Attach portaudio error codes to portaudio errors to allow for better error handling. This would allow users to handle specific portaudio and host api error codes. The _check function was also updated to use these new exceptions. The from_errno and from_errinfo are a convenience that I thought might make it easier for users to raise an exception with a default message. Hopefully this doesn't add too much complexity to the code.

These 3 commits access PA's Core Audio API from pa_mac_core.h, and provide a "get_channel_name" function for querying names of individual device channels along with MacCoreSettings (for use with extra_settings) to expose additional Core Audio capabilities. (See doc string in MacCoreSettings.)

My specific need for these extensions is that I have an RME Fireface UFX+ with a large number of input & output channels, and I didn't want to have to open many channels when only a couple are required. The get_channel_name function lets me display obtain a set of user-friendly channel names from the underlying device driver (via PortAudio), which I may display to the operator / user.

I'm open for input to help get these changes incorporated into python-sounddevice.

Thank you, David

Hi, I'm very new to Python and Spyder. I'm getting this on the Spyder console, seems related to #110 but I'm not sure what to make of it.

import sounddevice as sd Traceback (most recent call last):

File "", line 1, in import sounddevice as sd

File "C:\Users\James\Anaconda3\lib\site-packages\sounddevice.py", line 73, in import _sounddevice_data

ModuleNotFoundError: No module named '_sounddevice_data'

Hello, I'm trying to record audio with sounddevice library (latest version) on RaspberryPi 3 model B v1.2 with raspbian installed and updated. I'm using usb sound card for microphone input. Here is the simple script for recording: sound.zip

My problem is that sounddevice records so much noise, that it hurts my ears when I listen to the recorded file. Sometimes after fresh RPi boot, the first recording works ok, but when I try to record second time, then it records the noise. Other times (most of the time), it records the noise even after fresh boot.

I don't think its because of the usb sound card, because the script I'm using works ok with that card on my desktop pc (arch linux) and with the same version of the sounddevice.

I've tired recording with arecord program, which uses alsa and it works ok every time and I've also tried recording with audacity, which uses portaudio library and it also works ok. Do you have any idea why is this happening?

Thanks

Hello

I am on Arch Linux and I started using SD today. My initial tests worked just fine. But when I opened another application like Spotify that play music (It is just opened but it doesn't play anything) and then tried to play a sound, it doesn't work anymore. Even if I closed Spotify it won't work till I restart my PC.

Devices are listed normally and it doesn't return any errors. Is this a normal behaviour or is it a bug?

And just a small question, Can I play during the playing of another sound? For instance, if a sound's duration is 1s, can I play the sound then after 0.5s play it again without the second sound killing the first sound (I want them to play at the same time).

Thanks.

Hi, i have a strange problem, but i cant see how to come closer to the cause.

I use two (software) systems: yours and pydub see: https://github.com/jiaaro/pydub And i have two hardware systems: one is Ubuntu 16.04 and the other a Jessie for RaspPi(3).

If i let sounddevice play a ogg fiel on Rasp, it works. But it fails on ubuntu (i hear noise). If i do the same with pydub no problem with Ubuntu and on RaspPi , its in worse quality (not useful, but i can guess the voice file - whereas on first, its just noise).

Jürgen

On a german Windows 7 machine with sounddevice 0.3.3, I get a traceback (see below) when I call:

python -m sounddevice

The reason seems to be that a german Windows version has non ASCII characters in the device name. (e.g.: 'Primärer Soundaufnahmetreiber').

Possible Solution:

Line 1806: Change name=info['name'], To: name=repr(info['name']),

Traceback (most recent call last):
  File "C:\Python27\lib\runpy.py", line 162, in _run_module_as_main
    "__main__", fname, loader, pkg_name)
  File "C:\Python27\lib\runpy.py", line 72, in _run_code
    exec code in run_globals
  File "C:\Python27\lib\site-packages\sounddevice.py", line 2536, in <module>
    print(query_devices())
  File "C:\Python27\lib\site-packages\sounddevice.py", line 1810, in __repr__
    for idx, info in enumerate(self))
  File "C:\Python27\lib\site-packages\sounddevice.py", line 1810, in <genexpr>
    for idx, info in enumerate(self))
UnicodeEncodeError: 'ascii' codec can't encode character u'\ufffd' in position 18: ordinal not in range(128)

When using WASAPI shared mode in Windows, it is necessary to match the samplerate in sounddevice and Control Panel > Sound > device > Properties > Advanced tab > Default Format. In sounddevice v0.3.5, a samplerate mismatch throws an exception "sounddevice.PortAudioError: Error opening RawOutputStream: Invalid device". I propose to instead detect the mismatch to the device default samplerate and clarify the exception message. This will better help users understand and fix this type of problem. Often, they will want to change the Windows setting.

