bastibe / SoundCard

A Pure-Python Real-Time Audio Library
https://soundcard.readthedocs.io
BSD 3-Clause "New" or "Revised" License
681 stars 69 forks source link

SoundCard

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SoundCard is a library for playing and recording audio without resorting to a CPython extension. Instead, it is implemented using the wonderful CFFI <http://cffi.readthedocs.io/en/latest/>__ and the native audio libraries of Linux, Windows and macOS.

The input and output data are scaled to 0dBFS (Full Scale). To avoid clipping
restrict all data between -1 and 1.

SoundCard is cross-platform, and supports Linux/pulseaudio, Mac/coreaudio, and Windows/WASAPI. While the programming interface is identical across platforms, sound card naming schemes and default block sizes can vary between devices and platforms.

SoundCard is still in development. All major features work on all platforms, but there are a few known issues that still need to be fixed. If you find a bug, please open an Issue, and I will try to fix it. Or open a Pull Request, and I will try to include your fix into SoundCard.

However, please be aware that this is a hobby project of mine that I am developing for free, and in my spare time. While I try to be as accomodating as possible, I can not guarantee a timely response to issues. Publishing Open Source Software on Github does not imply an obligation to fix your problem right now. Please be civil.

| SoundCard is licensed under the terms of the BSD 3-clause license | (c) 2016 Bastian Bechtold

|open-issues| |closed-issues| |open-prs| |closed-prs|

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Tutorial

Here is how you get to your Speakers and Microphones:

.. code:: python

import soundcard as sc

# get a list of all speakers:
speakers = sc.all_speakers()
# get the current default speaker on your system:
default_speaker = sc.default_speaker()
# get a list of all microphones:
mics = sc.all_microphones()
# get the current default microphone on your system:
default_mic = sc.default_microphone()

# search for a sound card by substring:
>>> sc.get_speaker('Scarlett')
<Speaker Focusrite Scarlett 2i2 (2 channels)>
>>> one_mic = sc.get_microphone('Scarlett')
<Microphone Focusrite Scalett 2i2 (2 channels)>
# fuzzy-search to get the same results:
one_speaker = sc.get_speaker('FS2i2')
one_mic = sc.get_microphone('FS2i2')

All of these functions return Speaker and Microphone objects, which can be used for playback and recording. All data passed in and out of these objects are frames × channels Numpy arrays.

.. code:: python

import numpy

>>> print(default_speaker)
<Speaker Focusrite Scarlett 2i2 (2 channels)>
>>> print(default_mic)
<Microphone Focusrite Scarlett 2i2 (2 channels)>

# record and play back one second of audio:
data = default_mic.record(samplerate=48000, numframes=48000)
# normalized playback
default_speaker.play(data/numpy.max(numpy.abs(data)), samplerate=48000)

# alternatively, get a `Recorder` and `Player` object
# and play or record continuously:
with default_mic.recorder(samplerate=48000) as mic, \
      default_speaker.player(samplerate=48000) as sp:
    for _ in range(100):
        data = mic.record(numframes=1024)
        sp.play(data)

Latency

By default, SoundCard records and plays at the operating system's default configuration. Particularly on laptops, this configuration might have extreme latencies, up to multiple seconds.

In order to request lower latencies, pass a blocksize to player or recorder. This tells the operating system your desired latency, and it will try to honor your request as best it can. On Windows/WASAPI, setting exclusive_mode=True might help, too (this is currently experimental).

Another source of latency is in the record function, which buffers output up to the requested numframes. In general, for optimal latency, you should use a numframes significantly lower than the blocksize above, maybe by a factor of two or four.

To get the audio data as quickly as absolutely possible, you can use numframes=None, which will return whatever audio data is available right now, without any buffering. Note that this might receive different numbers of frames each time.

With the above settings, block sizes of 256 samples or ten milliseconds are usually no problem. The total latency of playback and recording is dependent on how these buffers are handled by the operating system, though, and might be significantly higher.

