger (up to 1.5-2 s) than on my Wizard.
Please let me know if the above mentioned settings don't give you any delay. If there's no delay, could you please let me know the ROM (and any possible improvements) that you're using?
Please follow this thread also for news:
http://forum.labs.softjoys.com/viewtopic.php?t=409&postdays=0&postorder=asc&start=20
As for the X-Lite application - I get also delays and bad sound quality. As for its interface - it makes you go nuts, and even more. I never managed to find out how to change the settings for one SIP-account. (through the registry ? this is a joke, right?) Even the PC version is made as if that guy wanted to make a patience testing quiz, not a softphone.
Thank you all.
Click to expand...
Click to collapse
You can change the settings in the X-Lite app by dialing *311 and *611, these take you to the settings pages. I have however not been able to connect to the network in any way, perhaps you can shed light on the right settings, pof? I am using a voipstunt account, but I have no idea what settings I should use to get a connection to the internet. I have entered the voipstunt settings I found on their website and my account data, but for some reason the app wil not connect.
edit: For some reason when dialing a fixed line phone numer, the Caller is is displayed on the screen as the whole phone numer followed by @voipstunt.com, e.g [email protected]. It looks like the app is trying to reach a voipstunt subsciber instead of a fixed line. How did you get this to work pof? The app keeps trying for a while and then reports Call failed: 480, temporarily not available.
edit 2: Duh, I really have to learn to try more before I post here. I think that by editting the right options and using it over 3G instead of AS I have been able to connect. I cannot try any calls however, none of my contacts with fixed lines appears to be home at the moment (Or they are refusing my calls, also a likely possibility)
Really annoying though is that the app does not sound the waiting tone when it is calling a no., so you have no idea wether or not it is dialing or the connection is just not being made.
I've just installed the latest sjphone beta for ce, to use with fwd. I can't find where to change the advanced audio options. In fact the only audio control I can find is the audio wizard. Can anyoine help me.
In addition my usb headphones long since stopped working, would sound and mic be better with a bluetooth headset (in my case the itech r35) or through the phone itself?
Currently audio is unuseable using the phone, and apparently not working at all with the bluetooth headset.
Your assistance appreciated.
Hi,
i can't use the BTAudioToogle Tool for my XDA trion.
Someone the same problem ?
or better:
Someone with a solution here ?
Same for me. I have enabled BTaudiotoggle, I can hear the headset (Motorola HS850) go live but the audio either from WM player or TCMP still comes from the speaker and not the headset.
Hope someone comes up with the answer soon.
Rob
Got the same problem. No sound from BT.
Had something similiar to this with my old unit, to fully enable it I would hit the headset button (which now starts up voice command, let it time out, and then play whatever music i wanted to. I would verify that now but i am at work, and don't think they would appreciate the music blaring if i am wrong on this one
When I hit the button on the headset, it starts the Voice Dialer, the audio cuts in for a second or two, then drops back to the device speaker.
It nearly works, hope someone can hack this soon, just whats needed for podcasts and the like. No good for music tho!
Rob
I would also be very interested in a solution to this problem as I used to use BTaudiotoggle to listen to Tomtom commands in the car (normally have the music too loud to hear the commands through the loudspeaker ).
Worked great with my XDA Mini S but doesn't work with my new MDA Vario II and I didn't really want to have to buy stereo headphones, just wanted to be able to use my standard mono bluetooth headset like before.
If there is no way of using this program, are there any other programs that will send all audio to a mono BT Headset?
yep would really like an updated version of BTaudio too so i can listen to some audio books via my mono headset
You should test the trial version of bluemusic from teksoftco.
www.teksoftco.com . It gives ou a full help file on how to use the software and you have 30 sec to test that it working. I know that it works on the devices having TI Omap processors so it should work for all of you guys.
Cheers,
Raul
Just tried it out. It does the same thing that the other sotware does, mainly enable the headset where you can hear a hiss out of it, but still kees playing out of the back speaker.
