gmane.comp.audio.portaudio.devel

Compiling Portaudio for OSX 10.8.3

Wouter Janmaat — 2013-05-13T14:39:02

Hi All,

it's my first time using PortAudio and I'm currently trying to compile for OSX 10.8.3 using the ./config && make commands from the tutorial but I keep getting errors. They look like this:

checking build system type... i386-apple-darwin12.3.0
checking host system type... i386-apple-darwin12.3.0
checking target system type... i386-apple-darwin12.3.0
checking for gcc... no
checking for cc... no
checking for cl.exe... no
configure: error: in `/Users/wouterjanmaat/Downloads/portaudio':
configure: error: no acceptable C compiler found in $PATH
See `config.log' for more details


I tried using the disable universal binaries flag but that didn't help.I tried both the latest nightly build and the latest stable build.

Please help me out

Thanks

Wouter

Linux Latency problem

Alexander Carôt — 2013-05-13T07:45:10

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Portaudio and Remote Desktop Sessions

Geordi Sinclair — 2013-05-10T17:18:57

Hello,

I am using the latest version of Portaudio, from svn.  It is built into iaxclient.dll.  It works very well on Windows XP and Windows 7 , but I have having odd issues with the audio when using Remote Desktop Connection (from windows 7 64 bit) to a remote desktop session (Windows 2008 Server).  I allow the microphone and speaker of windows 7 to be available over the RDC.  I can see them on the RDS, as "Remote Audio", but when I use them, the audio is very broken up in both directions.

If I check the audio from the local mic to the RDS, via an audio recorder, the quality is fine.

Any ideas?

Thanks

Geordi
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Buffer alignment requirement of various audio format

Pradeep Kumar — 2013-05-07T07:42:32

Hi All,

I've some question on requirement of alignment of data buffer used by
Pa_WriteStream or Pa_Callback.
I want to support for two audio format paInt16 and paFloat32.
So do I've to align my data buffer on 32byte boundary for paFloat32 and
16byte boundary of paInt16?

Also if I use Pa_RingBuffer in my application do I ever need to worry about
alignment or ring buffer internally take care of alignment?

Please share some thoughts.

Regards,
Pradeep
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Pa_Initialize crash

Sergio Teran — 2013-05-06T10:32:43

Hello,

I am trying to use portaudio in my Qt desktop app to use the audio
IO functionality  but I am not being able to initialize the library.

My project will build fine but as soon as I execute it it will crash
automatically if I have the Pa_Initialize() call in it. If I comment it out
it continues with no problem.

I did a whole new blank project and I only have this in it:

MainWindow::MainWindow(QWidget *parent)

    : QMainWindow(parent)

{

    // Test UI with just one button

     button = new QPushButton(tr("OKK"), this);

    connect(button, SIGNAL(clicked()), this, SLOT(close()));


    PaError err;

    err = Pa_Initialize();


    if(err != paNoError){

        qDebug() << "Pa_Initialize passed ... ";

        return;

    }

    else

        qDebug() << "THERE WAS AN ERROR";

}


can anyone help me out to what could be the problem?

The error I get when the program stops is:


The program has unexpectedly finished.

C:\Qt\dev\Projects\build-PaTest_record-Desktop_Qt_5_0_2_MinGW_32bit-Debug\deb

Noise when writing silence in pa_callback

Pradeep Kumar — 2013-05-02T11:59:33

Hi,

I've an user application which does both audio and video processing. I'm
getting loud noise whenever the playback is interrupted in the middle i.e
if application is playing a video file and suddenly user initiate some
other task from application, then playback is interrupted to its current
state  till the new task finishes, Now during this interruption since I'm
not decoding any audio sample my ring buffer is empty and I'm writing
silence in callback. However I'm hearing loud noise instead.
Please suggest what could be going wrong.

Thanks and Regards,
Pradeep
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CMake project fix on Windows

adrien courdavault — 2013-05-02T08:32:58

HI

I'm a portaudio user, and first of all thank you all for your work.