Additionally, it might help to experiment with advice from here: https://askubuntu.com/questions/707171/how-can-i-fix-choppy-audio and edit your /etc/pulse/default.pa file to replace the line saying ::

load-module module-udev-detect

with ::

load-module module-udev-detect tsched=0

and then do not forget to restart pulseaudio with ::

pulseaudio -k

Channel Maps

Some professional sound cards have large numbers of channels. If you want to record or play only a subset of those channels, you can specify a channel map. A channel map consists of a list of channel specifiers, which refer to the channels of the audio backend in use. The index of each of those specifiers in the the channel map list indicates the channel index in the numpy data array used in SoundCard:

.. code:: python

# record one second of audio from backend channels 0 to 3:
data = default_mic.record(samplerate=48000, channels=[0, 1, 2, 3], numframes=48000)

# play back the recorded audio in reverse channel order:
default_speaker.play(data=data, channels=[3, 2, 1, 0], samplerate=48000)

The meaning of the channel specifiers depend on the backend in use. For WASAPI (Windows) and CoreAudio (macOS) the indices refer to the physical output channels of the sound device in use. For the PulseAudio backend (Linux) the specifiers refer to logical channel positions instead of physical hardware channels.

The channel position identifiers in the PulseAudio backend are based on: https://freedesktop.org/software/pulseaudio/doxygen/channelmap_8h.html Since the mapping of position indices to audio channels is not obvious, a dictionary containing all possible positions and channel indices can be retrieved by calling channel_name_map(). The positions for the indices up to 10 are: ::

'mono': -1,
'left': 0,
'right': 1,
'center': 2,
'rear-center': 3,
'rear-left': 4,
'rear-right': 5,
'lfe': 6,
'front-left-of-center': 7,
'front-right-of-center': 8,
'side-left': 9,
'side-right': 10

The identifier mono or the index -1 can be used for mono mix of all channels for both playback and recording. (CoreAudio/macOS defines channel -1 as silence for both playback and recording.) In addition to the indices, the PulseAudio backend allows the use of the name strings to define a channel map:

.. code:: python

# This example plays one second of noise on each channel defined in the channel map consecutively.
# The channel definition scheme using strings only works with the PulseAudio backend!

# This defines a channel map for a 7.1 audio sink device
channel_map = ['left', 'right', 'center', 'lfe', 'rear-left', 'rear-right', 'side-left', 'side-right']

num_channels = len(channel_map)
samplerate = 48000

# Create the multi channel noise array.
noise_samples = 48000
noise = numpy.random.uniform(-0.1, 0.1, noise_samples)
data = numpy.zeros((num_channels * noise_samples, num_channels), dtype=numpy.float32)
for channel in range(num_channels):
    data[channel * noise_samples:(channel + 1) * noise_samples, channel] = noise

# Playback using the 7.1 channel map.
default_speaker.play(data=data, channels=channel_map, samplerate=samplerate)

The available channels of each PulseAudio source or sink can be listed by ::

> pactl list sinks
> pactl list sources

The Channel Map property lists the channel identifier of the source/sink. ::

> pactl list sinks | grep  "Channel Map" -B 6

Sink #486
    State: SUSPENDED
    Name: alsa_output.usb-C-Media_Electronics_Inc._USB_Advanced_Audio_Device-00.analog-stereo
    Description: USB Advanced Audio Device Analog Stereo
    Driver: PipeWire
    Sample Specification: s24le 2ch 48000Hz
    Channel Map: front-left,front-right
--
Sink #488
        State: RUNNING
        Name: alsa_output.pci-0000_2f_00.4.analog-surround-71
        Description: Starship/Matisse HD Audio Controller Analog Surround 7.1
        Driver: PipeWire
        Sample Specification: s32le 8ch 48000Hz
        Channel Map: front-left,front-right,rear-left,rear-right,front-center,lfe,side-left,side-right

FAQ

Q: How to make it work on a headless Raspberry Pi?

A: PulseAudio is not installed by default on the Raspberry Pi OS Lite distribution (https://www.raspberrypi.org/software/operating-systems/). In order to use soundcard, you have to install PulseAudio first, and edit the configuration (with a fix to avoid the main output to be in mono-only). ::

sudo apt install -y python3-pip python3-numpy pulseaudio
sudo nano /usr/share/pulseaudio/alsa-mixer/profile-sets/default.conf
# comment the block [Mapping analog-mono] with ';'
pulseaudio -D
python3 -m pip install soundcard

Known Issues:

Changelog