My experience is a rassah's. All the Bluemusic software appears to do on my Vario II is cause the bluetooth headset to hiss slightly (whilst sound is still blasted out of the phone's speaker).
Also FWIW the teksoftco site doesn't appear to work too well on the PIE browser - defaulting to a no frames version with just a login box as far as I can see.
Needless to say I have uninstalled the software again.
Rufus.
@RufusA and @rassah guys you should check if you don't have the same problem as ulischultz in this thread :
http://forum.xda-developers.com/viewtopic.php?t=61739
We already tested the software with him on his treo device and as you can see in the thread " after pressing the button on the headset, the sound was redirected". This is mentioned in the help section of the software. ( on the universal/exec for example this step is not needed)
@RufusA - thanks for visiting the ppc version of the webpage (which is still under construction) - even though the subject is irelevant in this topic. I will send you a PM when its finished and tell me your opinion then (even though you have to admit that not many companies have a ppc version of their website - not to mention the smaller ones like ours).
If the problem still persists there is an opened thread on the forum :
http://www.teksoftco.com/forum/viewtopic.php?t=162
Please submit all the malfunctionality and we will fix them.
Cheers,
rain
I downloaded the trial too and had the same problem as before unfortunately.
I did get a little further in so much as, if I pressed the call button on my BT Headset as per the instructions, I could get the music/audio transferred to the headset for approx. 5 seconds but then it would revert back to the phones loudspeaker. I couldn't get the 30 seconds that the trial software advertises
Fingers crossed someone will be able to work out a fix for this soon.
Thanks Nicky for the input. As i see the problem of bt headset being disconnected after 5-10 sec is a general one (and unfortunatelly is a problem in the RUU). We are trying to find a workarround with ulischultz but wouldn't want to get to driver level as clearly it will be fixed by HTC in the next RUU release if they don't want this devices smashed to their heads.
I will keep you updated if we find a solution for the interrupt issue.
Regards,
Raul
Solution
I have identified a solution to this which is working nearly perfectly for me. However, if you do use this then using the BT headset to trigger the voice dial/voice command/voice commander will no longer work.
Acquire a copy of of the BTAudioToggle software inc. BTAudioOn.exe
Copy this to a location on the phone.
Using a registry editor amend the following entries from
HKEY_LOCAL_MACHINE\SOFTWARE\OEM\VoiceCommand
Path = location of BTAudioOn.Exe
e.g. Path = \Program Files\BTAudio\BTAudioOn.Exe
VoiceCmdDuration = 86400 (DEC) = 24 hours
This says for how long (in seconds) the audio will be passed to the headset for before timing out.
This however, will timeout earlier if no audio is coming through. I am not sure what will happen if something like TomTom is running but not speaking for a while. I have not performed timings as yet.
To activate audio through headset, press the main control button of the headset as if answering call/activating voice command etc. To deactivate, just press the button again.
Voice control apps will still work, but must be triggered from the phone itself, but voice will be picked up by the BT headset.
There appear to be some small bugs with this, such as after a phone call, using headset or not, the audio will stop being passed after hangup.
Please let me know if this work for you!
Looks an interesting solution. I've installed BTAudioOn on my storage card, and changed the VoiceCommand Path entry.
However I couldn't find a VoiceCmdDuraion entry under
HKEY_LOCAL_MACHINE\SOFTWARE\OEM\VoiceCommand
so I've added a new DWord of that name with the value 86400.
Yet this doesn't appear to have made a significant difference i.e. Pressing talk button on Jabra headset produces 10 secs of sound before "timing" out.
Have I made a mistake, or do I need to have something else installed to get this to work?
TIA - Rufus.
I was going to try and make the suggested changes to the registry but I don't know how and am not sure what program I need to do it.
Could someone please let me know what program I need to install to be able to change the registry?
Thanks
Vs1979:
IT REALLY WORKS!!!!!!!!!
I can't tell you how happy I am!!!!
I did it the way you described it. However I also did not have an entry "VoiceCmdDuration" I created the DWORD-entry and it works now!
