Compare commits

..

No commits in common. "master" and "nocallback" have entirely different histories.

32 changed files with 697 additions and 2223 deletions

View file

@ -1,9 +0,0 @@
always_for_in = true
whitespace_typedefs = true
whitespace_ops_in_indices = true
remove_extra_newlines = true
import_to_using = true
short_to_long_function_def = true
format_docstrings = true
align_pair_arrow = false
conditional_to_if = true

31
.appveyor.yml Normal file
View file

@ -0,0 +1,31 @@
# Documentation: https://github.com/JuliaCI/Appveyor.jl
environment:
matrix:
- julia_version: 1
- julia_version: nightly
platform:
- x86
- x64
matrix:
allow_failures:
- julia_version: nightly
branches:
only:
- master
- /release-.*/
notifications:
- provider: Email
on_build_success: false
on_build_failure: false
on_build_status_changed: true
install:
- ps: iex ((new-object net.webclient).DownloadString("https://raw.githubusercontent.com/JuliaCI/Appveyor.jl/version-1/bin/install.ps1"))
build_script:
- echo "%JL_BUILD_SCRIPT%"
- C:\julia\bin\julia -e "%JL_BUILD_SCRIPT%"
test_script:
- echo "%JL_TEST_SCRIPT%"
- C:\julia\bin\julia -e "%JL_TEST_SCRIPT%"
on_success:
- echo "%JL_CODECOV_SCRIPT%"
- C:\julia\bin\julia -e "%JL_CODECOV_SCRIPT%"

View file

@ -1,22 +0,0 @@
Thanks for contributing a pull request!
Please be aware that we are a loose team of volunteers so patience is
necessary. Assistance handling other issues is very welcome. We value
all user contributions, no matter how minor they are. If we are slow to
review, either the pull request needs some benchmarking, tinkering,
convincing, etc. or more likely the reviewers are simply busy. In either
case, we ask for your understanding during the review process.
Again, thanks for contributing!
#### What does this implement/fix?
Explain your changes. Please be as descriptive as possible.
#### Reference issue
Example: Fixes #1234.
#### Additional information
Any additional information you think is important.

View file

@ -1,25 +0,0 @@
name: CompatHelper
on:
schedule:
- cron: 0 0 * * *
workflow_dispatch:
jobs:
CompatHelper:
runs-on: ubuntu-latest
steps:
- name: "Install CompatHelper"
run: |
import Pkg
name = "CompatHelper"
uuid = "aa819f21-2bde-4658-8897-bab36330d9b7"
version = "2"
Pkg.add(; name, uuid, version)
shell: julia --color=yes {0}
- name: "Run CompatHelper"
run: |
import CompatHelper
CompatHelper.main()
shell: julia --color=yes {0}
env:
GITHUB_TOKEN: ${{ secrets.GITHUB_TOKEN }}
COMPATHELPER_PRIV: ${{ secrets.COMPATHELPER_PRIV }}

View file

@ -1,16 +0,0 @@
name: Build documentation
on:
push:
branches:
- 'master'
jobs:
document:
runs-on: ubuntu-latest
steps:
- uses: actions/checkout@v2
- uses: julia-actions/setup-julia@latest
with:
version: '1.6'
- uses: julia-actions/julia-docdeploy@releases/v1
env:
GITHUB_TOKEN: ${{ secrets.GITHUB_TOKEN }}

View file

@ -1,15 +0,0 @@
name: TagBot
on:
issue_comment:
types:
- created
workflow_dispatch:
jobs:
TagBot:
if: github.event_name == 'workflow_dispatch' || github.actor == 'JuliaTagBot'
runs-on: ubuntu-latest
steps:
- uses: JuliaRegistries/TagBot@v1
with:
token: ${{ secrets.GITHUB_TOKEN }}
ssh: ${{ secrets.COMPATHELPER_PRIV }}

View file

@ -1,41 +0,0 @@
name: Tests
on:
pull_request:
push:
branches:
- master
tags: '*'
jobs:
test:
timeout-minutes: 30
name: ${{ matrix.version }} - ${{ matrix.os }} - ${{ matrix.arch }}
runs-on: ${{ matrix.os }}
strategy:
fail-fast: false
matrix:
version:
- '1.6'
- '1'
- 'nightly'
os:
- ubuntu-latest
- macOS-latest
- windows-latest
arch:
- x64
- x86
steps:
- uses: actions/checkout@v2
- uses: julia-actions/setup-julia@v1
with:
version: ${{ matrix.version }}
- uses: julia-actions/julia-buildpkg@v1
- uses: julia-actions/julia-runtest@v1
- uses: julia-actions/julia-processcoverage@v1
- uses: codecov/codecov-action@v1
with:
file: lcov.info

1
.gitignore vendored
View file

@ -2,7 +2,6 @@
*.o *.o
deps/deps.jl deps/deps.jl
deps/build.log deps/build.log
docs/build
*.wav *.wav
*.flac *.flac
*.cov *.cov

16
.travis.yml Normal file
View file

@ -0,0 +1,16 @@
# Documentation: http://docs.travis-ci.com/user/languages/julia/
language: julia
os:
- linux
- osx
julia:
- 1
- nightly
matrix:
allow_failures:
- julia: nightly
fast_finish: true
notifications:
email: true
after_success:
- julia -e 'using Pkg; Pkg.add("Coverage"); using Coverage; Codecov.submit(process_folder())'

View file

@ -1,26 +1,20 @@
name = "PortAudio" name = "PortAudio"
uuid = "80ea8bcb-4634-5cb3-8ee8-a132660d1d2d" uuid = "80ea8bcb-4634-5cb3-8ee8-a132660d1d2d"
repo = "https://github.com/JuliaAudio/PortAudio.jl.git" repo = "https://github.com/JuliaAudio/PortAudio.jl.git"
version = "1.3.0" version = "1.1.0"
[deps] [deps]
alsa_plugins_jll = "5ac2f6bb-493e-5871-9171-112d4c21a6e7" Libdl = "8f399da3-3557-5675-b5ff-fb832c97cbdb"
libportaudio_jll = "2d7b7beb-0762-5160-978e-1ab83a1e8a31"
LinearAlgebra = "37e2e46d-f89d-539d-b4ee-838fcccc9c8e" LinearAlgebra = "37e2e46d-f89d-539d-b4ee-838fcccc9c8e"
SampledSignals = "bd7594eb-a658-542f-9e75-4c4d8908c167" SampledSignals = "bd7594eb-a658-542f-9e75-4c4d8908c167"
Suppressor = "fd094767-a336-5f1f-9728-57cf17d0bbfb" libportaudio_jll = "2d7b7beb-0762-5160-978e-1ab83a1e8a31"
[compat] [compat]
julia = "1.6" julia = "1.3"
alsa_plugins_jll = "1.2.2"
libportaudio_jll = "19.6.0"
SampledSignals = "2.1.1"
Suppressor = "0.2"
[extras] [extras]
Documenter = "e30172f5-a6a5-5a46-863b-614d45cd2de4"
LibSndFile = "b13ce0c6-77b0-50c6-a2db-140568b8d1a5"
Test = "8dfed614-e22c-5e08-85e1-65c5234f0b40" Test = "8dfed614-e22c-5e08-85e1-65c5234f0b40"
TestSetExtensions = "98d24dd4-01ad-11ea-1b02-c9a08f80db04"
[targets] [targets]
test = ["Documenter", "LibSndFile", "Test"] test = ["Test", "TestSetExtensions"]