Also I like to use CMake as build system, and this morning when I
tried to build with visual stuio 11 express on my win7 (note asio
build was not tested) the portaudio project after a project generation
(cmake -G"Visual Sutio 11")
I could not because of a dependency on ksguid.lib


A quick google search gave me this thread.
http://music.columbia.edu/pipermail/portaudio/2011-August/012849.html

As it is pretty old I thought it would have been integrated, but I
tried this patch on the cmake project to test just in case :


Index: CMakeLists.txt
===================================================================
--- CMakeLists.txt    (revision 1890)
+++ CMakeLists.txt    (working copy)
< at >< at > -269,7 +269,7 < at >< at >
 INCLUDE_DIRECTORIES( src/common )

 IF(WIN32 AND MSVC)
-ADD_DEFINITIONS(-D_CRT_SECURE_NO_WARNINGS)
+ADD_DEFINITIONS(-D_CRT_SECURE_NO_WARNINGS -DPA_WDMKS_NO_KSGUID_LIB)
 ENDIF(WIN32 AND MSVC)

 ADD_DEFINITIONS(-DPORTAUDIO_CMAKE_GENERATED)

PA/ASIO: MOTU drivers causing Pa_Initialize() to fail -- possible fix committed

Ross Bencina — 2013-05-02T01:23:33

Hi Everyone,

I recently had a report of Pa_Initialize() completely failing on Windows 
if a MOTU ASIO driver was installed but the device was not connected and 
powered on.

This failure would affect all host APIs not just ASIO.

More details here:

https://www.assembla.com/spaces/portaudio/tickets/221

I have comitted a possible fix:

https://www.assembla.com/code/portaudio/subversion/commit/1890

In summary: I moved most of the device parsing code to a separate 
function. If any error is encountered during this phase the device is 
skipped, but Pa_Initialize() will proceed with other devices and no 
error will be returned.

Since this is a critical-path change to device scanning I wanted 
everyone to be aware of the change. You might care if either:

a) I made an error in the fix
b) Your software will fail if the MOTU drivers are installed

Either way I would appreciate it if you could review my checkin and test 
the change.

This also highlights the problems of trying to map multiple failure 
modes to a

Unescaped paths in configure script

Tim Neumann — 2013-04-27T15:09:02

I just ran the latest configure script (svn snapshot v19 27-Apr-2013
01:23:06 UTC)
and the script stumbled when checking
$PKG_CONFIG
because my pkg_config was located at
/c/Program Files/GnuWin32/bin/pkg-config
(running under mingw/msys)
Notice the space between Program _ Files.
This can be easily fixed by properly escaping/evaluating the whole variable
like this:
$"$PKG_CONFIG"

The error occured twice, once on line 15478 and on line 15533.

-Greetings
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Portaudio on the Raspberry Pi

Derek Gonyeo — 2013-04-27T00:41:31

Hi,

Sorry if this is the wrong place for this, but I'm having problems using
portaudio on the raspberry pi in C.

The code I have worked on my laptop, but for my project I need it to work
on the Raspberry Pi. I simply want to store the values given to me each
callback in an array. My problem is that I can't seem to be able to find
any audio devices. When I call Pa_GetDeviceCount(), it returns 0. Mplayer
is able to output audio.

My code is available at
https://github.com/dgonyeo/odd/blob/master/computer_program/odd_audio.c if
it helps. It's being run as root.

I have a USB dongle thingy that has a mic in plugged into it, that's just
plug and play in Fedora.

I'm currently running raspbian, a debian based distro for the pi. Kernel
3.6.11.

Is this a problem with the pi, or am I missing something obvious?

Thanks,
Derek Gonyeo
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Channel count must be a power of 2 with Mac CoreAudioHostAPI

Cartwright, Richard — 2013-04-24T06:29:05

Hi All,

It looks like the Mac CoreAudio HostAPI makes an assumption that the
number of channels on the device is a power of 2. I have a device (Echo
AudioFire12) for which the assumption doesn't hold and am looking for
suggestions on how best to patch PortAudio to fix.

pa_mac_core_blocking.c line 173:
      err = PaUtil_InitializeRingBuffer(
            &blio->inputRingBuffer,
            1, ringBufferSize*blio->inputSampleSizePow2*inChan,
            data );

pa_ringbuffer.c line 66:
ring_buffer_size_t PaUtil_InitializeRingBuffer( PaUtilRingBuffer *rbuf,
ring_buffer_size_t elementSizeBytes, ring_buffer_size_t elementCount, void
*dataPtr )
{
if( ((elementCount-1) & elementCount) != 0) return -1; /* Not Power of
two. */
...