Great!
Thanks very much for that great solution!
@RufusA
As i gave you the link to the other thread you're not the only user experiencing this. There are several others complaining about this RUU bug.
@Nicky
You need to install a ppc 2005 regeditor (for example Resco Explorer has one built in, or you can find one on buzzdev site).
RufusA said:
Looks an interesting solution. I've installed BTAudioOn on my storage card, and changed the VoiceCommand Path entry.
However I couldn't find a VoiceCmdDuraion entry under
HKEY_LOCAL_MACHINE\SOFTWARE\OEM\VoiceCommand
so I've added a new DWord of that name with the value 86400.
Yet this doesn't appear to have made a significant difference i.e. Pressing talk button on Jabra headset produces 10 secs of sound before "timing" out.
Have I made a mistake, or do I need to have something else installed to get this to work?
TIA - Rufus.
Click to expand...
Click to collapse
Hi,
Did you Soft reset your device?
I made the registry change only, have not installed BTAudioOn and all my Voice command Audio etc... now comes through the headset and does not time out after a few seconds. I did find though that decreasing this value to lets say 60, would actually time the audio feed out even if not in use so I have stuck in a huge figure, will still need to do more testing to see if drains the battery any quicker and see if I can find another stirng which will do more for me.
Hi,
it work not for me
i have changed the path to \xx\BTAudioOn.EXE
i have add the VoiceDuration = 86400 (DWord DEC)
but it is the same problem, after 2 seconds the sound goes back to PPC.
This subject seems to have been beaten to death in the other device forums, but I haven't seen any direct references to this in the TyTN forums, so here goes: has anyone who experienced the ( every 10-20s) skipping problem using the A2DP profile found a solution?
I've seen a couple of registry hacks in other forums which seem to have worked for others, but nothing so far about the TyTN.
I have everything working now with WMP except this and the issue with the ring tone being delayed and coming only from the headset; see my other post this morning.
So far no other problems with the TyTN.
/POL
Skipping happens on my TyTN mostly when the WiFi radio is on. When turned off, the music skips only every now and then.
Stefan Mensink said:
Skipping happens on my TyTN mostly when the WiFi radio is on. When turned off, the music skips only every now and then.
Click to expand...
Click to collapse
Not in my case. Wifi is turned off, and I still get skipping every 10-20 seconds.
/POL
Improvement of Bluetooth Listen Music
Find this registry : :idea:
HKLM/SOFTWARE/Microsoft/Bluetooth/A2DP/Settings/UseJointStereo (Dword:1)
Change the Value to 0
Also you can add a new registry to improve the quality:
HKLM\Software\Microsoft\Bluetooth\A2DP\Settings
Add : BitPool [Dword : 48] Maximum is 58
sas90850 said:
Improvement of Bluetooth Listen Music
Find this registry : :idea:
HKLM/SOFTWARE/Microsoft/Bluetooth/A2DP/Settings/UseJointStereo (Dword:1)
Change the Value to 0
Also you can add a new registry to improve the quality:
HKLM\Software\Microsoft\Bluetooth\A2DP\Settings
Add : BitPool [Dword : 48] Maximum is 58
Click to expand...
Click to collapse
Yes, I've done these mods which set to stereo (why on earth have mono as default?) and select audio quality, but these don't affect the skipping.
Unless lowering the BitPool value will cure the skipping - but I want to keep it at 48 at least.
/POL
Check through my notes thread here. There's been some research going on with priority settings and specific bitpool min/max values. However, there appears to be more than just reg settings at work though. The entire device slows to a crawl too. My gut feeling is that this won't be addressed until HTC releases their first official update, rumored to be in a few weeks or so.
i have done that but still skips every 30 secs or so.
Interestingly, I could hear the same "tick" on my 8125 as well but it sounded like a slight scratch on a record. On the TyTN its magnified to this major dropout. I'm really thinking our only A2DP solution will be to wait for the updated ROM I'm afraid.