View file

@ -1,23 +1,17 @@
PortAudio.jl PortAudio.jl
============ ============
[![Dev](https://img.shields.io/badge/docs-dev-blue.svg)](https://JuliaAudio.github.io/PortAudio.jl/dev) [![Build Status](https://travis-ci.org/JuliaAudio/PortAudio.jl.svg?branch=master)](https://travis-ci.org/JuliaAudio/PortAudio.jl)
[![Tests](https://github.com/JuliaAudio/PortAudio.jl/actions/workflows/Tests.yml/badge.svg)](https://github.com/JuliaAudio/PortAudio.jl/actions/workflows/Tests.yml) [![Build status](https://ci.appveyor.com/api/projects/status/6x1ha7uvrnel060g/branch/master?svg=true)](https://ci.appveyor.com/project/ssfrr/portaudio-jl/branch/master)
[![codecov](https://codecov.io/gh/JuliaAudio/PortAudio.jl/branch/master/graph/badge.svg?token=mgDAi8ulPY)](https://codecov.io/gh/JuliaAudio/PortAudio.jl)
PortAudio.jl is a wrapper for [libportaudio](http://www.portaudio.com/), which gives cross-platform access to audio devices. It is compatible with the types defined in [SampledSignals.jl](https://github.com/JuliaAudio/SampledSignals.jl). It provides a `PortAudioStream` type, which can be read from and written to. PortAudio.jl is a wrapper for [libportaudio](http://www.portaudio.com/), which gives cross-platform access to audio devices. It is compatible with the types defined in [SampledSignals.jl](https://github.com/JuliaAudio/SampledSignals.jl). It provides a `PortAudioStream` type, which can be read from and written to.
## Opening a stream ## Opening a stream
The easiest way to open a source or sink is with the default `PortAudioStream()` constructor, The easiest way to open a source or sink is with the default `PortAudioStream()` constructor, which will open a 2-in, 2-out stream to your system's default device(s). The constructor can also take the input and output channel counts as positional arguments, or a variety of other keyword arguments.
which will open a 2-in, 2-out stream to your system's default device(s).
The constructor can also take the input and output channel counts as positional arguments,
or a variety of other keyword arguments.
If named keyword arguments `latency` or `samplerate` are unspecified, then PortAudio will use device defaults.
```julia ```julia
PortAudioStream(inchans=2, outchans=2; eltype=Float32, samplerate=48000, latency=0.1) PortAudioStream(inchans=2, outchans=2; eltype=Float32, samplerate=48000Hz, blocksize=4096, synced=false)
``` ```
You can open a specific device by adding it as the first argument, either as a `PortAudioDevice` instance or by name. You can also give separate names or devices if you want different input and output devices You can open a specific device by adding it as the first argument, either as a `PortAudioDevice` instance or by name. You can also give separate names or devices if you want different input and output devices
@ -31,34 +25,26 @@ You can get a list of your system's devices with the `PortAudio.devices()` funct
```julia ```julia
julia> PortAudio.devices() julia> PortAudio.devices()
14-element Vector{PortAudio.PortAudioDevice}: 6-element Array{PortAudio.PortAudioDevice,1}:
"sof-hda-dsp: - (hw:0,0)" 2→2 PortAudio.PortAudioDevice("AirPlay","Core Audio",0,2,0)
"sof-hda-dsp: - (hw:0,3)" 0→2 PortAudio.PortAudioDevice("Built-in Microph","Core Audio",2,0,1)
"sof-hda-dsp: - (hw:0,4)" 0→2 PortAudio.PortAudioDevice("Built-in Output","Core Audio",0,2,2)
"sof-hda-dsp: - (hw:0,5)" 0→2 PortAudio.PortAudioDevice("JackRouter","Core Audio",2,2,3)
PortAudio.PortAudioDevice("After Effects 13.5","Core Audio",0,0,4)
"upmix" 8→8 PortAudio.PortAudioDevice("Built-In Aggregate","Core Audio",2,2,5)
"vdownmix" 6→6
"dmix" 0→2
"default" 32→32
``` ```
### Input/Output Synchronization
The `synced` keyword argument to `PortAudioStream` controls whether the input and output ringbuffers are kept synchronized or not, which only effects duplex streams. It should be set to `true` if you need consistent input-to-output latency. In a synchronized stream, the underlying PortAudio callback will only read and write to the buffers an equal number of frames. In a synchronized stream, the user must also read and write an equal number of frames to the stream. If it is only written to or read from, it will eventually block. This is why it is `false` by default.
## Reading and Writing ## Reading and Writing
The `PortAudioStream` type has `source` and `sink` fields which are of type `PortAudioSource <: SampleSource` and `PortAudioSink <: SampleSink`, respectively. are subtypes of `SampleSource` and `SampleSink`, respectively (from [SampledSignals.jl](https://github.com/JuliaAudio/SampledSignals.jl)). This means they support all the stream and buffer features defined there. For example, if you load SampledSignals with `using SampledSignals` you can read 5 seconds to a buffer with `buf = read(stream.source, 5s)`, regardless of the sample rate of the device. The `PortAudioStream` type has `source` and `sink` fields which are of type `PortAudioSource <: SampleSource` and `PortAudioSink <: SampleSink`, respectively. are subtypes of `SampleSource` and `SampleSink`, respectively (from [SampledSignals.jl](https://github.com/JuliaAudio/SampledSignals.jl)). This means they support all the stream and buffer features defined there. For example, if you load SampledSignals with `using SampledSignals` you can read 5 seconds to a buffer with `buf = read(stream.source, 5s)`, regardless of the sample rate of the device.
PortAudio.jl also provides convenience wrappers around the `PortAudioStream` type so you can read and write to it directly, e.g. `write(stream, stream)` will set up a loopback that will read from the input and play it back on the output. PortAudio.jl also provides convenience wrappers around the `PortAudioStream` type so you can read and write to it directly, e.g. `write(stream, stream)` will set up a loopback that will read from the input and play it back on the output.
## Debugging
If you are experiencing issues and wish to view detailed logging and debug information, set
```
ENV["JULIA_DEBUG"] = :PortAudio
```
before using the package.
## Examples ## Examples
### Set up an audio pass-through from microphone to speaker ### Set up an audio pass-through from microphone to speaker
@ -82,7 +68,7 @@ end
### Open your built-in microphone and speaker by name ### Open your built-in microphone and speaker by name
```julia ```julia
PortAudioStream("default", "default") do stream PortAudioStream("Built-in Microph", "Built-in Output") do stream
write(stream, stream) write(stream, stream)
end end
``` ```
@ -90,18 +76,13 @@ end
### Record 10 seconds of audio and save to an ogg file ### Record 10 seconds of audio and save to an ogg file
```julia ```julia
julia> import LibSndFile # must be in Manifest for FileIO.save to work julia> using PortAudio, SampledSignals, LibSndFile
julia> using PortAudio: PortAudioStream julia> stream = PortAudioStream("Built-in Microph", 2, 0)
PortAudio.PortAudioStream{Float32,SIUnits.SIQuantity{Int64,0,0,-1,0,0,0,0,0,0}}
julia> using SampledSignals: s Samplerate: 48000 s⁻¹
Buffer Size: 4096 frames
julia> using FileIO: save 2 channel source: "Built-in Microph"
julia> stream = PortAudioStream(1, 0) # default input (e.g., built-in microphone)
PortAudioStream{Float32}
Samplerate: 44100.0Hz
2 channel source: "default"
julia> buf = read(stream, 10s) julia> buf = read(stream, 10s)
480000-frame, 2-channel SampleBuf{Float32, 2, SIUnits.SIQuantity{Int64,0,0,-1,0,0,0,0,0,0}} 480000-frame, 2-channel SampleBuf{Float32, 2, SIUnits.SIQuantity{Int64,0,0,-1,0,0,0,0,0,0}}
@ -114,13 +95,13 @@ julia> close(stream)
julia> save(joinpath(homedir(), "Desktop", "myvoice.ogg"), buf) julia> save(joinpath(homedir(), "Desktop", "myvoice.ogg"), buf)
``` ```
### Play an audio signal through the default sound output device ## Building the shim library
```julia Because PortAudio calls its callback from a separate audio thread, we can't handle it in Julia directly. To work around this we've included a small shim library written in C that uses ring buffers to pass audio data between the callback context and the main Julia context. To build the shim you'll need a few prerequisites:
using PortAudio, SampledSignals
S = 8192 # sampling rate (samples / second) * libportaudio
x = cos.(2pi*(1:2S)*440/S) # A440 tone for 2 seconds * make
PortAudioStream(0, 2; samplerate=S) do stream * a C compiler (gcc on linux/macOS, mingw64 on Windows)
write(stream, x) * The `RingBuffers` julia package, installed in a folder next to this one. The portaudio shim links against the `pa_ringbuffer` library that comes with `RingBuffers`.
end
``` To build the shim, go into the `deps/src` directory and type `make`.

View file

@ -1,2 +0,0 @@
[deps]
Documenter = "e30172f5-a6a5-5a46-863b-614d45cd2de4"

View file

@ -1,12 +0,0 @@
using PortAudio
using Documenter: deploydocs, makedocs
makedocs(
sitename = "PortAudio.jl",
modules = [PortAudio],
pages = [
"Public interface" => "index.md",
"Internals" => "internals.md"
]
)
deploydocs(repo = "github.com/JuliaAudio/PortAudio.jl.git")

View file

@ -1,10 +0,0 @@
# Public interface
```@index
Pages = ["index.md"]
```
```@autodocs
Modules = [PortAudio]
Private = false
```

View file

@ -1,10 +0,0 @@
# Internals
```@index
Pages = ["internals.md"]
```
```@autodocs
Modules = [PortAudio]
Public = false
```

View file

@ -1,11 +1,9 @@
using PortAudio using PortAudio
""" """Continuously read from the default audio input and plot an
Continuously read from the default audio input and plot an ASCII level/peak meter"""
ASCII level/peak meter
"""
function micmeter(metersize) function micmeter(metersize)
mic = PortAudioStream(1, 0; latency = 0.1) mic = PortAudioStream(1, 0; blocksize=512)
signalmax = zero(eltype(mic)) signalmax = zero(eltype(mic))
println("Press Ctrl-C to quit") println("Press Ctrl-C to quit")
@ -18,32 +16,28 @@ function micmeter(metersize)
end end
end end
""" """Print an ASCII level meter of the given size. Signal and peak
Print an ASCII level meter of the given size. Signal and peak levels are assumed to be scaled from 0.0-1.0, with peak >= signal"""
levels are assumed to be scaled from 0.0-1.0, with peak >= signal
"""
function printmeter(metersize, signal, peak) function printmeter(metersize, signal, peak)
# calculate the positions in terms of characters # calculate the positions in terms of characters
peakpos = clamp(round(Int, peak * metersize), 0, metersize) peakpos = clamp(round(Int, peak * metersize), 0, metersize)
meterchars = clamp(round(Int, signal * metersize), 0, peakpos - 1) meterchars = clamp(round(Int, signal * metersize), 0, peakpos-1)
blankchars = max(0, peakpos - meterchars - 1) blankchars = max(0, peakpos-meterchars-1)
for position in 1:meterchars for position in 1:meterchars
printstyled(">", color = barcolor(metersize, position)) printstyled(">", color=barcolor(metersize, position))
end end
print(" "^blankchars) print(" " ^ blankchars)
printstyled("|", color = barcolor(metersize, peakpos)) printstyled("|", color=barcolor(metersize, peakpos))
print(" "^(metersize - peakpos)) print(" " ^ (metersize - peakpos))
end end
""" """Compute the proper color for a given position in the bar graph. The first
Compute the proper color for a given position in the bar graph. The first
half of the bar should be green, then the remainder is yellow except the final half of the bar should be green, then the remainder is yellow except the final
character, which is red. character, which is red."""
"""
function barcolor(metersize, position) function barcolor(metersize, position)
if position / metersize <= 0.5 if position/metersize <= 0.5
:green :green
elseif position == metersize elseif position == metersize
:red :red