python-sounddevice already has a nice callback interface. Do you think it would be possible to add a module within sounddevice that exposes an asyncio interface for it? This would let you do things like:

import sounddevice._async as sd

async def some_coroutine():
    async for chunk in sd.Stream(channels=2):
        # Process chunk, but let Python do things while waiting for the next chunk

I'd be happy to implement this if there's any interest. If not I'll write a separate async wrapper module that imports python-sounddevice. Let me know!

query_devices() crashes:

    File "C:\Program Files\Anaconda3\lib\site-packages\sounddevice.py", line 712, in query_devices
    _lib.Pa_HostApiTypeIdToHostApiIndex(_lib.paMME)) else 'utf-8'),
   UnicodeDecodeError: 'mbcs' codec can't decode bytes in position 0--1: No mapping for the Unicode   character exists in the target code page.

Changing the "mbcs" at line 710 to "utf-8" fixed the problem in my environment.

OS: Windows 7 SP1 and Windows 10 (Japanese) Python 3.5 sounddevice ver 0.3.6 (wheel from http://www.lfd.uci.edu/~gohlke/pythonlibs/)

People who want to create their own callback-based stream classes can directly inherit from _StreamBase, avoiding the unnecessary (and non-functional) read() and write() methods.

Also, if desired, they can quite easily implement a single stream class that (based on constructor arguments) supports only inputs, only outputs, or both inputs and outputs.

I am trying to write/use your code to make a recording that terminates when either 1) there is a keyboard press, or 2) there is a (configurable) number of seconds that have passed without any speech.

Normally when I want to run one python file within another, I will use the subprocess function. But when I try to use subprocess to run rec_unlimited.py, I can't use KeyboardInterrupt to stop the recording b/c it will simply stop the subprocess rather than the recording.

Does this make sense? Sorry I am a psychologist - not a computer scientist. Do you have any suggestions or resources you can point me towards to get what I described in the first paragraph?

Thanks :)

Is there a method to list asound.conf pcms?

pcm.donglein {
    type=plug
    slave = {
    pcm "plughw:3,0"
    channels 1
    rate 4800}
}

It will list everything else but the pcm I want to use (rate change)

(venv) stuart@ubuntu:~/g-kws$ python3 -m sounddevice --help
   0 Loopback: PCM (hw:0,0), ALSA (32 in, 32 out)
   1 Loopback: PCM (hw:0,1), ALSA (32 in, 32 out)
   2 HDA Intel PCH: ALC221 Analog (hw:1,0), ALSA (2 in, 2 out)
   3 HDA Intel PCH: ALC221 Alt Analog (hw:1,2), ALSA (2 in, 0 out)
   4 HDA NVidia: HDMI 0 (hw:2,3), ALSA (0 in, 8 out)
   5 HDA NVidia: HDMI 1 (hw:2,7), ALSA (0 in, 2 out)
   6 HDA NVidia: HDMI 2 (hw:2,8), ALSA (0 in, 8 out)
   7 HDA NVidia: HDMI 3 (hw:2,9), ALSA (0 in, 8 out)
   8 HDA NVidia: HDMI 4 (hw:2,10), ALSA (0 in, 8 out)
   9 HDA NVidia: HDMI 5 (hw:2,11), ALSA (0 in, 8 out)
  10 HDA NVidia: HDMI 6 (hw:2,12), ALSA (0 in, 8 out)
  11 VIA USB Dongle: Audio (hw:3,0), ALSA (2 in, 2 out)
  12 USB Audio Device: - (hw:4,0), ALSA (1 in, 2 out)
  13 sysdefault, ALSA (128 in, 128 out)
  14 front, ALSA (0 in, 32 out)
  15 surround21, ALSA (32 in, 32 out)
  16 surround40, ALSA (0 in, 32 out)
  17 surround41, ALSA (0 in, 32 out)
  18 surround50, ALSA (0 in, 32 out)
  19 surround51, ALSA (0 in, 32 out)
  20 surround71, ALSA (0 in, 32 out)
  21 lavrate, ALSA (128 in, 128 out)
  22 samplerate, ALSA (128 in, 128 out)
  23 speexrate, ALSA (128 in, 128 out)
  24 pulse, ALSA (32 in, 32 out)
  25 speex, ALSA (1 in, 1 out)
  26 upmix, ALSA (8 in, 8 out)
  27 vdownmix, ALSA (6 in, 6 out)
  28 pipewire, ALSA (64 in, 64 out)
  29 dmix, ALSA (0 in, 2 out)
* 30 default, ALSA (32 in, 32 out)

Maybe it has something to do with Pulseaudio or there is a limit on devices returned?