In the calling code, ringBufferSize has been carefully chosen to be a
power of 2 (computeRingBufferSize(): pa_mac_core_utilities.c lines 283 and
following). Similarly blio->inputSampleSizePow2 is a power of 2. However,
inChan is not necessarily a power of 2, which means that
PaUtil_In

Fast Audio playback in Portaudio

Pradeep Kumar — 2013-04-16T14:51:11

Hi

I'm extracting audio samples using ffms2 and sending them to audio output
using portaudio.
Things where working fine but recently I did some restructuring and
optimization in the code due to which
audio playback has become very fast.
Earlier it was working with normal rate but after restructuring it has
become quite fast.

In my code I've just replace some for loops with memcpy to speed up the
things. The audio playback is part of
another application which also does video playback. So I tried to optimize
the function which was rendering both audio and video.

The audio parameters are, sample rate: 44100/22050(as per audio file),
sample format 32bit float, buffer len ; 2048
Can some suggest what could be going wrong. I tried to play with latency
but it is not helping.

Another issue which I'm facing is that for mono channel audio is quite
clear but for stereo I'm getting noise in my playback.

Here is my callback function and sample filling function.
The two buffer are filler based on which buffer is used

PortAudio Installation

sinem cimen — 2013-04-12T10:11:10

I installed portaudio and follow read me txt. and run cpp code it works
before, but the second time when I tried to use portaudio it gives an error
like:
in compiling of 'make all ' part.


root< at >ubuntu:/usr/src/portaudio/bindings/cpp/build/gnu# make all
g++ -shared -o ../../lib/libportaudiocpp.so
../../source/portaudiocpp/BlockingStream.o
../../source/portaudiocpp/CallbackInterface.o
../../source/portaudiocpp/CallbackStream.o
../../source/portaudiocpp/CFunCallbackStream.o
../../source/portaudiocpp/CppFunCallbackStream.o
../../source/portaudiocpp/Device.o
../../source/portaudiocpp/DirectionSpecificStreamParameters.o
../../source/portaudiocpp/Exception.o ../../source/portaudiocpp/HostApi.o
../../source/portaudiocpp/InterfaceCallbackStream.o
../../source/portaudiocpp/MemFunCallbackStream.o
../../source/portaudiocpp/Stream.o
../../source/portaudiocpp/StreamParameters.o
../../source/portaudiocpp/System.o
../../source/portaudiocpp/SystemDeviceIterator.o
../../source/portaudiocpp/SystemHostApiIterator.o
/usr/bin/ld

cracking noises on Mac with samplerate = 48kHz

Albert Zeyer — 2013-04-10T20:21:53

Hi,

I noticed some cracking/clicking noises on Mac. I use samplerate = 48kHz.

They are not that easy to reproduce. They don't appear always. When
they appear, it's maybe every 3 seconds or so and more often during
loud parts, often after some bass part.

I think I have never noticed them with samplerate = 44.1kHz.

I am just calling

ret = Pa_OpenDefaultStream(
  &player->outStream->stream,
  0,
  player->outNumChannels, // numOutputChannels
  paInt16, // sampleFormat
  player->outSamplerate, // sampleRate
  AUDIO_BUFFER_SIZE / 2, // framesPerBuffer,
  &paStreamCallback,
  player //void *userData
  );

I am using PortAudio for a music player. The code can be found here:
https://github.com/albertz/music-player/

I want to use the highest possible sample rate. How can I do that?

I haven't passed any flags to PortAudio before. Which one do make
sense? I don't fully understand all of them. The documentation lacks
some information there.

paClipOff: What is the default clipping? What samples are out-of-range?

Real Time Audio Play on PC with Windows XP

Umar Hamid — 2013-04-07T13:48:19

Dear all,

I am new to audio programming and want to use portaudio for real time audio
play on PC with windows XP.

I have one computer that is generating a sine wave of a particular
frequency having 1024 time samples and the

sampling rate is 3.2 kHz. Now i send this data to another computer over LAN
using TCP/IP. I have gone through

the built-in test example of pa_sine.c in the portaudio example folder.

I want to play the sine wave received on the 2nd computer in real time with
the same sampling rate using portaudio.

When the sine wave is played with sampling rate (3200) and buffers (64) and
tab_size (1024) there are breaks/glitches

in the audio.

I have also used another approach. I made a circular buffer of 3200. The
incoming data of 1024 samples comes into this buffer.

When the buffer the filled it gets play with sampling rate (3200) and
buffers (64) and tab_size (3200) but still the problem

of breaks/glitches in the audio appears after some time of program run.