View file

@ -1,156 +1,127 @@
using Distributed, PortAudio # Thanks to Jiahao Chen for this great example!
# Modified from Jiahao Chen's example in the obsolete AudioIO module. ##
# Will use first output device found in system's listing or DEFAULTDEVICE if set below ## NOTE: THIS NEEDS TO BE PORTED OVER TO THE NEW ARCHITECTURE
const DEFAULTDEVICE = -1 ##
function paudio() using AudioIO
devs = PortAudio.devices() import AudioIO.play
if DEFAULTDEVICE < 0
devnum = findfirst(x -> x.maxoutchans > 0, devs)
(devnum == nothing) && error("No output device for audio found")
else
devnum = DEFAULTDEVICE + 1
end
return ostream = PortAudioStream(devs[devnum].name, 0, 2)
end
play(ostream, sample::Array{Float64, 1}) = write(ostream, sample) type note{S<:Real, T<:Real}
play(ostr, sample::Array{Int64, 1}) = play(ostr, Float64.(sample))
struct Note{S <: Real, T <: Real}
pitch::S pitch::S
duration::T duration::T
sustained::Bool sustained::Bool
end end
function play( function play(A::note, samplingfreq::Real=44100, shape::Function=t->0.6sin(t)+0.2sin(2t)+.05*sin(8t))
ostream, timesamples=0:1/samplingfreq:(A.duration*(A.sustained ? 0.98 : 0.9))
A::Note, v = Float64[shape(2π*A.pitch*t) for t in timesamples]
samplingfreq::Real = 44100,
shape::Function = t -> 0.6sin(t) + 0.2sin(2t) + 0.05 * sin(8t),
)
timesamples = 0:(1 / samplingfreq):(A.duration * (A.sustained ? 0.98 : 0.9))
v = Float64[shape(2π * A.pitch * t) for t in timesamples]
if !A.sustained if !A.sustained
decay_length = div(length(timesamples), 5) decay_length = int(length(timesamples) * 0.2)
v[(end - decay_length):(end - 1)] = v[end-decay_length:end-1] = v[end-decay_length:end-1] .* linspace(1, 0, decay_length)
v[(end - decay_length):(end - 1)] .* LinRange(1, 0, decay_length)
end end
play(ostream, v) play(v)
sleep(A.duration) sleep(A.duration)
end end
function parsevoice(melody::String; tempo = 132, beatunit = 4, lyrics = nothing) function parsevoice(melody::String; tempo=132, beatunit=4, lyrics=nothing)
ostream = paudio() # initialize audio for output play([0]) #Force AudioIO to initialize
lyrics_syllables = lyrics == nothing ? nothing : split(lyrics) lyrics_syllables = lyrics==nothing? nothing : split(lyrics)
lyrics_syllables != nothing && (lyrics_syllables[end] *= "\n")
note_idx = 1 note_idx = 1
oldduration = 4 oldduration = 4
for line in split(melody, '\n') for line in split(melody, '\n')
percent_idx = findfirst('%', line) # Trim comment percent_idx = findfirst(line, '%') #Trim comment
percent_idx == nothing || (line = line[1:(percent_idx - 1)]) percent_idx == 0 || (line = line[1:percent_idx-1])
for token in split(line) for token in split(line)
pitch, duration, dotted, sustained = parsetoken(token) pitch, duration, dotted, sustained =parsetoken(token)
duration == nothing && (duration = oldduration) duration==nothing && (duration = oldduration)
oldduration = duration oldduration = duration
dotted && (duration *= 1.5) dotted && (duration *= 1.5)
if lyrics_syllables != nothing && 1 <= note_idx <= length(lyrics_syllables) if lyrics_syllables!=nothing && 1<=note_idx<=length(lyrics_syllables) #Print the lyrics, omitting hyphens
# Print the lyrics, omitting hyphens if lyrics_syllables[note_idx][end]=='-'
if lyrics_syllables[note_idx][end] == '-' print(lyrics_syllables[note_idx][1:end-1])
print(join(split(lyrics_syllables[note_idx][:], "")[1:(end - 1)]), "")
else else
print(lyrics_syllables[note_idx], ' ') print(lyrics_syllables[note_idx], ' ')
end end
end end
play(ostream, Note(pitch, (beatunit / duration) * (60 / tempo), sustained)) play(note(pitch, (beatunit/duration)*(60/tempo), sustained))
note_idx += 1 note_idx += 1
end end
println()
end end
end end
function parsetoken(token, Atuning::Real = 220) function parsetoken(token::String, Atuning::Real=220)
state = :findpitch state = :findpitch
pitch = 0.0 pitch = 0.0
sustain = dotted = false sustain = dotted = false
lengthbuf = Char[] lengthbuf = Char[]
for char in token for char in token
if state == :findpitch if state == :findpitch
scale_idx = scale_idx = findfirst('a':'g', char) + findfirst('A':'G', char)
something(findfirst(char, String(collect('a':'g'))), 0) + if scale_idx!=0
something(findfirst(char, String(collect('A':'G'))), 0) const halfsteps = [12, 14, 3, 5, 7, 8, 10]
if scale_idx != 0 pitch = Atuning*2^(halfsteps[scale_idx]/12)
halfsteps = [12, 14, 3, 5, 7, 8, 10]
pitch = Atuning * 2^(halfsteps[scale_idx] / 12)
state = :findlength state = :findlength
elseif char == 'r' elseif char=='r'
pitch, state = 0, :findlength pitch, state = 0, :findlength
else else
error("unknown pitch: $char") error("unknown pitch: $char")
end end
elseif state == :findlength elseif state == :findlength
if char == '#' if char == '#' ; pitch *= 2^(1/12) #sharp
pitch *= 2^(1 / 12) # sharp elseif char == 'b' ; pitch /= 2^(1/12) #flat
elseif char == 'b' elseif char == '\''; pitch *= 2 #higher octave
pitch /= 2^(1 / 12) # flat elseif char == ',' ; pitch /= 2 #lower octave
elseif char == '\'' elseif char == '.' ; dotted = true #dotted note
pitch *= 2 # higher octave elseif char == '~' ; sustain = true #tied note
elseif char == ','
pitch /= 2 # lower octave
elseif char == '.'
dotted = true # dotted note
elseif char == '~'
sustain = true # tied note
else else
push!(lengthbuf, char) push!(lengthbuf, char)
# Check for "is" and "es" suffixes for sharps and flats #Check for "is" and "es" suffixes for sharps and flats
if length(lengthbuf) >= 2 if length(lengthbuf) >= 2
if lengthbuf[(end - 1):end] == "is" if lengthbuf[end-1:end] == "is"
pitch *= 2^(1 / 12) pitch *= 2^(1/12)
lengthbuf = lengthbuf[1:(end - 2)] lengthbuf = lengthbuf[1:end-2]
elseif lengthbuf[(end - 1):end] == "es" elseif lengthbuf[end-1:end] == "es"
pitch /= 2^(1 / 12) pitch /= 2^(1/12)
lengthbuf = lengthbuf[1:(end - 2)] lengthbuf = lengthbuf[1:end-2]
end end
end end
end end
end end
end end
#finalize length #finalize length
lengthstr = String(lengthbuf) lengthstr = convert(String, lengthbuf)
duration = isempty(lengthstr) ? nothing : tryparse(Int, lengthstr) duration = isempty(lengthstr) ? nothing : parseint(lengthstr)
return (pitch, duration, sustain, dotted) return (pitch, duration, sustain, dotted)
end end
parsevoice( parsevoice("""
"""
f# f# g a a g f# e d d e f# f#~ f#8 e e2 f# f# g a a g f# e d d e f# f#~ f#8 e e2
f#4 f# g a a g f# e d d e f# e~ e8 d d2 f#4 f# g a a g f# e d d e f# e~ e8 d d2
e4 e f# d e f#8~ g8 f#4 d e f#8~ g f#4 e d e a, e4 e f# d e f#8~ g8 f#4 d e f#8~ g f#4 e d e a,
f#2 f#4 g a a g f# e d d e f# e~ e8 d8 d2""", f#2 f#4 g a a g f# e d d e f# e~ e8 d8 d2""",
lyrics = """ lyrics="""
Freu- de, schö- ner Göt- ter- fun- ken, Toch- ter aus E- li- - si- um! Freu- de, schö- ner Göt- ter- fun- ken, Toch- ter aus E- li- - si- um!
Wir be- tre- ten feu- er- trun- ken, Himm- li- sche, dein Hei- - lig- thum! Wir be- tre- ten feu- er- trun- ken, Himm- li- sche, dein Hei- - lig- thum!
Dei- ne Zau- ber bin den - wie- der, was die - Mo- de streng ge- theilt, Dei- ne Zau- ber bin den - wie- der, was die - Mo- de streng ge- theilt,
al- le mensch- en wer- den Brü- der wo dein sanf- ter Flü- - gel weilt. al- le mensch- en wer- den Brü- der wo dein sanf- ter Flü- - gel weilt.
""", """)
)
# And now with harmony! # And now with harmony!
soprano = @spawn parsevoice( soprano = @async parsevoice("""
"""
f'#. f'#. g'. a'. a'. g'. f'#. e'~ e'8 d.'4 d.' e.' f#'. f#'.~ f#' e'8 e'4~ e'2 f'#. f'#. g'. a'. a'. g'. f'#. e'~ e'8 d.'4 d.' e.' f#'. f#'.~ f#' e'8 e'4~ e'2
""", """, lyrics="Freu- de, schö- ner Göt- ter- fun- ken, Toch- ter aus E- li- - si- um!"
lyrics = "Freu- de, schö- ner Göt- ter- fun- ken, Toch- ter aus E- li- - si- um!",
) )
alto = @spawn parsevoice(""" alto = @async parsevoice("""
a. a. a. a. a. a. a. a~ g8 f#.4 a. a. a. a.~ a a8 a4~ a2 a. a. a. a. a. a. a. a~ g8 f#.4 a. a. a. a.~ a a8 a4~ a2
""") """)
tenor = @spawn parsevoice(""" tenor = @async parsevoice("""
d. d. e. f#. f#. e. d. d~ e8 f#.4 f#. a,. d. d.~ d c#8 c#4 c#2 d. d. e. f#. f#. e. d. d~ e8 f#.4 f#. a,. d. d.~ d c#8 c#4 c#2
""") """)
bass = @spawn parsevoice(""" bass = @async parsevoice("""
d. d. d. d. a,. a,. a,. b,~ c8 d. a., a., a., a., a, a8, a,4 a,2 d. d. d. d. a,. a,. a,. b,~ c8 d. a., a., a., a., a, a8, a,4 a,2
""") """)
wait(soprano) wait(soprano)
@ -158,21 +129,19 @@ wait(alto)
wait(tenor) wait(tenor)
wait(bass) wait(bass)
soprano = @spawn parsevoice( soprano = @async parsevoice("""
"""
f'#.4 f'#. g'. a'. a'. g'. f'#. e'. d'. d'. e'. f'#. e'.~ e' d'8 d'4~ d'2 f'#.4 f'#. g'. a'. a'. g'. f'#. e'. d'. d'. e'. f'#. e'.~ e' d'8 d'4~ d'2
""", """, lyrics="Wir be- tre- ten feu- er- trun- ken, Himm- li- sche, dein Hei- - lig- thum!")
lyrics = "Wir be- tre- ten feu- er- trun- ken, Himm- li- sche, dein Hei- - lig- thum!", alto = @async parsevoice("""
)
alto = @spawn parsevoice("""
a.4 a. b. c'. c'. b. a. g. f#. f#. g. f#. g.~ g4 f#8 f#~ f#2 a.4 a. b. c'. c'. b. a. g. f#. f#. g. f#. g.~ g4 f#8 f#~ f#2
""") """)
tenor = @spawn parsevoice(""" tenor = @async parsevoice("""
d.4 d. d. d. d. d. d. d. d. d. c#. d. c#.~ c# d8 d d2 d.4 d. d. d. d. d. d. d. d. d. c#. d. c#.~ c# d8 d d2
""") """)
bass = @spawn parsevoice(""" bass = @async parsevoice("""
d.4 d. d. d. a,. a,. a,. a., a., a., a., a., a.,~ a, a,8 d, d,2 d.4 d. d. d. a,. a,. a,. a., a., a., a., a., a.,~ a, a,8 d, d,2
""") """)
wait(soprano) wait(soprano)
wait(alto) wait(alto)
wait(tenor) wait(tenor)