Let's hope they don't take the coward's way out and remove A2DP altogether as was done in several Operator's initial AKU2 release for the Wizard.
I think you guys should look into optimising your SD cards and trying the reg entries sleuth255 indicated. My A2DP seems to work fine with probably 99% skip free playing. I'm currently one floor above my Nokia HS-12W and am still connected to my TyTN ok in certain parts of the room. If i am right beside it or in the same room I dont get any skips at all; apart from when I lose or reaquire a GSM signal.
well, that's encouraging news efjay! Much better results than I ever got. Btw: are you running the newer DoPod rom onto your TyTN by any chance?
edit: nevermind; I just used the Wiki to crossref. If your signature is correct then you've got the standard TyTN ROM build running.
Yes im running the TyTN rom. I actually had the Dopod rom but apart from a few things in the extended rom wasnt too different so i flashed back to the TyTN one. A2DP works ok now, was listening most of the day and the TyTN battery even outlasted the Nokia battery.
Another good point in the Nokia's favour is you can re-establish a stereo connection without having to go to the BT settings - just push the Audio button and it reconnects for you and the music starts.
efjay said:
I think you guys should look into optimising your SD cards and trying the reg entries sleuth255 indicated. My A2DP seems to work fine with probably 99% skip free playing. I'm currently one floor above my Nokia HS-12W and am still connected to my TyTN ok in certain parts of the room. If i am right beside it or in the same room I dont get any skips at all; apart from when I lose or reaquire a GSM signal.
Click to expand...
Click to collapse
Some good news on my part! I've done all the mods suggested by sleuth255 - except lowering the BitPool value to 5 - I suspect that would decrease the audio quality too much - and I now have skipping only about once a minute. Much better.
How do I go about optimizing the (micro)SD card?
/POL
politby said:
efjay said:
I think you guys should look into optimising your SD cards and trying the reg entries sleuth255 indicated. My A2DP seems to work fine with probably 99% skip free playing. I'm currently one floor above my Nokia HS-12W and am still connected to my TyTN ok in certain parts of the room. If i am right beside it or in the same room I dont get any skips at all; apart from when I lose or reaquire a GSM signal.
Click to expand...
Click to collapse
Some good news on my part! I've done all the mods suggested by sleuth255 - except lowering the BitPool value to 5 - I suspect that would decrease the audio quality too much - and I now have skipping only about once a minute. Much better.
How do I go about optimizing the (micro)SD card?
/POL
Click to expand...
Click to collapse
Forgot to say that this time I actually listened to music for about 10 minutes and the device does indeed slow to a crawl. Pretty much non-responsive, actually.
/POL
politby said:
efjay said:
I think you guys should look into optimising your SD cards and trying the reg entries sleuth255 indicated. My A2DP seems to work fine with probably 99% skip free playing. I'm currently one floor above my Nokia HS-12W and am still connected to my TyTN ok in certain parts of the room. If i am right beside it or in the same room I dont get any skips at all; apart from when I lose or reaquire a GSM signal.
Click to expand...
Click to collapse
Some good news on my part! I've done all the mods suggested by sleuth255 - except lowering the BitPool value to 5 - I suspect that would decrease the audio quality too much - and I now have skipping only about once a minute. Much better.
How do I go about optimizing the (micro)SD card?
/POL
Click to expand...
Click to collapse
Check out this guide from Menneisyys
http://www.pocketpcmag.com/forum/topic.asp?TOPIC_ID=17921
Or the short version - get SK Tools (trial version should be ok, if not it costs $13) and format your card but make sure the backup FAT table is disabled.
I used it with the default cluster size values and also have other registry tweaks applied and A2DP works; I am even able to browse via GPRS when streaming music.
A2DP and WLAN - no such luck!
It became obvious to me today that using a stereo Bluetooth headset while connected via WiFi is a non starter. Continuously skipping audio and the device hangs after a few minutes and needs a restart.