Kindly provide your help in this r

Installing Portaudio Ubuntu 12.04

Samir Trajano Feitosa — 2013-04-05T13:53:09

Hi!
Please, help me to install portaudio in ubuntu.
In that folder should I install linux portaudio?
How should I install?
How should I compile some C code that uses portaudio?
I only get this kind of message.:
*undefined reference to* `*Pa_IsFormatSupported*'

Thank you.

portaudio/asio with asio4all callback schedulingquestion

Andras Kutrovics — 2013-03-31T19:30:56

Hi All,

I'm using portaudio/asio with asio4all driver as part of an
application which does constant streaming to several usb audio devices. OS
is Windows XP.

My callback is just copying data between application buffers and portaudio
buffers (its runtime is around 1 ms).
The requested framesperbuffer is fix 160 sample, samplerate is 8000Hz, so I
expect the callback is invoked at every 20ms.

Everything is fine, but I'm started to do some benchmarks:
sometimes the callback is scheduled only at around 27-30 ms.
The next schedule is always 10-15ms respectively, so it seems portaudio or
asio4all driver tries to compensate, and keep 20ms in average.

I cannot hear any gaps when this 10 ms delay happens, I just need to
explain this behaviour.
My understanding is that ASIO has double-buffered architecture. It
concludes that 30ms callback frequency is not a problem,
as it's still outputting the previous 20ms. As long as the average callback
frequency is maintained as above, it should be fine.

Is it a correct assum

Failed to update list of audio devices

Alexander Klyushin — 2013-03-29T16:58:42

Hi community,



I have a problem with device refreshing.

Attached is an example code I’ve used to reproduce the problem(Slightly
changed pa_devs example)

1.    Start the example

2.    Press enter

3.    Connect or disconnect audio devices and press the enter key

4.    Compare two outputs they are identical. Number of devices was not
changed.

Looks like portaudio did not see new connected or disconnected audio
devices at all



The problem is valid for Mac OS X 10.8.2. Portaudio version: *
pa_stable_v19_20111121.tgz*<http://www.portaudio.com/archives/pa_stable_v19_20111121.tgz>
* **, November 21, 2011, SVN rev 1788*

Sorry if I missed something.



Any feedback will be appreciated

Alexander Klyushin
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Can't record on Linux with USB sound card.

Klavs Taube — 2013-03-28T07:10:11

Hello. I've implemented sound card testing software using PortAudio. It works fine on Mac, but it doesn't on Linux ( Ubuntu 12.04). After I install the PortAudio library everything works fine, but after some time (perhaps, system reboot), it won't record sound anymore. I understand this is a system configuration fault. What could be the problem? I'm pretty sure I don't do any updates after I install PortAudio. Although, before installing it, I apt-get install libasound2 and libasound2-dev.
Best regards, Klavs Taube



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Patest_multi_sine.c only plays on front 2 channels

Anonymous — 2013-03-28T02:59:24

I am trying to get started using PortAudio with 5.1 channels.  My  
setup is a 32-bit PC running Puppy Linux Slacko 5.5. It has a  
SoundBlaster SB0200 with EMU10k1x chip. The Alsa library is v1.0.26,  
and driver is v1.0.24. I am using pa_stable_v19_20111121.tgz.  I have  
tested all 5.1 channels using this command:

% speaker-test -Dplug:surround51 -c6

This works properly on all 6 channels, though it complains many times  
of broken pipes.  Compiling the patest_multi_sine.c example works  
only on front-right and front-left channels.  It reports two sets of  
64 interleaved channels.  patest_multi_sine only specifies 6  
channels, though, and not the surround51 selection.  So I tried  
setting surround51 as the default.

Running aplay -L reveals:

surround51:CARD=Live,DEV=0

Based on this, I created ~/.asoundrc containing:

pcm.!default surround51:Live

Now by default, speaker test does not need the -Dplug:surround51  
switch, so this works on all 6 channels:

% speaker-test -c6

And patest_multi_sine rep

Pausing the playback

Roman Cheplyaka — 2013-03-25T14:34:13

Hello,

This message[1] suggests that, to pause the audio playback, the callback
should be outputting silence.

[1]: http://music.columbia.edu/pipermail/portaudio/2003-November/002680.html

A more natural way to do it, it seems, is just not to return from the
callback until the playback should be resumed.

Are there any downsides to this solution?

Roman