View file

@ -1,65 +0,0 @@
using PortAudio
using DSP
function create_measure_signal()
signal = zeros(Float32, 20000)
for i in 1:3
signal = vcat(signal, rand(Float32, 100), zeros(Float32, i * 10000))
end
return signal
end
function measure_latency(in_latency = 0.1, out_latency = 0.1; is_warmup = false)
in_stream = PortAudioStream(1, 0; latency = in_latency)
out_stream = PortAudioStream(0, 1; latency = out_latency)
cond = Base.Event()
writer_start_time = Int64(0)
reader_start_time = Int64(0)
reader = Threads.@spawn begin
wait(cond)
writer_start_time = time_ns() |> Int64
return read(in_stream, 100000)
end
signal = create_measure_signal()
writer = Threads.@spawn begin
wait(cond)
reader_start_time = time_ns() |> Int64
write(out_stream, signal)
end
notify(cond)
wait(reader)
wait(writer)
recorded = collect(reader.result)[:, 1]
close(in_stream)
close(out_stream)
diff = reader_start_time - writer_start_time |> abs
diff_in_ms = diff / 10^6 # 1 ms = 10^6 ns
if !is_warmup && diff_in_ms > 1
@warn "Threads start time difference $diff_in_ms ms is bigger than 1 ms"
end
delay = finddelay(recorded, signal) / 48000
return trunc(Int, delay * 1000)# result in ms
end
measure_latency(0.1, 0.1; is_warmup = true) # warmup
latencies = [0.1, 0.01, 0.005]
for in_latency in latencies
for out_latency in latencies
measure = measure_latency(in_latency, out_latency)
println("$measure ms latency for in_latency=$in_latency, out_latency=$out_latency")
end
end

View file

@ -1,89 +0,0 @@
#=
This code illustrates real-time octave down shift
using a crude FFT-based method.
It also plots the input and output signals and their spectra.
This code uses the system defaults for the audio input and output devices.
If you use the built-in speakers and built-in microphone,
you will likely get undesirable audio feedback.
It works "best" if you play the audio output through headphones
so that the output does not feed back into the input.
The spectrum plotting came from the example in
https://github.com/JuliaAudio/PortAudio.jl/blob/master/examples
=#
using PortAudio: PortAudioStream
using SampledSignals: Hz, domain
using SampledSignals: (..) # see EllipsisNotation.jl and IntervalSets.jl
using FFTW: fft, ifft
using Plots: plot, gui, default; default(label="")
function pitch_halver(x) # decrease pitch by one octave via FFT
N = length(x)
mod(N,2) == 0 || throw("N must be multiple of 2")
F = fft(x) # original spectrum
Fnew = [F[1:N÷2]; zeros(N+1); F[(N÷2+2):N]]
out = 2 * real(ifft(Fnew))[1:N]
out.samplerate /= 2 # trick!
return out
end
# Plot input and output signals and their spectra.
# Quantize the vertical axis limits to reduce plot jitter.
function plotter(buf, out, N, fmin, fmax, fs; quant::Number = 0.1)
bmax = quant * ceil(maximum(abs, buf) / quant)
xticks = [1, N]; ylims = (-1,1) .* bmax; yticks = (-1:1)*bmax
p1 = plot(buf; xticks, ylims, yticks, title="input")
p3 = plot(out; xticks, ylims, yticks, title="output")
X = (2/N) * abs.(fft(buf)[fmin..fmax]) # spectrum
Xmax = quant * ceil(maximum(X) / quant)
xlims = (fs[1], fs[end]); ylims = (0, Xmax); yticks = [0,Xmax]
p2 = plot(fs, X; xlims, ylims, yticks)
Y = (2/N) * abs.(fft(out)[fmin..fmax])
p4 = plot(fs, Y; xlims, ylims, yticks)
plot(p1, p2, p3, p4)
end
"""
octave_shift(seconds; N, ...)
Shift audio down by one octave.
# Input
* `seconds` : how long to run in seconds; defaults to 300 (5 minutes)
# Options
* `N` : buffer size; default 1024 samples
* `fmin`,`fmax` : range of frequencies to display; default 0Hz to 4000Hz
"""
function octave_shift(
seconds::Number = 300;
N::Int = 1024,
fmin::Number = 0Hz,
fmax::Number = 4000Hz,
# undocumented options below here that are unlikely to be modified
in_stream = PortAudioStream(1, 0), # default input device
out_stream = PortAudioStream(0, 1), # default output device
buf::AbstractArray = read(in_stream, N), # warm-up
fs = Float32[float(f) for f in domain(fft(buf)[fmin..fmax])],
Niters::Int = ceil(Int, seconds * in_stream.sample_rate / N),
)
for _ in 1:Niters
read!(in_stream, buf)
out = pitch_halver(buf) # decrease pitch by one octave
write(out_stream, out)
plotter(buf, out, N, fmin, fmax, fs); gui()
end
nothing
end
octave_shift(5)

View file

@ -6,7 +6,7 @@ module SpectrumExample
using GR, PortAudio, SampledSignals, FFTW using GR, PortAudio, SampledSignals, FFTW
const N = 1024 const N = 1024
const stream = PortAudioStream(1, 0) const stream = PortAudioStream(1, 0, blocksize=N)
const buf = read(stream, N) const buf = read(stream, N)
const fmin = 0Hz const fmin = 0Hz
const fmax = 10000Hz const fmax = 10000Hz
@ -14,7 +14,7 @@ const fs = Float32[float(f) for f in domain(fft(buf)[fmin..fmax])]
while true while true
read!(stream, buf) read!(stream, buf)
plot(fs, abs.(fft(buf)[fmin..fmax]), xlim = (fs[1], fs[end]), ylim = (0, 100)) plot(fs, abs.(fft(buf)[fmin..fmax]), xlim=(fs[1],fs[end]), ylim=(0,100))
end end
end end

View file

@ -1,21 +0,0 @@
#=
This example illustrates synthesizing a long tone in small pieces
and routing it to the default audio output device using `write()`.
=#
using PortAudio: PortAudioStream, write
stream = PortAudioStream(0, 1; warn_xruns=false)
function play_tone(stream, freq::Real, duration::Real; buf_size::Int = 1024)
S = stream.sample_rate
current = 1
while current < duration*S
x = 0.7 * sin.(2π * (current .+ (1:buf_size)) * freq / S)
write(stream, x)
current += buf_size
end
nothing
end
play_tone(stream, 440, 2)

View file

@ -8,8 +8,8 @@ The rightmosts column is discarded and the leftmost column is
left alone. left alone.
""" """
function shift1!(buf::AbstractMatrix) function shift1!(buf::AbstractMatrix)
for col in size(buf, 2):-1:2 for col in size(buf,2):-1:2
@. buf[:, col] = buf[:, col - 1] @. buf[:, col] = buf[:, col-1]
end end
end end
@ -20,7 +20,7 @@ function processbuf!(readbuf, win, dispbuf, fftbuf, fftplan)
readbuf .*= win readbuf .*= win
A_mul_B!(fftbuf, fftplan, readbuf) A_mul_B!(fftbuf, fftplan, readbuf)
shift1!(dispbuf) shift1!(dispbuf)
@. dispbuf[end:-1:1, 1] = log(clamp(abs(fftbuf[1:D]), 0.0001, Inf)) @. dispbuf[end:-1:1,1] = log(clamp(abs(fftbuf[1:D]), 0.0001, Inf))
end end
function processblock!(src, buf, win, dispbufs, fftbuf, fftplan) function processblock!(src, buf, win, dispbufs, fftbuf, fftplan)
@ -31,17 +31,17 @@ function processblock!(src, buf, win, dispbufs, fftbuf, fftplan)
end end
N = 1024 # size of audio read N = 1024 # size of audio read
N2 = N ÷ 2 + 1 # size of rfft output N2 = N÷2+1 # size of rfft output
D = 200 # number of bins to display D = 200 # number of bins to display
M = 200 # amount of history to keep M = 200 # amount of history to keep
src = PortAudioStream(1, 2) src = PortAudioStream(1, 2, blocksize=N)
buf = Array{Float32}(N) # buffer for reading buf = Array{Float32}(N) # buffer for reading
fftplan = plan_rfft(buf; flags = FFTW.EXHAUSTIVE) fftplan = plan_rfft(buf; flags=FFTW.EXHAUSTIVE)
fftbuf = Array{Complex{Float32}}(N2) # destination buf for FFT fftbuf = Array{Complex{Float32}}(N2) # destination buf for FFT
dispbufs = [zeros(Float32, D, M) for i in 1:5, j in 1:5] # STFT bufs dispbufs = [zeros(Float32, D, M) for i in 1:5, j in 1:5] # STFT bufs
win = gaussian(N, 0.125) win = gaussian(N, 0.125)
scene = Scene(resolution = (1000, 1000)) scene = Scene(resolution=(1000,1000))
#pre-fill the display buffer so we can do a reasonable colormap #pre-fill the display buffer so we can do a reasonable colormap
for _ in 1:M for _ in 1:M
@ -53,7 +53,7 @@ heatmaps = map(enumerate(IndexCartesian(), dispbufs)) do ibuf
buf = ibuf[2] buf = ibuf[2]
# some function of the 2D index and the value # some function of the 2D index and the value
heatmap(buf, offset = (i[2] * size(buf, 2), i[1] * size(buf, 1))) heatmap(buf, offset=(i[2]*size(buf, 2), i[1]*size(buf, 1)))
end end
center!(scene) center!(scene)