There goes my dream of listening to Internet radio over a WiFi connecIttion. It works over a UMTS connection but since my carrier doesn't have an unlimited data plan, that's an expensive proposition.
Didn't HTC test such an obvious scenario?
Bummer.
@efjay:
Ok, so I'm now up to speed on SD card optimization having read through Menneisyys' post on this. Now to the nitty gritty... I have a 2gig SANDisk card so FAT16 is pretty much out due to the huge cluster size that it would create. I'm going to eliminate my FAT backup, go with FAT32/8k cluster sizes. I forget how many entries a FAT32 table can have but if that won't give me a full 2GB then I'll bump the cluster size up to 16K.
One A2DP note however: I did test with an mp3 file located in main memory and there was no improvement in skipping, so I came to the initial conclusion that SD card access speed wasn't an issue.
Sleuth255 said:
@efjay:
Ok, so I'm now up to speed on SD card optimization having read through Menneisyys' post on this. Now to the nitty gritty... I have a 2gig SANDisk card so FAT16 is pretty much out due to the huge cluster size that it would create. I'm going to eliminate my FAT backup, go with FAT32/8k cluster sizes. I forget how many entries a FAT32 table can have but if that won't give me a full 2GB then I'll bump the cluster size up to 16K.
One A2DP note however: I did test with an mp3 file located in main memory and there was no improvement in skipping, so I came to the initial conclusion that SD card access speed wasn't an issue.
Click to expand...
Click to collapse
Let us know how it goes. I am still getting skip-free playback via A2DP. If it doesnt work maybe the additional reg tweaks I applied to the system cache are what is making the difference.
efjay said:
Sleuth255 said:
@efjay:
Ok, so I'm now up to speed on SD card optimization having read through Menneisyys' post on this. Now to the nitty gritty... I have a 2gig SANDisk card so FAT16 is pretty much out due to the huge cluster size that it would create. I'm going to eliminate my FAT backup, go with FAT32/8k cluster sizes. I forget how many entries a FAT32 table can have but if that won't give me a full 2GB then I'll bump the cluster size up to 16K.
One A2DP note however: I did test with an mp3 file located in main memory and there was no improvement in skipping, so I came to the initial conclusion that SD card access speed wasn't an issue.
Click to expand...
Click to collapse
Let us know how it goes. I am still getting skip-free playback via A2DP. If it doesnt work maybe the additional reg tweaks I applied to the system cache are what is making the difference.
Click to expand...
Click to collapse
I've found another issue with the wireless stereo profile - sometimes when a call comes in, pressing the call button on the headset will answer the call, but there is no sound (caller can't hear me and I can't hear the caller). To get the call going, I have to "disconnect handsfree" from the call in progress menu and then press the headset button again.
Anyone else seen this?
/POL
joint stereo is not mono
joint stereo is merely a way to improve compression efficiency -- and can even be used with lossless (stereo) codecs
so is there any progress? I have the same trouble with my Anycom BSH-100 .. it is skipping every 30-50s
thx
Hey guys,
I am trying to record some pcm data from the built in microphone in my windows mobile 6 phone. (ATT Tilt)
I am using Waveform Audio functions to accomplish this and have a program written that looks for available input devices, lets me select from the found devices and then proceeds to store a few second's worth of data into my buffer.
Everything goes smoothly - the device opens and all of my calls to the waveform audio functions return MMSYSERR_NOERROR. When I start recording, I physically loop for the expected amount of time while I wait for the buffer to fill and once it finishes my WAVEHDR contains the expected number of samples.
The problem is that WAVEHDR.lpData is still all zero'd out (I zero it before I begin using it), except for the first byte which is always set to 0x03.
My two questions right now are these:
1 - My phone reports a single device, whose szPname is "Audio Input". Sounds pretty generic to me. I've not been able to find anything about what I should expect to see on google or searching the forums. I am wondering if this is really all I should be seeing?
2 - Assuming I am connecting to the correct input device to receive mic input, are there any cases where it might legitimately fill my buffer with zeroes?
Thanks,
Nick