Just tried a lite version on a pi3 that doesn't have pulseaudio running

* 0 VIA USB Dongle: Audio (hw:1,0), ALSA (2 in, 2 out)
  1 ladspa, ALSA (128 in, 128 out)
  2 deepfilter, ALSA (128 in, 128 out)
  3 donglein, ALSA (128 in, 128 out)

All those are alsa pcms that just don't show on my x86 machine with pulseaudio / desktop install, but list fine on the pi

Hi,

I can run the python code directly no problem from python. C++ compiled code run from Terminal (macos) Unix file also works. I am having some trouble when double clicking the package created by cmake for sounddevice to receive data.

        q = queue.Queue()
        def callback(indata, frames, time, status):
            q.put(bytes(indata))
        if args.samplerate is None:
            device_info = sd.query_devices(args.device, 'input')
            args.samplerate = int(device_info['default_samplerate'])
        with sd.RawInputStream(samplerate=args.samplerate, blocksize = 8000, device=args.device, dtype='int16',
                                    channels=1, callback=callback):
         ....

In c++ code:

    Py_Initialize();
    fp = _Py_wfopen("test.py", L"r");
    PyRun_SimpleFile(fp, "test.py");
    Py_Finalize();

Where test.py is the working python code. Compiled everything runs fine when running the file from terminal but when compiled into a "dragndrop" from cmake it can get the device information (and sample rate) but it gets no data in the "q" variable.

Any ideas what I might be missing?

Thanks!

I'm not quite sure yet if we should do this, this has to be tested thoroughly.

With a bit of luck, this might fix #90.

However, it might also have bad side effects ...

I want to use a pyqt5 GUI to playback a sound with a soundcard. The index of my asio Fireface USB is 66.

import sounddevice as sd
data, fs = sf.read(....)
asio_out = sd.AsioSettings(channel_selectors=[1, 2, 3, 4, 5,6,7,8])
sd.default.extra_settings = None,asio_out
sd.default.device[1] = 66
# mapping from 1 to 8
sd.play(data, samplerate=fs,mapping=[1])
sd.wait()

its work.so i try to use this function within a pyqt5 GUI and create a thread to control.

class New_thread(Qthread):
     def __init__(self,wave_dir,flag):
           super(New_thread,self).__init__()
           self.flag = flag
           self.wave_dir = wave_dir
     def run(self):
           if self.flag:
                 # run 
                 data,fs = soundfile.read(self.weave_dir)
                 asio_out = sd.AsioSetting(channel_selections = [1,2,3,4,5,6,7,8])
                 sd.default.extra_setting = None,asio_out
                 sd.default.device[1] = 66
                 sd.play(data, samplerate=fs,mapping=[1])
                 sd.wait()
           else:
                 # create new thread to stop playing back
                 data = np.empty((1000))
                 sd.default.device[1] = 66
                 sd.play(data, samplerate=48000,mapping=[1])
                 sd.stop()

but its fail.

Traceback (most recent call last):
  File ".\GUI_FOR_LISTENING_TEST.py", line 64, in run
    sd.play(data, samplerate=fs,mapping=[1])
  File "D:\Users\Administrator\miniconda3\envs\listening3.8\lib\site-packages\sounddevice.py", line 175, in play
    ctx.start_stream(OutputStream, samplerate, ctx.output_channels,
  File "D:\Users\Administrator\miniconda3\envs\listening3.8\lib\site-packages\sounddevice.py", line 2582, in start_stream
    self.stream = StreamClass(samplerate=samplerate,
  File "D:\Users\Administrator\miniconda3\envs\listening3.8\lib\site-packages\sounddevice.py", line 1494, in __init__
    _StreamBase.__init__(self, kind='output', wrap_callback='array',
  File "D:\Users\Administrator\miniconda3\envs\listening3.8\lib\site-packages\sounddevice.py", line 898, in __init__
    _check(_lib.Pa_OpenStream(self._ptr, iparameters, oparameters,
  File "D:\Users\Administrator\miniconda3\envs\listening3.8\lib\site-packages\sounddevice.py", line 2745, in _check
    raise PortAudioError(errormsg, err, hosterror_info)
sounddevice.PortAudioError: Error opening OutputStream: Unanticipated host error [PaErrorCode -9999]: 'Failed to load ASIO driver' [ASIO error 0]

how can we fix it,thanks a lot.

soundcare info:Fireface UFX, connected with 8 loudspeakers

good day,my bro! i just want to playback some audios and meet some probles,to be specific,could you tell us the usage of sounddevic.play(,,mapping=), especially the "mapping=". thanks a lot!