View file

@ -2,32 +2,27 @@ using Makie, GeometryTypes
using PortAudio using PortAudio
N = 1024 # size of audio read N = 1024 # size of audio read
N2 = N ÷ 2 + 1 # size of rfft output N2 = N÷2+1 # size of rfft output
D = 200 # number of bins to display D = 200 # number of bins to display
M = 100 # number of lines to draw M = 100 # number of lines to draw
S = 0.5 # motion speed of lines S = 0.5 # motion speed of lines
src = PortAudioStream(1, 2) src = PortAudioStream(1, 2, blocksize=N)
buf = Array{Float32}(N) buf = Array{Float32}(N)
fftbuf = Array{Complex{Float32}}(N2) fftbuf = Array{Complex{Float32}}(N2)
magbuf = Array{Float32}(N2) magbuf = Array{Float32}(N2)
fftplan = plan_rfft(buf; flags = FFTW.EXHAUSTIVE) fftplan = plan_rfft(buf; flags=FFTW.EXHAUSTIVE)
scene = Scene(resolution = (500, 500)) scene = Scene(resolution=(500,500))
ax = axis(0:0.1:1, 0:0.1:1, 0:0.1:0.5) ax = axis(0:0.1:1, 0:0.1:1, 0:0.1:0.5)
center!(scene) center!(scene)
ls = map(1:M) do _ ls = map(1:M) do _
yoffset = to_node(to_value(scene[:time])) yoffset = to_node(to_value(scene[:time]))
offset = lift_node(scene[:time], yoffset) do t, yoff offset = lift_node(scene[:time], yoffset) do t, yoff
Point3f0(0.0f0, (t - yoff) * S, 0.0f0) Point3f0(0.0f0, (t-yoff)*S, 0.0f0)
end end
l = lines( l = lines(linspace(0,1,D), 0.0f0, zeros(Float32, D),
linspace(0, 1, D), offset=offset, color=(:black, 0.1))
0.0f0,
zeros(Float32, D),
offset = offset,
color = (:black, 0.1),
)
(yoffset, l) (yoffset, l)
end end
@ -36,7 +31,7 @@ while isopen(scene[:screen])
isopen(scene[:screen]) || break isopen(scene[:screen]) || break
read!(src, buf) read!(src, buf)
A_mul_B!(fftbuf, fftplan, buf) A_mul_B!(fftbuf, fftplan, buf)
@. magbuf = log(clamp(abs(fftbuf), 0.0001, Inf)) / 10 + 0.5 @. magbuf = log(clamp(abs(fftbuf), 0.0001, Inf))/10+0.5
line[:z] = magbuf[1:D] line[:z] = magbuf[1:D]
push!(yoffset, to_value(scene[:time])) push!(yoffset, to_value(scene[:time]))
end end

View file

@ -1 +0,0 @@
The clang generators will automatically generate wrappers for a C library based on its headers. So everything you see in libportaudio.jl is automatically generated from the C library. If a newer version of portaudio adds more features, we won't have to add new wrappers: clang will handle it for us. It is easy to use currently unused features: the wrappers have already been written for us. Even though it does an admirable job, clang doesn't handle errors and set locks. Fortunately, it's very easy to add secondary wrappers, or just do it at point of use.

View file

@ -1,16 +0,0 @@
using Clang.Generators
using libportaudio_jll
cd(@__DIR__)
include_dir = joinpath(libportaudio_jll.artifact_dir, "include") |> normpath
portaudio_h = joinpath(include_dir, "portaudio.h")
options = load_options(joinpath(@__DIR__, "generator.toml"))
args = get_default_args()
push!(args, "-I$include_dir")
ctx = create_context(portaudio_h, args, options)
build!(ctx)

View file

@ -1,9 +0,0 @@
[general]
library_name = "libportaudio"
output_file_path = "../src/LibPortAudio.jl"
module_name = "LibPortAudio"
jll_pkg_name = "libportaudio_jll"
export_symbol_prefixes = ["Pa", "pa"]
use_julia_native_enum_type = true
auto_mutability = true

11
runtests.sh Executable file
View file

@ -0,0 +1,11 @@
#!/bin/bash
# Runs the tests including generating an lcov.info file
# abort on failure
set -e
julia -e 'using Coverage; clean_folder(".");'
julia --color=yes --inline=no --code-coverage=user test/runtests.jl
mkdir -p coverage
julia -e 'using Coverage; res=process_folder(); LCOV.writefile("coverage/lcov.info", res)'

File diff suppressed because it is too large Load diff

View file

@ -1,328 +1,158 @@
module LibPortAudio # Low-level wrappers for Portaudio calls
using libportaudio_jll
export libportaudio_jll
function Pa_GetVersion()
ccall((:Pa_GetVersion, libportaudio), Cint, ())
end
function Pa_GetVersionText()
ccall((:Pa_GetVersionText, libportaudio), Ptr{Cchar}, ())
end
mutable struct PaVersionInfo
versionMajor::Cint
versionMinor::Cint
versionSubMinor::Cint
versionControlRevision::Ptr{Cchar}
versionText::Ptr{Cchar}
end
# no prototype is found for this function at portaudio.h:114:22, please use with caution
function Pa_GetVersionInfo()
ccall((:Pa_GetVersionInfo, libportaudio), Ptr{PaVersionInfo}, ())
end
# General type aliases
const PaTime = Cdouble
const PaError = Cint const PaError = Cint
const PaSampleFormat = Culong
const PaDeviceIndex = Cint
const PaHostApiIndex = Cint
const PaHostApiTypeId = Cint
# PaStream is always used as an opaque type, so we're always dealing
# with the pointer
const PaStream = Ptr{Cvoid}
const PaStreamCallback = Cvoid
const PaStreamFlags = Culong
@enum PaErrorCode::Int32 begin const paNoFlag = PaStreamFlags(0x00)
paNoError = 0
paNotInitialized = -10000 const PA_NO_ERROR = 0
paUnanticipatedHostError = -9999 const PA_INPUT_OVERFLOWED = -10000 + 19
paInvalidChannelCount = -9998 const PA_OUTPUT_UNDERFLOWED = -10000 + 20
paInvalidSampleRate = -9997
paInvalidDevice = -9996 # sample format types
paInvalidFlag = -9995 const paFloat32 = PaSampleFormat(0x01)
paSampleFormatNotSupported = -9994 const paInt32 = PaSampleFormat(0x02)
paBadIODeviceCombination = -9993 const paInt24 = PaSampleFormat(0x04)
paInsufficientMemory = -9992 const paInt16 = PaSampleFormat(0x08)
paBufferTooBig = -9991 const paInt8 = PaSampleFormat(0x10)
paBufferTooSmall = -9990 const paUInt8 = PaSampleFormat(0x20)
paNullCallback = -9989 const paNonInterleaved = PaSampleFormat(0x80000000)
paBadStreamPtr = -9988
paTimedOut = -9987 const type_to_fmt = Dict{Type, PaSampleFormat}(
paInternalError = -9986 Float32 => 1,
paDeviceUnavailable = -9985 Int32 => 2,
paIncompatibleHostApiSpecificStreamInfo = -9984 # Int24 => 4,
paStreamIsStopped = -9983 Int16 => 8,
paStreamIsNotStopped = -9982 Int8 => 16,
paInputOverflowed = -9981 UInt8 => 3
paOutputUnderflowed = -9980 )
paHostApiNotFound = -9979
paInvalidHostApi = -9978 const PaStreamCallbackResult = Cint
paCanNotReadFromACallbackStream = -9977 # Callback return values
paCanNotWriteToACallbackStream = -9976 const paContinue = PaStreamCallbackResult(0)
paCanNotReadFromAnOutputOnlyStream = -9975 const paComplete = PaStreamCallbackResult(1)
paCanNotWriteToAnInputOnlyStream = -9974 const paAbort = PaStreamCallbackResult(2)
paIncompatibleStreamHostApi = -9973
paBadBufferPtr = -9972 """
Call the given expression in a separate thread, waiting on the result. This is
useful when running code that would otherwise block the Julia process (like a
`ccall` into a function that does IO).
"""
macro tcall(ex)
:(fetch(Base.Threads.@spawn $(esc(ex))))
end end
function Pa_GetErrorText(errorCode) # because we're calling Pa_ReadStream and PA_WriteStream from separate threads,
ccall((:Pa_GetErrorText, libportaudio), Ptr{Cchar}, (PaError,), errorCode) # we put a mutex around libportaudio calls
const pamutex = ReentrantLock()
macro locked(ex)
quote
lock(pamutex) do
$(esc(ex))
end
end
end end
function Pa_Initialize() function Pa_Initialize()
ccall((:Pa_Initialize, libportaudio), PaError, ()) err = @locked ccall((:Pa_Initialize, libportaudio), PaError, ())
handle_status(err)
end end
function Pa_Terminate() function Pa_Terminate()
ccall((:Pa_Terminate, libportaudio), PaError, ()) err = @locked ccall((:Pa_Terminate, libportaudio), PaError, ())
handle_status(err)
end end
const PaDeviceIndex = Cint Pa_GetVersion() = @locked ccall((:Pa_GetVersion, libportaudio), Cint, ())
const PaHostApiIndex = Cint function Pa_GetVersionText()
versionPtr = @locked ccall((:Pa_GetVersionText, libportaudio), Ptr{Cchar}, ())
function Pa_GetHostApiCount() unsafe_string(versionPtr)
ccall((:Pa_GetHostApiCount, libportaudio), PaHostApiIndex, ())
end end
function Pa_GetDefaultHostApi() # Host API Functions
ccall((:Pa_GetDefaultHostApi, libportaudio), PaHostApiIndex, ())
end
@enum PaHostApiTypeId::UInt32 begin # A Host API is the top-level of the PortAudio hierarchy. Each host API has a
paInDevelopment = 0 # unique type ID that tells you which native backend it is (JACK, ALSA, ASIO,
paDirectSound = 1 # etc.). On a given system you can identify each backend by its index, which
paMME = 2 # will range between 0 and Pa_GetHostApiCount() - 1. You can enumerate through
paASIO = 3 # all the host APIs on the system by iterating through those values.
paSoundManager = 4
paCoreAudio = 5 # PaHostApiTypeId values
paOSS = 7 const pa_host_api_names = Dict{PaHostApiTypeId, String}(
paALSA = 8 0 => "In Development", # use while developing support for a new host API
paAL = 9 1 => "Direct Sound",
paBeOS = 10 2 => "MME",
paWDMKS = 11 3 => "ASIO",
paJACK = 12 4 => "Sound Manager",
paWASAPI = 13 5 => "Core Audio",
paAudioScienceHPI = 14 7 => "OSS",
end 8 => "ALSA",
9 => "AL",
10 => "BeOS",
11 => "WDMKS",
12 => "Jack",
13 => "WASAPI",
14 => "AudioScience HPI"
)
mutable struct PaHostApiInfo mutable struct PaHostApiInfo
structVersion::Cint struct_version::Cint
type::PaHostApiTypeId api_type::PaHostApiTypeId
name::Ptr{Cchar} name::Ptr{Cchar}
deviceCount::Cint deviceCount::Cint
defaultInputDevice::PaDeviceIndex defaultInputDevice::PaDeviceIndex
defaultOutputDevice::PaDeviceIndex defaultOutputDevice::PaDeviceIndex
end end
function Pa_GetHostApiInfo(hostApi) Pa_GetHostApiInfo(i) = unsafe_load(@locked ccall((:Pa_GetHostApiInfo, libportaudio),
ccall( Ptr{PaHostApiInfo}, (PaHostApiIndex,), i))
(:Pa_GetHostApiInfo, libportaudio),
Ptr{PaHostApiInfo},
(PaHostApiIndex,),
hostApi,
)
end
function Pa_HostApiTypeIdToHostApiIndex(type) # Device Functions
ccall(
(:Pa_HostApiTypeIdToHostApiIndex, libportaudio),
PaHostApiIndex,
(PaHostApiTypeId,),
type,
)
end
function Pa_HostApiDeviceIndexToDeviceIndex(hostApi, hostApiDeviceIndex)
ccall(
(:Pa_HostApiDeviceIndexToDeviceIndex, libportaudio),
PaDeviceIndex,
(PaHostApiIndex, Cint),
hostApi,
hostApiDeviceIndex,
)
end
mutable struct PaHostErrorInfo
hostApiType::PaHostApiTypeId
errorCode::Clong
errorText::Ptr{Cchar}
end
function Pa_GetLastHostErrorInfo()
ccall((:Pa_GetLastHostErrorInfo, libportaudio), Ptr{PaHostErrorInfo}, ())
end
function Pa_GetDeviceCount()
ccall((:Pa_GetDeviceCount, libportaudio), PaDeviceIndex, ())
end
function Pa_GetDefaultInputDevice()
ccall((:Pa_GetDefaultInputDevice, libportaudio), PaDeviceIndex, ())
end
function Pa_GetDefaultOutputDevice()
ccall((:Pa_GetDefaultOutputDevice, libportaudio), PaDeviceIndex, ())
end
const PaTime = Cdouble
const PaSampleFormat = Culong
mutable struct PaDeviceInfo mutable struct PaDeviceInfo
structVersion::Cint struct_version::Cint
name::Ptr{Cchar} name::Ptr{Cchar}
hostApi::PaHostApiIndex host_api::PaHostApiIndex
maxInputChannels::Cint max_input_channels::Cint
maxOutputChannels::Cint max_output_channels::Cint
defaultLowInputLatency::PaTime default_low_input_latency::PaTime
defaultLowOutputLatency::PaTime default_low_output_latency::PaTime
defaultHighInputLatency::PaTime default_high_input_latency::PaTime
defaultHighOutputLatency::PaTime default_high_output_latency::PaTime
defaultSampleRate::Cdouble default_sample_rate::Cdouble
end end
function Pa_GetDeviceInfo(device) Pa_GetDeviceCount() = @locked ccall((:Pa_GetDeviceCount, libportaudio), PaDeviceIndex, ())
ccall((:Pa_GetDeviceInfo, libportaudio), Ptr{PaDeviceInfo}, (PaDeviceIndex,), device)
end
struct PaStreamParameters Pa_GetDeviceInfo(i) = unsafe_load(@locked ccall((:Pa_GetDeviceInfo, libportaudio),
Ptr{PaDeviceInfo}, (PaDeviceIndex,), i))
Pa_GetDefaultInputDevice() = @locked ccall((:Pa_GetDefaultInputDevice, libportaudio),
PaDeviceIndex, ())
Pa_GetDefaultOutputDevice() = @locked ccall((:Pa_GetDefaultOutputDevice, libportaudio),
PaDeviceIndex, ())
# Stream Functions
mutable struct Pa_StreamParameters
device::PaDeviceIndex device::PaDeviceIndex
channelCount::Cint channelCount::Cint
sampleFormat::PaSampleFormat sampleFormat::PaSampleFormat
suggestedLatency::PaTime suggestedLatency::PaTime
hostApiSpecificStreamInfo::Ptr{Cvoid} hostAPISpecificStreamInfo::Ptr{Cvoid}
end
function Pa_IsFormatSupported(inputParameters, outputParameters, sampleRate)
ccall(
(:Pa_IsFormatSupported, libportaudio),
PaError,
(Ptr{PaStreamParameters}, Ptr{PaStreamParameters}, Cdouble),
inputParameters,
outputParameters,
sampleRate,
)
end
const PaStream = Cvoid
const PaStreamFlags = Culong
mutable struct PaStreamCallbackTimeInfo
inputBufferAdcTime::PaTime
currentTime::PaTime
outputBufferDacTime::PaTime
end
const PaStreamCallbackFlags = Culong
@enum PaStreamCallbackResult::UInt32 begin
paContinue = 0
paComplete = 1
paAbort = 2
end
# typedef int PaStreamCallback ( const void * input , void * output , unsigned long frameCount , const PaStreamCallbackTimeInfo * timeInfo , PaStreamCallbackFlags statusFlags , void * userData )
const PaStreamCallback = Cvoid
function Pa_OpenStream(
stream,
inputParameters,
outputParameters,
sampleRate,
framesPerBuffer,
streamFlags,
streamCallback,
userData,
)
ccall(
(:Pa_OpenStream, libportaudio),
PaError,
(
Ptr{Ptr{PaStream}},
Ptr{PaStreamParameters},
Ptr{PaStreamParameters},
Cdouble,
Culong,
PaStreamFlags,
Ptr{Cvoid},
Ptr{Cvoid},
),
stream,
inputParameters,
outputParameters,
sampleRate,
framesPerBuffer,
streamFlags,
streamCallback,
userData,
)
end
function Pa_OpenDefaultStream(
stream,
numInputChannels,
numOutputChannels,
sampleFormat,
sampleRate,
framesPerBuffer,
streamCallback,
userData,
)
ccall(
(:Pa_OpenDefaultStream, libportaudio),
PaError,
(
Ptr{Ptr{PaStream}},
Cint,
Cint,
PaSampleFormat,
Cdouble,
Culong,
Ptr{Cvoid},
Ptr{Cvoid},
),
stream,
numInputChannels,
numOutputChannels,
sampleFormat,
sampleRate,
framesPerBuffer,
streamCallback,
userData,
)
end
function Pa_CloseStream(stream)
ccall((:Pa_CloseStream, libportaudio), PaError, (Ptr{PaStream},), stream)
end
# typedef void PaStreamFinishedCallback ( void * userData )
const PaStreamFinishedCallback = Cvoid
function Pa_SetStreamFinishedCallback(stream, streamFinishedCallback)
ccall(
(:Pa_SetStreamFinishedCallback, libportaudio),
PaError,
(Ptr{PaStream}, Ptr{Cvoid}),
stream,
streamFinishedCallback,
)
end
function Pa_StartStream(stream)
ccall((:Pa_StartStream, libportaudio), PaError, (Ptr{PaStream},), stream)
end
function Pa_StopStream(stream)
ccall((:Pa_StopStream, libportaudio), PaError, (Ptr{PaStream},), stream)
end
function Pa_AbortStream(stream)
ccall((:Pa_AbortStream, libportaudio), PaError, (Ptr{PaStream},), stream)
end
function Pa_IsStreamStopped(stream)
ccall((:Pa_IsStreamStopped, libportaudio), PaError, (Ptr{PaStream},), stream)
end
function Pa_IsStreamActive(stream)
ccall((:Pa_IsStreamActive, libportaudio), PaError, (Ptr{PaStream},), stream)
end end
mutable struct PaStreamInfo mutable struct PaStreamInfo
@ -332,108 +162,112 @@ mutable struct PaStreamInfo
sampleRate::Cdouble sampleRate::Cdouble
end end
function Pa_GetStreamInfo(stream) # function Pa_OpenDefaultStream(inChannels, outChannels,
ccall((:Pa_GetStreamInfo, libportaudio), Ptr{PaStreamInfo}, (Ptr{PaStream},), stream) # sampleFormat::PaSampleFormat,
# sampleRate, framesPerBuffer)
# streamPtr = Ref{PaStream}(0)
# err = ccall((:Pa_OpenDefaultStream, libportaudio),
# PaError, (Ref{PaStream}, Cint, Cint,
# PaSampleFormat, Cdouble, Culong,
# Ref{Cvoid}, Ref{Cvoid}),
# streamPtr, inChannels, outChannels, sampleFormat, sampleRate,
# framesPerBuffer, C_NULL, C_NULL)
# handle_status(err)
#
# streamPtr[]
# end
#
function Pa_OpenStream(inParams, outParams,
sampleRate, framesPerBuffer,
flags::PaStreamFlags,
callback, userdata)
streamPtr = Ref{PaStream}(0)
err = @locked ccall((:Pa_OpenStream, libportaudio), PaError,
(Ref{PaStream}, Ref{Pa_StreamParameters}, Ref{Pa_StreamParameters},
Cdouble, Culong, PaStreamFlags, Ref{Cvoid},
# it seems like we should be able to use Ref{T} here, with
# userdata::T above, and avoid the `pointer_from_objref` below.
# that's not working on 0.6 though, and it shouldn't really
# matter because userdata should be GC-rooted anyways
Ptr{Cvoid}),
streamPtr, inParams, outParams,
float(sampleRate), framesPerBuffer, flags,
callback === nothing ? C_NULL : callback,
userdata === nothing ? C_NULL : pointer_from_objref(userdata))
handle_status(err)
streamPtr[]
end end
function Pa_GetStreamTime(stream) function Pa_StartStream(stream::PaStream)
ccall((:Pa_GetStreamTime, libportaudio), PaTime, (Ptr{PaStream},), stream) err = @locked ccall((:Pa_StartStream, libportaudio), PaError,
(PaStream,), stream)
handle_status(err)
end end
function Pa_GetStreamCpuLoad(stream) function Pa_StopStream(stream::PaStream)
ccall((:Pa_GetStreamCpuLoad, libportaudio), Cdouble, (Ptr{PaStream},), stream) err = @locked ccall((:Pa_StopStream, libportaudio), PaError,
(PaStream,), stream)
handle_status(err)
end end
function Pa_ReadStream(stream, buffer, frames) function Pa_CloseStream(stream::PaStream)
ccall( err = @locked ccall((:Pa_CloseStream, libportaudio), PaError,
(:Pa_ReadStream, libportaudio), (PaStream,), stream)
PaError, handle_status(err)
(Ptr{PaStream}, Ptr{Cvoid}, Culong),
stream,
buffer,
frames,
)
end end
function Pa_WriteStream(stream, buffer, frames) function Pa_GetStreamReadAvailable(stream::PaStream)
ccall( avail = @locked ccall((:Pa_GetStreamReadAvailable, libportaudio), Clong,
(:Pa_WriteStream, libportaudio), (PaStream,), stream)
PaError, avail >= 0 || handle_status(avail)
(Ptr{PaStream}, Ptr{Cvoid}, Culong), avail
stream,
buffer,
frames,
)
end end
function Pa_GetStreamReadAvailable(stream) function Pa_GetStreamWriteAvailable(stream::PaStream)
ccall((:Pa_GetStreamReadAvailable, libportaudio), Clong, (Ptr{PaStream},), stream) avail = @locked ccall((:Pa_GetStreamWriteAvailable, libportaudio), Clong,
(PaStream,), stream)
avail >= 0 || handle_status(avail)
avail
end end
function Pa_GetStreamWriteAvailable(stream) function Pa_ReadStream(stream::PaStream, buf::Array, frames::Integer=length(buf),
ccall((:Pa_GetStreamWriteAvailable, libportaudio), Clong, (Ptr{PaStream},), stream) show_warnings::Bool=true)
frames <= length(buf) || error("Need a buffer at least $frames long")
err = @tcall @locked ccall((:Pa_ReadStream, libportaudio), PaError,
(PaStream, Ptr{Cvoid}, Culong),
stream, buf, frames)
handle_status(err, show_warnings)
buf
end end
function Pa_GetSampleSize(format) function Pa_WriteStream(stream::PaStream, buf::Array, frames::Integer=length(buf),
ccall((:Pa_GetSampleSize, libportaudio), PaError, (PaSampleFormat,), format) show_warnings::Bool=true)
frames <= length(buf) || error("Need a buffer at least $frames long")
err = @tcall @locked ccall((:Pa_WriteStream, libportaudio), PaError,
(PaStream, Ptr{Cvoid}, Culong),
stream, buf, frames)
handle_status(err, show_warnings)
nothing
end end
function Pa_Sleep(msec) # function Pa_GetStreamInfo(stream::PaStream)
ccall((:Pa_Sleep, libportaudio), Cvoid, (Clong,), msec) # infoptr = ccall((:Pa_GetStreamInfo, libportaudio), Ptr{PaStreamInfo},
end # (PaStream, ), stream)
#
const paNoDevice = PaDeviceIndex(-1) # unsafe_load(infoptr)
# end
const paUseHostApiSpecificDeviceSpecification = PaDeviceIndex(-2) #
# General utility function to handle the status from the Pa_* functions
const paFloat32 = PaSampleFormat(0x00000001) function handle_status(err::PaError, show_warnings::Bool=true)
if err == PA_OUTPUT_UNDERFLOWED || err == PA_INPUT_OVERFLOWED
const paInt32 = PaSampleFormat(0x00000002) if show_warnings
msg = @locked ccall((:Pa_GetErrorText, libportaudio),
const paInt24 = PaSampleFormat(0x00000004) Ptr{Cchar}, (PaError,), err)
@warn("libportaudio: " * unsafe_string(msg))
const paInt16 = PaSampleFormat(0x00000008) end
elseif err != PA_NO_ERROR
const paInt8 = PaSampleFormat(0x00000010) msg = @locked ccall((:Pa_GetErrorText, libportaudio),
Ptr{Cchar}, (PaError,), err)
const paUInt8 = PaSampleFormat(0x00000020) throw(ErrorException("libportaudio: " * unsafe_string(msg)))
const paCustomFormat = PaSampleFormat(0x00010000)
const paNonInterleaved = PaSampleFormat(0x80000000)
const paFormatIsSupported = 0
const paFramesPerBufferUnspecified = 0
const paNoFlag = PaStreamFlags(0)
const paClipOff = PaStreamFlags(0x00000001)
const paDitherOff = PaStreamFlags(0x00000002)
const paNeverDropInput = PaStreamFlags(0x00000004)
const paPrimeOutputBuffersUsingStreamCallback = PaStreamFlags(0x00000008)
const paPlatformSpecificFlags = PaStreamFlags(0xffff0000)
const paInputUnderflow = PaStreamCallbackFlags(0x00000001)
const paInputOverflow = PaStreamCallbackFlags(0x00000002)
const paOutputUnderflow = PaStreamCallbackFlags(0x00000004)
const paOutputOverflow = PaStreamCallbackFlags(0x00000008)
const paPrimingOutput = PaStreamCallbackFlags(0x00000010)
# exports
const PREFIXES = ["Pa", "pa"]
for name in names(@__MODULE__; all = true), prefix in PREFIXES
if startswith(string(name), prefix)
@eval export $name
end end
end end
end # module

View file

@ -1,29 +0,0 @@
# precompile some important functions
const DEFAULT_SINK_MESSENGER_TYPE = Messenger{Float32, SampledSignalsWriter, Tuple{Matrix{Float32}, Int64, Int64}, Int64}
const DEFAULT_SOURCE_MESSENGER_TYPE = Messenger{Float32, SampledSignalsReader, Tuple{Matrix{Float32}, Int64, Int64}, Int64}
const DEFAULT_STREAM_TYPE = PortAudioStream{DEFAULT_SINK_MESSENGER_TYPE, DEFAULT_SOURCE_MESSENGER_TYPE}
const DEFAULT_SINK_TYPE = PortAudioSink{DEFAULT_SINK_MESSENGER_TYPE, DEFAULT_SOURCE_MESSENGER_TYPE}
const DEFAULT_SOURCE_TYPE = PortAudioSource{DEFAULT_SINK_MESSENGER_TYPE, DEFAULT_SOURCE_MESSENGER_TYPE}
precompile(close, (DEFAULT_STREAM_TYPE,))
precompile(devices, ())
precompile(__init__, ())
precompile(isopen, (DEFAULT_STREAM_TYPE,))
precompile(nchannels, (DEFAULT_SINK_TYPE,))
precompile(nchannels, (DEFAULT_SOURCE_TYPE,))
precompile(PortAudioStream, (Int, Int))
precompile(PortAudioStream, (String, Int, Int))
precompile(PortAudioStream, (String, String, Int, Int))
precompile(samplerate, (DEFAULT_STREAM_TYPE,))
precompile(send, (DEFAULT_SINK_MESSENGER_TYPE,))
precompile(send, (DEFAULT_SOURCE_MESSENGER_TYPE,))
precompile(unsafe_read!, (DEFAULT_SOURCE_TYPE, Vector{Float32}, Int, Int))
precompile(unsafe_read!, (DEFAULT_SOURCE_TYPE, Matrix{Float32}, Int, Int))
precompile(unsafe_write, (DEFAULT_SINK_TYPE, Vector{Float32}, Int, Int))
precompile(unsafe_write, (DEFAULT_SINK_TYPE, Matrix{Float32}, Int, Int))

View file

@ -1,65 +1,9 @@
#!/usr/bin/env julia #!/usr/bin/env julia
using Base.Sys: iswindows
using Documenter: doctest
using PortAudio:
combine_default_sample_rates,
devices,
get_default_input_index,
get_default_output_index,
get_device,
get_input_type,
get_output_type,
handle_status,
initialize,
name,
PortAudioException,
PortAudio,
PortAudioDevice,
PortAudioStream,
safe_load,
seek_alsa_conf,
terminate,
write_buffer
using PortAudio.LibPortAudio:
Pa_AbortStream,
PaError,
PaErrorCode,
paFloat32,
Pa_GetDefaultHostApi,
Pa_GetDeviceInfo,
Pa_GetHostApiCount,
Pa_GetLastHostErrorInfo,
Pa_GetSampleSize,
Pa_GetStreamCpuLoad,
Pa_GetStreamInfo,
Pa_GetStreamReadAvailable,
Pa_GetStreamTime,
Pa_GetStreamWriteAvailable,
Pa_GetVersionInfo,
Pa_HostApiDeviceIndexToDeviceIndex,
paHostApiNotFound,
Pa_HostApiTypeIdToHostApiIndex,
PaHostErrorInfo,
paInDevelopment,
paInvalidDevice,
Pa_IsFormatSupported,
Pa_IsStreamActive,
paNoError,
paNoFlag,
paNotInitialized,
Pa_OpenDefaultStream,
paOutputUnderflowed,
Pa_SetStreamFinishedCallback,
Pa_Sleep,
Pa_StopStream,
PaStream,
PaStreamInfo,
PaStreamParameters,
PaVersionInfo
using SampledSignals: nchannels, s, SampleBuf, samplerate, SinSource
using Test: @test, @test_logs, @test_nowarn, @testset, @test_throws
@testset "Tests without sound" begin using PortAudio
using Test
@testset "PortAudio Tests" begin
@testset "Reports version" begin @testset "Reports version" begin
io = IOBuffer() io = IOBuffer()
PortAudio.versioninfo(io) PortAudio.versioninfo(io)
@ -69,188 +13,6 @@ using Test: @test, @test_logs, @test_nowarn, @testset, @test_throws
end end
@testset "Can list devices without crashing" begin @testset "Can list devices without crashing" begin
display(devices()) PortAudio.devices()
println()
end end
@testset "libortaudio without sound" begin
@test handle_status(Pa_GetHostApiCount()) >= 0
@test handle_status(Pa_GetDefaultHostApi()) >= 0
# version info not available on windows?
if !Sys.iswindows()
@test safe_load(Pa_GetVersionInfo(), ErrorException("no info")) isa
PaVersionInfo
end
@test safe_load(Pa_GetLastHostErrorInfo(), ErrorException("no info")) isa
PaHostErrorInfo
@test PaErrorCode(Pa_IsFormatSupported(C_NULL, C_NULL, 0.0)) == paInvalidDevice
@test PaErrorCode(
Pa_OpenDefaultStream(Ref(C_NULL), 0, 0, paFloat32, 0.0, 0, C_NULL, C_NULL),
) == paInvalidDevice
end
@testset "Errors without sound" begin
@test sprint(showerror, PortAudioException(paNotInitialized)) ==
"PortAudioException: PortAudio not initialized"
@test_throws KeyError("foobarbaz") get_device("foobarbaz")
@test_throws KeyError(-1) get_device(-1)
@test_throws ArgumentError("Could not find alsa.conf in ()") seek_alsa_conf(())
@test_logs (:warn, "libportaudio: Output underflowed") handle_status(
PaError(paOutputUnderflowed),
)
@test_throws PortAudioException(paNotInitialized) handle_status(
PaError(paNotInitialized),
)
Pa_Sleep(1)
@test Pa_GetSampleSize(paFloat32) == 4
end
# make sure we can terminate, then reinitialize
terminate()
initialize()
end
if isempty(devices())
@test_throws ArgumentError("No input device available") get_default_input_index()
else
@testset "Tests with sound" begin
# these default values are specific to local machines
input_name = get_device(get_default_input_index()).name
output_name = get_device(get_default_output_index()).name
@testset "Interactive tests" begin
println("Recording...")
stream = PortAudioStream(input_name, output_name, 2, 0; adjust_channels = true)
buffer = read(stream, 5s)
@test size(buffer) ==
(round(Int, 5 * samplerate(stream)), nchannels(stream.source))
close(stream)
sleep(1)
println("Playing back recording...")
PortAudioStream(input_name, output_name, 0, 2; adjust_channels = true) do stream
write(stream, buffer)
end
sleep(1)
println("Testing pass-through")
stream = PortAudioStream(input_name, output_name, 2, 2; adjust_channels = true)
write_buffer(stream.sink_messenger.buffer, acquire_lock = false)
sink = stream.sink
source = stream.source
@test sprint(show, stream) == """
PortAudioStream{Float32}
Samplerate: 44100Hz
2 channel sink: $(repr(output_name))
2 channel source: $(repr(input_name))"""
@test sprint(show, source) == "2 channel source: $(repr(input_name))"
@test sprint(show, sink) == "2 channel sink: $(repr(output_name))"
write(stream, stream, 5s)
@test PaErrorCode(handle_status(Pa_StopStream(stream.pointer_to))) == paNoError
@test isopen(stream)
close(stream)
sleep(1)
@test !isopen(stream)
@test !isopen(sink)
@test !isopen(source)
println("done")
end
@testset "Samplerate-converting writing" begin
PortAudioStream(input_name, output_name, 0, 2; adjust_channels = true) do stream
write(
stream,
SinSource(eltype(stream), samplerate(stream) * 0.8, [220, 330]),
3s,
)
println("expected blip")
write(
stream,
SinSource(eltype(stream), samplerate(stream) * 1.2, [220, 330]),
3s,
)
end
end
sleep(1)
# no way to check that the right data is actually getting read or written here,
# but at least it's not crashing.
@testset "Queued Writing" begin
PortAudioStream(input_name, output_name, 0, 2; adjust_channels = true) do stream
buffer = SampleBuf(
rand(eltype(stream), 48000, nchannels(stream.sink)) * 0.1,
samplerate(stream),
)
frame_count_1 = @async write(stream, buffer)
frame_count_2 = @async write(stream, buffer)
@test fetch(frame_count_1) == 48000
println("expected blip")
@test fetch(frame_count_2) == 48000
end
sleep(1)
end
@testset "Queued Reading" begin
PortAudioStream(input_name, output_name, 2, 0; adjust_channels = true) do stream
buffer = SampleBuf(
rand(eltype(stream), 48000, nchannels(stream.source)) * 0.1,
samplerate(stream),
)
frame_count_1 = @async read!(stream, buffer)
frame_count_2 = @async read!(stream, buffer)
@test fetch(frame_count_1) == 48000
@test fetch(frame_count_2) == 48000
end
sleep(1)
end
@testset "Constructors" begin
PortAudioStream(2, maximum; adjust_channels = true) do stream
@test isopen(stream)
end
PortAudioStream(output_name; adjust_channels = true) do stream
@test isopen(stream)
end
PortAudioStream(input_name, output_name; adjust_channels = true) do stream
@test isopen(stream)
end
end
@testset "Errors with sound" begin
big = typemax(Int)
@test_throws DomainError(
typemax(Int),
"$big exceeds maximum output channels for $output_name",
) PortAudioStream(input_name, output_name, 0, big)
@test_throws ArgumentError("Input or output must have at least 1 channel") PortAudioStream(
input_name,
output_name,
0,
0;
adjust_channels = true,
)
@test_throws ArgumentError("""
Default sample rate 0 for input \"$input_name\" disagrees with
default sample rate 1 for output \"$output_name\".
Please specify a sample rate.
""") combine_default_sample_rates(
get_device(input_name),
0,
get_device(output_name),
1,
)
end
@testset "libportaudio with sound" begin
@test PaErrorCode(Pa_HostApiTypeIdToHostApiIndex(paInDevelopment)) ==
paHostApiNotFound
@test Pa_HostApiDeviceIndexToDeviceIndex(paInDevelopment, 0) == 0
stream = PortAudioStream(input_name, output_name, 2, 2; adjust_channels = true)
pointer_to = stream.pointer_to
@test handle_status(Pa_GetStreamReadAvailable(pointer_to)) >= 0
@test handle_status(Pa_GetStreamWriteAvailable(pointer_to)) >= 0
@test Bool(handle_status(Pa_IsStreamActive(pointer_to)))
@test safe_load(Pa_GetStreamInfo(pointer_to), ErrorException("no info")) isa
PaStreamInfo
@test Pa_GetStreamTime(pointer_to) >= 0
@test Pa_GetStreamCpuLoad(pointer_to) >= 0
@test PaErrorCode(handle_status(Pa_AbortStream(pointer_to))) == paNoError
@test PaErrorCode(
handle_status(Pa_SetStreamFinishedCallback(pointer_to, C_NULL)),
) == paNoError
end
end
doctest(PortAudio)
end end

View file

@ -38,28 +38,24 @@ end
end end
@testset "Samplerate-converting writing" begin @testset "Samplerate-converting writing" begin
stream = PortAudioStream(0, 2) stream = PortAudioStream(0, 2)
write(stream, SinSource(eltype(stream), samplerate(stream) * 0.8, [220, 330]), 3s) write(stream, SinSource(eltype(stream), samplerate(stream)*0.8, [220, 330]), 3s)
write(stream, SinSource(eltype(stream), samplerate(stream) * 1.2, [220, 330]), 3s) write(stream, SinSource(eltype(stream), samplerate(stream)*1.2, [220, 330]), 3s)
flush(stream) flush(stream)
close(stream) close(stream)
end end
@testset "Open Device by name" begin @testset "Open Device by name" begin
stream = PortAudioStream(default_indev, default_outdev) stream = PortAudioStream(default_indev, default_outdev)
buf = read(stream, 0.001s) buf = read(stream, 0.001s)
@test size(buf) == @test size(buf) == (round(Int, 0.001 * samplerate(stream)), nchannels(stream.source))
(round(Int, 0.001 * samplerate(stream)), nchannels(stream.source))
write(stream, buf) write(stream, buf)
io = IOBuffer() io = IOBuffer()
show(io, stream) show(io, stream)
@test occursin( @test occursin("""
""" PortAudioStream{Float32}
PortAudioStream{Float32} Samplerate: 44100.0Hz
Samplerate: 44100.0Hz Buffer Size: 4096 frames
Buffer Size: 4096 frames 2 channel sink: "$default_outdev"
2 channel sink: "$default_outdev" 2 channel source: "$default_indev\"""", String(take!(io)))
2 channel source: "$default_indev\"""",
String(take!(io)),
)
close(stream) close(stream)
end end
@testset "Error on wrong name" begin @testset "Error on wrong name" begin
@ -69,10 +65,7 @@ PortAudioStream{Float32}
# but at least it's not crashing. # but at least it's not crashing.
@testset "Queued Writing" begin @testset "Queued Writing" begin
stream = PortAudioStream(0, 2) stream = PortAudioStream(0, 2)
buf = SampleBuf( buf = SampleBuf(rand(eltype(stream), 48000, nchannels(stream.sink))*0.1, samplerate(stream))
rand(eltype(stream), 48000, nchannels(stream.sink)) * 0.1,
samplerate(stream),
)
t1 = @async write(stream, buf) t1 = @async write(stream, buf)
t2 = @async write(stream, buf) t2 = @async write(stream, buf)
@test fetch(t1) == 48000 @test fetch(t1) == 48000
@ -82,10 +75,7 @@ PortAudioStream{Float32}
end end
@testset "Queued Reading" begin @testset "Queued Reading" begin
stream = PortAudioStream(2, 0) stream = PortAudioStream(2, 0)
buf = SampleBuf( buf = SampleBuf(rand(eltype(stream), 48000, nchannels(stream.source))*0.1, samplerate(stream))
rand(eltype(stream), 48000, nchannels(stream.source)) * 0.1,
samplerate(stream),
)
t1 = @async read!(stream, buf) t1 = @async read!(stream, buf)
t2 = @async read!(stream, buf) t2 = @async read!(stream, buf)
@test fetch(t1) == 48000 @test fetch(t1) == 48000